diff options
| author | James Cowgill <james410@cowgill.org.uk> | 2013-08-23 09:57:55 +0100 |
|---|---|---|
| committer | James Cowgill <james410@cowgill.org.uk> | 2013-08-23 09:57:55 +0100 |
| commit | 9a298ca833d9b6a3425bb30c2e52cf04e34aeb7c (patch) | |
| tree | d46630a885bcea03bbea036b86c645dc6c55708d /src/SFML/Audio | |
| parent | 0969839d538a385254c6eced9648acc7299876cc (diff) | |
Imported Upstream version 2.1+dfsg
Diffstat (limited to 'src/SFML/Audio')
22 files changed, 1352 insertions, 7562 deletions
diff --git a/src/SFML/Audio/OpenAL.hpp b/src/SFML/Audio/ALCheck.cpp index c1dea8c..0558f11 100755..100644 --- a/src/SFML/Audio/OpenAL.hpp +++ b/src/SFML/Audio/ALCheck.cpp @@ -1,132 +1,109 @@ -//////////////////////////////////////////////////////////// -// -// SFML - Simple and Fast Multimedia Library -// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com) -// -// This software is provided 'as-is', without any express or implied warranty. -// In no event will the authors be held liable for any damages arising from the use of this software. -// -// Permission is granted to anyone to use this software for any purpose, -// including commercial applications, and to alter it and redistribute it freely, -// subject to the following restrictions: -// -// 1. The origin of this software must not be misrepresented; -// you must not claim that you wrote the original software. -// If you use this software in a product, an acknowledgment -// in the product documentation would be appreciated but is not required. -// -// 2. Altered source versions must be plainly marked as such, -// and must not be misrepresented as being the original software. -// -// 3. This notice may not be removed or altered from any source distribution. -// -//////////////////////////////////////////////////////////// - -#ifndef SFML_OPENAL_HPP -#define SFML_OPENAL_HPP - -//////////////////////////////////////////////////////////// -// Headers -//////////////////////////////////////////////////////////// -#include <SFML/Config.hpp> - -#if defined(SFML_SYSTEM_MACOS) -#include <OpenAL/al.h> -#include <OpenAL/alc.h> -#else -#include <AL/al.h> -#include <AL/alc.h> -#endif - -#include <iostream> -#include <string> - - -namespace sf -{ -namespace priv -{ -//////////////////////////////////////////////////////////// -/// Let's define a macro to quickly check every OpenAL -/// API calls -/// -//////////////////////////////////////////////////////////// -#ifdef SFML_DEBUG - - // If in debug mode, perform a test on every call - #define ALCheck(Func) ((Func), priv::ALCheckError(__FILE__, __LINE__)) - -#else - - // Else, we don't add any overhead - #define ALCheck(Func) (Func) - -#endif - - -//////////////////////////////////////////////////////////// -/// Check last OpenAL error -/// -//////////////////////////////////////////////////////////// -inline void ALCheckError(const std::string& File, unsigned int Line) -{ - // Get the last error - ALenum ErrorCode = alGetError(); - - if (ErrorCode != AL_NO_ERROR) - { - std::string Error, Desc; - - // Decode the error code - switch (ErrorCode) - { - case AL_INVALID_NAME : - { - Error = "AL_INVALID_NAME"; - Desc = "an unacceptable name has been specified"; - break; - } - - case AL_INVALID_ENUM : - { - Error = "AL_INVALID_ENUM"; - Desc = "an unacceptable value has been specified for an enumerated argument"; - break; - } - - case AL_INVALID_VALUE : - { - Error = "AL_INVALID_VALUE"; - Desc = "a numeric argument is out of range"; - break; - } - - case AL_INVALID_OPERATION : - { - Error = "AL_INVALID_OPERATION"; - Desc = "the specified operation is not allowed in the current state"; - break; - } - - case AL_OUT_OF_MEMORY : - { - Error = "AL_OUT_OF_MEMORY"; - Desc = "there is not enough memory left to execute the command"; - break; - } - } - - // Log the error - std::cerr << "An internal OpenAL call failed in " - << File.substr(File.find_last_of("\\/") + 1) << " (" << Line << ") : " - << Error << ", " << Desc - << std::endl; - } -} - -} // namespace priv - -} // namespace sf - - -#endif // SFML_OPENAL_HPP +////////////////////////////////////////////////////////////
+//
+// SFML - Simple and Fast Multimedia Library
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
+//
+// This software is provided 'as-is', without any express or implied warranty.
+// In no event will the authors be held liable for any damages arising from the use of this software.
+//
+// Permission is granted to anyone to use this software for any purpose,
+// including commercial applications, and to alter it and redistribute it freely,
+// subject to the following restrictions:
+//
+// 1. The origin of this software must not be misrepresented;
+// you must not claim that you wrote the original software.
+// If you use this software in a product, an acknowledgment
+// in the product documentation would be appreciated but is not required.
+//
+// 2. Altered source versions must be plainly marked as such,
+// and must not be misrepresented as being the original software.
+//
+// 3. This notice may not be removed or altered from any source distribution.
+//
+////////////////////////////////////////////////////////////
+
+////////////////////////////////////////////////////////////
+// Headers
+////////////////////////////////////////////////////////////
+#include <SFML/Audio/ALCheck.hpp>
+#include <SFML/Audio/AudioDevice.hpp>
+#include <SFML/System/Err.hpp>
+
+
+namespace sf
+{
+namespace priv
+{
+////////////////////////////////////////////////////////////
+void alCheckError(const std::string& file, unsigned int line)
+{
+ // Get the last error
+ ALenum errorCode = alGetError();
+
+ if (errorCode != AL_NO_ERROR)
+ {
+ std::string error, description;
+
+ // Decode the error code
+ switch (errorCode)
+ {
+ case AL_INVALID_NAME :
+ {
+ error = "AL_INVALID_NAME";
+ description = "an unacceptable name has been specified";
+ break;
+ }
+
+ case AL_INVALID_ENUM :
+ {
+ error = "AL_INVALID_ENUM";
+ description = "an unacceptable value has been specified for an enumerated argument";
+ break;
+ }
+
+ case AL_INVALID_VALUE :
+ {
+ error = "AL_INVALID_VALUE";
+ description = "a numeric argument is out of range";
+ break;
+ }
+
+ case AL_INVALID_OPERATION :
+ {
+ error = "AL_INVALID_OPERATION";
+ description = "the specified operation is not allowed in the current state";
+ break;
+ }
+
+ case AL_OUT_OF_MEMORY :
+ {
+ error = "AL_OUT_OF_MEMORY";
+ description = "there is not enough memory left to execute the command";
+ break;
+ }
+ }
+
+ // Log the error
+ err() << "An internal OpenAL call failed in "
+ << file.substr(file.find_last_of("\\/") + 1) << " (" << line << ") : "
+ << error << ", " << description
+ << std::endl;
+ }
+}
+
+
+////////////////////////////////////////////////////////////
+/// Make sure that OpenAL is initialized
+////////////////////////////////////////////////////////////
+void ensureALInit()
+{
+ // The audio device is instanciated on demand rather than at global startup,
+ // which solves a lot of weird crashes and errors.
+ // It is destroyed at global exit which is fine.
+
+ static AudioDevice globalDevice;
+}
+
+} // namespace priv
+
+} // namespace sf
diff --git a/src/SFML/Audio/AudioResource.cpp b/src/SFML/Audio/ALCheck.hpp index a1a352c..7c7d659 100755..100644 --- a/src/SFML/Audio/AudioResource.cpp +++ b/src/SFML/Audio/ALCheck.hpp @@ -1,7 +1,7 @@ ////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
@@ -22,39 +22,57 @@ //
////////////////////////////////////////////////////////////
+#ifndef SFML_ALCHECK_HPP
+#define SFML_ALCHECK_HPP
+
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
-#include <SFML/Audio/AudioResource.hpp>
-#include <SFML/Audio/AudioDevice.hpp>
+#include <SFML/Config.hpp>
+#include <iostream>
+#include <string>
+#include <al.h>
+#include <alc.h>
namespace sf
{
+namespace priv
+{
////////////////////////////////////////////////////////////
-/// Default constructor
+/// Let's define a macro to quickly check every OpenAL API calls
////////////////////////////////////////////////////////////
-AudioResource::AudioResource()
-{
- priv::AudioDevice::AddReference();
-}
+#ifdef SFML_DEBUG
+
+ // If in debug mode, perform a test on every call
+ #define alCheck(Func) ((Func), priv::alCheckError(__FILE__, __LINE__))
+
+#else
+
+ // Else, we don't add any overhead
+ #define alCheck(Func) (Func)
+
+#endif
////////////////////////////////////////////////////////////
-/// Copy constructor
+/// Check the last OpenAL error
+///
+/// \param file Source file where the call is located
+/// \param line Line number of the source file where the call is located
+///
////////////////////////////////////////////////////////////
-AudioResource::AudioResource(const AudioResource&)
-{
- priv::AudioDevice::AddReference();
-}
-
+void alCheckError(const std::string& file, unsigned int line);
////////////////////////////////////////////////////////////
-/// Destructor
+/// Make sure that OpenAL is initialized
+///
////////////////////////////////////////////////////////////
-AudioResource::~AudioResource()
-{
- priv::AudioDevice::RemoveReference();
-}
+void ensureALInit();
+
+} // namespace priv
} // namespace sf
+
+
+#endif // SFML_ALCHECK_HPP
diff --git a/src/SFML/Audio/AudioDevice.cpp b/src/SFML/Audio/AudioDevice.cpp index 096a163..d218b91 100755..100644 --- a/src/SFML/Audio/AudioDevice.cpp +++ b/src/SFML/Audio/AudioDevice.cpp @@ -1,7 +1,7 @@ ////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
@@ -26,143 +26,98 @@ // Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/AudioDevice.hpp>
-#include <SFML/Audio/AudioResource.hpp>
+#include <SFML/Audio/ALCheck.hpp>
#include <SFML/Audio/Listener.hpp>
-#include <algorithm>
-#include <iostream>
+#include <SFML/System/Err.hpp>
+namespace
+{
+ ALCdevice* audioDevice = NULL;
+ ALCcontext* audioContext = NULL;
+}
+
namespace sf
{
namespace priv
{
////////////////////////////////////////////////////////////
-// Static member data
-////////////////////////////////////////////////////////////
-AudioDevice* AudioDevice::ourInstance;
-
-
-////////////////////////////////////////////////////////////
-/// Default constructor
-////////////////////////////////////////////////////////////
-AudioDevice::AudioDevice() :
-myRefCount(0)
+AudioDevice::AudioDevice()
{
// Create the device
- myDevice = alcOpenDevice(NULL);
+ audioDevice = alcOpenDevice(NULL);
- if (myDevice)
+ if (audioDevice)
{
// Create the context
- myContext = alcCreateContext(myDevice, NULL);
+ audioContext = alcCreateContext(audioDevice, NULL);
- if (myContext)
+ if (audioContext)
{
// Set the context as the current one (we'll only need one)
- alcMakeContextCurrent(myContext);
-
- // Initialize the listener, located at the origin and looking along the Z axis
- Listener::SetPosition(0.f, 0.f, 0.f);
- Listener::SetTarget(0.f, 0.f, -1.f);
+ alcMakeContextCurrent(audioContext);
}
else
{
- std::cerr << "Failed to create the audio context" << std::endl;
+ err() << "Failed to create the audio context" << std::endl;
}
}
else
{
- std::cerr << "Failed to open the audio device" << std::endl;
+ err() << "Failed to open the audio device" << std::endl;
}
}
////////////////////////////////////////////////////////////
-/// Destructor
-////////////////////////////////////////////////////////////
AudioDevice::~AudioDevice()
{
// Destroy the context
alcMakeContextCurrent(NULL);
- if (myContext)
- alcDestroyContext(myContext);
-
- // Destroy the device
- if (myDevice)
- alcCloseDevice(myDevice);
-}
-
-
-////////////////////////////////////////////////////////////
-/// Get the unique instance of the class
-////////////////////////////////////////////////////////////
-AudioDevice& AudioDevice::GetInstance()
-{
- // Create the audio device if it doesn't exist
- if (!ourInstance)
- ourInstance = new AudioDevice;
-
- return *ourInstance;
-}
-
+ if (audioContext)
+ alcDestroyContext(audioContext);
-////////////////////////////////////////////////////////////
-/// Add a reference to the audio device
-////////////////////////////////////////////////////////////
-void AudioDevice::AddReference()
-{
- // Create the audio device if it doesn't exist
- if (!ourInstance)
- ourInstance = new AudioDevice;
-
- // Increase the references count
- ourInstance->myRefCount++;
+ // Destroy the device
+ if (audioDevice)
+ alcCloseDevice(audioDevice);
}
////////////////////////////////////////////////////////////
-/// Remove a reference to the audio device
-////////////////////////////////////////////////////////////
-void AudioDevice::RemoveReference()
+bool AudioDevice::isExtensionSupported(const std::string& extension)
{
- // Decrease the references count
- ourInstance->myRefCount--;
+ ensureALInit();
- // Destroy the audio device if the references count reaches 0
- if (ourInstance->myRefCount == 0)
- {
- delete ourInstance;
- ourInstance = NULL;
- }
+ if ((extension.length() > 2) && (extension.substr(0, 3) == "ALC"))
+ return alcIsExtensionPresent(audioDevice, extension.c_str()) != AL_FALSE;
+ else
+ return alIsExtensionPresent(extension.c_str()) != AL_FALSE;
}
////////////////////////////////////////////////////////////
-/// Get the OpenAL audio device
-////////////////////////////////////////////////////////////
-ALCdevice* AudioDevice::GetDevice() const
+int AudioDevice::getFormatFromChannelCount(unsigned int channelCount)
{
- return myDevice;
-}
-
+ ensureALInit();
-////////////////////////////////////////////////////////////
-/// Get the OpenAL format that matches the given number of channels
-////////////////////////////////////////////////////////////
-ALenum AudioDevice::GetFormatFromChannelsCount(unsigned int ChannelsCount) const
-{
// Find the good format according to the number of channels
- switch (ChannelsCount)
+ int format = 0;
+ switch (channelCount)
{
- case 1 : return AL_FORMAT_MONO16;
- case 2 : return AL_FORMAT_STEREO16;
- case 4 : return alGetEnumValue("AL_FORMAT_QUAD16");
- case 6 : return alGetEnumValue("AL_FORMAT_51CHN16");
- case 7 : return alGetEnumValue("AL_FORMAT_61CHN16");
- case 8 : return alGetEnumValue("AL_FORMAT_71CHN16");
+ case 1 : format = AL_FORMAT_MONO16; break;
+ case 2 : format = AL_FORMAT_STEREO16; break;
+ case 4 : format = alGetEnumValue("AL_FORMAT_QUAD16"); break;
+ case 6 : format = alGetEnumValue("AL_FORMAT_51CHN16"); break;
+ case 7 : format = alGetEnumValue("AL_FORMAT_61CHN16"); break;
+ case 8 : format = alGetEnumValue("AL_FORMAT_71CHN16"); break;
+ default : format = 0; break;
}
- return 0;
+ // Fixes a bug on OS X
+ if (format == -1)
+ format = 0;
+
+ return format;
}
} // namespace priv
diff --git a/src/SFML/Audio/AudioDevice.hpp b/src/SFML/Audio/AudioDevice.hpp index 6b47177..01c7b4c 100755..100644 --- a/src/SFML/Audio/AudioDevice.hpp +++ b/src/SFML/Audio/AudioDevice.hpp @@ -1,7 +1,7 @@ ////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
@@ -28,90 +28,59 @@ ////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
-#include <SFML/Audio/OpenAL.hpp>
#include <set>
#include <string>
namespace sf
{
-class AudioResource;
-
namespace priv
{
-
////////////////////////////////////////////////////////////
-/// AudioDevice is the high-level wrapper around the audio API,
-/// it manages creation and destruction of the audio device and context
-/// and stores the device capabilities
+/// \brief High-level wrapper around the audio API, it manages
+/// the creation and destruction of the audio device and
+/// context and stores the device capabilities
+///
////////////////////////////////////////////////////////////
class AudioDevice
{
public :
////////////////////////////////////////////////////////////
- /// Get the unique instance of the class
- ///
- /// \return Unique instance of the class
- ///
- ////////////////////////////////////////////////////////////
- static AudioDevice& GetInstance();
-
- ////////////////////////////////////////////////////////////
- /// Add a reference to the audio device
+ /// \brief Default constructor
///
////////////////////////////////////////////////////////////
- static void AddReference();
+ AudioDevice();
////////////////////////////////////////////////////////////
- /// Remove a reference to the audio device
+ /// \brief Destructor
///
////////////////////////////////////////////////////////////
- static void RemoveReference();
+ ~AudioDevice();
////////////////////////////////////////////////////////////
- /// Get the OpenAL audio device
- ///
- /// \return OpenAL device (cannot be NULL)
+ /// \brief Check if an OpenAL extension is supported
///
- ////////////////////////////////////////////////////////////
- ALCdevice* GetDevice() const;
-
- ////////////////////////////////////////////////////////////
- /// Get the OpenAL format that matches the given number of channels
+ /// This functions automatically finds whether it
+ /// is an AL or ALC extension, and calls the corresponding
+ /// function.
///
- /// \param ChannelsCount : Number of channels
+ /// \param extension Name of the extension to test
///
- /// \return OpenAL device (cannot be NULL)
+ /// \return True if the extension is supported, false if not
///
////////////////////////////////////////////////////////////
- ALenum GetFormatFromChannelsCount(unsigned int ChannelsCount) const;
-
-private :
+ static bool isExtensionSupported(const std::string& extension);
////////////////////////////////////////////////////////////
- /// Default constructor
+ /// \brief Get the OpenAL format that matches the given number of channels
///
- ////////////////////////////////////////////////////////////
- AudioDevice();
-
- ////////////////////////////////////////////////////////////
- /// Destructor
+ /// \param channelCount Number of channels
+ ///
+ /// \return Corresponding format
///
////////////////////////////////////////////////////////////
- ~AudioDevice();
-
- ////////////////////////////////////////////////////////////
- // Static member data
- ////////////////////////////////////////////////////////////
- static AudioDevice* ourInstance; ///< Unique instance of the audio device
-
- ////////////////////////////////////////////////////////////
- // Member data
- ////////////////////////////////////////////////////////////
- ALCdevice* myDevice; ///< Audio device
- ALCcontext* myContext; ///< Audio context
- unsigned int myRefCount; ///< References count
+ static int getFormatFromChannelCount(unsigned int channelCount);
};
} // namespace priv
diff --git a/src/SFML/Audio/CMakeLists.txt b/src/SFML/Audio/CMakeLists.txt new file mode 100644 index 0000000..818b9b6 --- /dev/null +++ b/src/SFML/Audio/CMakeLists.txt @@ -0,0 +1,53 @@ +
+set(INCROOT ${PROJECT_SOURCE_DIR}/include/SFML/Audio)
+set(SRCROOT ${PROJECT_SOURCE_DIR}/src/SFML/Audio)
+
+# all source files
+set(SRC
+ ${SRCROOT}/ALCheck.cpp
+ ${SRCROOT}/ALCheck.hpp
+ ${SRCROOT}/AudioDevice.cpp
+ ${SRCROOT}/AudioDevice.hpp
+ ${INCROOT}/Export.hpp
+ ${SRCROOT}/Listener.cpp
+ ${INCROOT}/Listener.hpp
+ ${SRCROOT}/Music.cpp
+ ${INCROOT}/Music.hpp
+ ${SRCROOT}/Sound.cpp
+ ${INCROOT}/Sound.hpp
+ ${SRCROOT}/SoundBuffer.cpp
+ ${INCROOT}/SoundBuffer.hpp
+ ${SRCROOT}/SoundBufferRecorder.cpp
+ ${INCROOT}/SoundBufferRecorder.hpp
+ ${SRCROOT}/SoundFile.cpp
+ ${SRCROOT}/SoundFile.hpp
+ ${SRCROOT}/SoundRecorder.cpp
+ ${INCROOT}/SoundRecorder.hpp
+ ${SRCROOT}/SoundSource.cpp
+ ${INCROOT}/SoundSource.hpp
+ ${SRCROOT}/SoundStream.cpp
+ ${INCROOT}/SoundStream.hpp
+)
+source_group("" FILES ${SRC})
+
+# let CMake know about our additional audio libraries paths (on Windows and OSX)
+if(WINDOWS)
+ set(CMAKE_INCLUDE_PATH ${CMAKE_INCLUDE_PATH} "${PROJECT_SOURCE_DIR}/extlibs/headers/AL")
+ set(CMAKE_INCLUDE_PATH ${CMAKE_INCLUDE_PATH} "${PROJECT_SOURCE_DIR}/extlibs/headers/libsndfile/windows")
+elseif (MACOSX)
+ set(CMAKE_INCLUDE_PATH ${CMAKE_INCLUDE_PATH} "${PROJECT_SOURCE_DIR}/extlibs/headers/libsndfile/osx")
+ set(CMAKE_LIBRARY_PATH ${CMAKE_LIBRARY_PATH} "${PROJECT_SOURCE_DIR}/extlibs/libs-osx/Frameworks")
+endif()
+
+# find external libraries
+find_package(OpenAL REQUIRED)
+find_package(Sndfile REQUIRED)
+
+# add include paths of external libraries
+include_directories(${OPENAL_INCLUDE_DIR} ${SNDFILE_INCLUDE_DIR})
+
+# define the sfml-audio target
+sfml_add_library(sfml-audio
+ SOURCES ${SRC}
+ DEPENDS sfml-system
+ EXTERNAL_LIBS ${OPENAL_LIBRARY} ${SNDFILE_LIBRARY})
diff --git a/src/SFML/Audio/Listener.cpp b/src/SFML/Audio/Listener.cpp index b51b3c5..a4995ce 100755..100644 --- a/src/SFML/Audio/Listener.cpp +++ b/src/SFML/Audio/Listener.cpp @@ -1,7 +1,7 @@ ////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
@@ -26,93 +26,86 @@ // Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/Listener.hpp>
-#include <SFML/Audio/OpenAL.hpp>
+#include <SFML/Audio/ALCheck.hpp>
namespace sf
{
////////////////////////////////////////////////////////////
-/// Change the global volume of all the sounds
-////////////////////////////////////////////////////////////
-void Listener::SetGlobalVolume(float Volume)
+void Listener::setGlobalVolume(float volume)
{
- ALCheck(alListenerf(AL_GAIN, Volume * 0.01f));
+ priv::ensureALInit();
+
+ alCheck(alListenerf(AL_GAIN, volume * 0.01f));
}
////////////////////////////////////////////////////////////
-/// Get the current value of the global volume of all the sounds
-////////////////////////////////////////////////////////////
-float Listener::GetGlobalVolume()
+float Listener::getGlobalVolume()
{
- float Volume = 0.f;
- ALCheck(alGetListenerf(AL_GAIN, &Volume));
+ priv::ensureALInit();
+
+ float volume = 0.f;
+ alCheck(alGetListenerf(AL_GAIN, &volume));
- return Volume;
+ return volume * 100;
}
////////////////////////////////////////////////////////////
-/// Change the position of the listener (take 3 values)
-////////////////////////////////////////////////////////////
-void Listener::SetPosition(float X, float Y, float Z)
+void Listener::setPosition(float x, float y, float z)
{
- ALCheck(alListener3f(AL_POSITION, X, Y, Z));
+ priv::ensureALInit();
+
+ alCheck(alListener3f(AL_POSITION, x, y, z));
}
////////////////////////////////////////////////////////////
-/// Change the position of the listener (take a 3D vector)
-////////////////////////////////////////////////////////////
-void Listener::SetPosition(const Vector3f& Position)
+void Listener::setPosition(const Vector3f& position)
{
- SetPosition(Position.x, Position.y, Position.z);
+ setPosition(position.x, position.y, position.z);
}
////////////////////////////////////////////////////////////
-/// Get the current position of the listener
-////////////////////////////////////////////////////////////
-Vector3f Listener::GetPosition()
+Vector3f Listener::getPosition()
{
- Vector3f Position;
- ALCheck(alGetListener3f(AL_POSITION, &Position.x, &Position.y, &Position.z));
+ priv::ensureALInit();
+
+ Vector3f position;
+ alCheck(alGetListener3f(AL_POSITION, &position.x, &position.y, &position.z));
- return Position;
+ return position;
}
////////////////////////////////////////////////////////////
-/// Change the orientation of the listener (the point
-/// he must look at) (take 3 values)
-////////////////////////////////////////////////////////////
-void Listener::SetTarget(float X, float Y, float Z)
+void Listener::setDirection(float x, float y, float z)
{
- float Orientation[] = {X, Y, Z, 0.f, 1.f, 0.f};
- ALCheck(alListenerfv(AL_ORIENTATION, Orientation));
+ priv::ensureALInit();
+
+ float orientation[] = {x, y, z, 0.f, 1.f, 0.f};
+ alCheck(alListenerfv(AL_ORIENTATION, orientation));
}
////////////////////////////////////////////////////////////
-/// Change the orientation of the listener (the point
-/// he must look at) (take a 3D vector)
-////////////////////////////////////////////////////////////
-void Listener::SetTarget(const Vector3f& Target)
+void Listener::setDirection(const Vector3f& direction)
{
- SetTarget(Target.x, Target.y, Target.z);
+ setDirection(direction.x, direction.y, direction.z);
}
////////////////////////////////////////////////////////////
-/// Get the current orientation of the listener (the point
-/// he's looking at)
-////////////////////////////////////////////////////////////
-Vector3f Listener::GetTarget()
+Vector3f Listener::getDirection()
{
- float Orientation[6];
- ALCheck(alGetListenerfv(AL_ORIENTATION, Orientation));
+ priv::ensureALInit();
+
+ float orientation[6];
+ alCheck(alGetListenerfv(AL_ORIENTATION, orientation));
- return Vector3f(Orientation[0], Orientation[1], Orientation[2]);
+ return Vector3f(orientation[0], orientation[1], orientation[2]);
}
} // namespace sf
diff --git a/src/SFML/Audio/Makefile b/src/SFML/Audio/Makefile deleted file mode 100755 index 0215e80..0000000 --- a/src/SFML/Audio/Makefile +++ /dev/null @@ -1,39 +0,0 @@ -SRC = $(wildcard *.cpp)
-SRCVORBIS = $(wildcard ./stb_vorbis/*.c)
-OBJ = $(SRC:.cpp=.o)
-OBJVORBIS = $(SRCVORBIS:.c=.o)
-
-ifeq ($(STATIC), yes)
- LIB = libsfml-audio-s.a
- LIBNAME = $(LIBPATH)/$(LIB)
- INSTALL =
-else
- LIB = libsfml-audio.so
- LIBNAME = $(LIBPATH)/$(LIB).$(VERSION)
- INSTALL = && $(LN) $(LNFLAGS) $(LIB).$(VERSION) $(DESTLIBDIR)/$(LIB)
-endif
-
-all: $(LIB)
-
-libsfml-audio-s.a: $(OBJ) $(OBJVORBIS)
- $(AR) $(ARFLAGS) $(LIBNAME) $(OBJ) $(OBJVORBIS)
-
-libsfml-audio.so: $(OBJ) $(OBJVORBIS)
- $(CPP) $(LDFLAGS) -Wl,-soname,$(LIB).$(VERSION) -o $(LIBNAME) $(OBJ) $(OBJVORBIS) -lsndfile -lopenal
-
-$(OBJ): %.o: %.cpp
- $(CPP) -o $@ -c $< $(CFLAGS)
-
-$(OBJVORBIS): %.o: %.c
- $(CC) -o $@ -c $< $(CFLAGSEXT)
-
-.PHONY: clean mrproper
-
-clean:
- @rm -rf $(OBJ) $(OBJVORBIS)
-
-mrproper: clean
- @rm -rf $(LIBNAME)
-
-install:
- @($(CP) $(LIBNAME) $(DESTLIBDIR) $(INSTALL))
diff --git a/src/SFML/Audio/Music.cpp b/src/SFML/Audio/Music.cpp index 5023ff5..ef784e4 100755..100644 --- a/src/SFML/Audio/Music.cpp +++ b/src/SFML/Audio/Music.cpp @@ -1,7 +1,7 @@ ////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
@@ -26,128 +26,126 @@ // Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/Music.hpp>
-#include <SFML/Audio/OpenAL.hpp>
+#include <SFML/Audio/ALCheck.hpp>
#include <SFML/Audio/SoundFile.hpp>
+#include <SFML/System/Lock.hpp>
+#include <SFML/System/Err.hpp>
#include <fstream>
-#include <iostream>
namespace sf
{
////////////////////////////////////////////////////////////
-/// Construct the music with a buffer size
-////////////////////////////////////////////////////////////
-Music::Music(std::size_t BufferSize) :
-myFile (NULL),
-myDuration(0.f),
-mySamples (BufferSize)
+Music::Music() :
+m_file (new priv::SoundFile),
+m_duration()
{
}
////////////////////////////////////////////////////////////
-/// Destructor
-////////////////////////////////////////////////////////////
Music::~Music()
{
// We must stop before destroying the file :)
- Stop();
+ stop();
- delete myFile;
+ delete m_file;
}
////////////////////////////////////////////////////////////
-/// Open a music file (doesn't play it -- call Play() for that)
-////////////////////////////////////////////////////////////
-bool Music::OpenFromFile(const std::string& Filename)
+bool Music::openFromFile(const std::string& filename)
{
// First stop the music if it was already running
- Stop();
-
- // Create the sound file implementation, and open it in read mode
- delete myFile;
- myFile = priv::SoundFile::CreateRead(Filename);
- if (!myFile)
- {
- std::cerr << "Failed to open \"" << Filename << "\" for reading" << std::endl;
- return false;
- }
+ stop();
- // Compute the duration
- myDuration = static_cast<float>(myFile->GetSamplesCount()) / myFile->GetSampleRate() / myFile->GetChannelsCount();
+ // Open the underlying sound file
+ if (!m_file->openRead(filename))
+ return false;
- // Initialize the stream
- Initialize(myFile->GetChannelsCount(), myFile->GetSampleRate());
+ // Perform common initializations
+ initialize();
return true;
}
////////////////////////////////////////////////////////////
-/// Open a music file from memory (doesn't play it -- call Play() for that)
-////////////////////////////////////////////////////////////
-bool Music::OpenFromMemory(const char* Data, std::size_t SizeInBytes)
+bool Music::openFromMemory(const void* data, std::size_t sizeInBytes)
{
// First stop the music if it was already running
- Stop();
-
- // Create the sound file implementation, and open it in read mode
- delete myFile;
- myFile = priv::SoundFile::CreateRead(Data, SizeInBytes);
- if (!myFile)
- {
- std::cerr << "Failed to open music from memory for reading" << std::endl;
- return false;
- }
+ stop();
- // Compute the duration
- myDuration = static_cast<float>(myFile->GetSamplesCount()) / myFile->GetSampleRate();
+ // Open the underlying sound file
+ if (!m_file->openRead(data, sizeInBytes))
+ return false;
- // Initialize the stream
- Initialize(myFile->GetChannelsCount(), myFile->GetSampleRate());
+ // Perform common initializations
+ initialize();
return true;
}
////////////////////////////////////////////////////////////
-/// /see SoundStream::OnStart
-////////////////////////////////////////////////////////////
-bool Music::OnStart()
+bool Music::openFromStream(InputStream& stream)
{
- return myFile && myFile->Restart();
+ // First stop the music if it was already running
+ stop();
+
+ // Open the underlying sound file
+ if (!m_file->openRead(stream))
+ return false;
+
+ // Perform common initializations
+ initialize();
+
+ return true;
}
////////////////////////////////////////////////////////////
-/// /see SoundStream::OnGetData
+Time Music::getDuration() const
+{
+ return m_duration;
+}
+
+
////////////////////////////////////////////////////////////
-bool Music::OnGetData(SoundStream::Chunk& Data)
+bool Music::onGetData(SoundStream::Chunk& data)
{
- if (myFile)
- {
- // Fill the chunk parameters
- Data.Samples = &mySamples[0];
- Data.NbSamples = myFile->Read(&mySamples[0], mySamples.size());
-
- // Check if we have reached the end of the audio file
- return Data.NbSamples == mySamples.size();
- }
- else
- {
- return false;
- }
+ Lock lock(m_mutex);
+
+ // Fill the chunk parameters
+ data.samples = &m_samples[0];
+ data.sampleCount = m_file->read(&m_samples[0], m_samples.size());
+
+ // Check if we have reached the end of the audio file
+ return data.sampleCount == m_samples.size();
}
////////////////////////////////////////////////////////////
-/// Get the sound duration
+void Music::onSeek(Time timeOffset)
+{
+ Lock lock(m_mutex);
+
+ m_file->seek(timeOffset);
+}
+
+
////////////////////////////////////////////////////////////
-float Music::GetDuration() const
+void Music::initialize()
{
- return myDuration;
+ // Compute the music duration
+ m_duration = seconds(static_cast<float>(m_file->getSampleCount()) / m_file->getSampleRate() / m_file->getChannelCount());
+
+ // Resize the internal buffer so that it can contain 1 second of audio samples
+ m_samples.resize(m_file->getSampleRate() * m_file->getChannelCount());
+
+ // Initialize the stream
+ SoundStream::initialize(m_file->getChannelCount(), m_file->getSampleRate());
}
} // namespace sf
diff --git a/src/SFML/Audio/Sound.cpp b/src/SFML/Audio/Sound.cpp index 84136db..8619cab 100755..100644 --- a/src/SFML/Audio/Sound.cpp +++ b/src/SFML/Audio/Sound.cpp @@ -1,7 +1,7 @@ ////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
@@ -27,378 +27,170 @@ ////////////////////////////////////////////////////////////
#include <SFML/Audio/Sound.hpp>
#include <SFML/Audio/SoundBuffer.hpp>
-#include <SFML/Audio/OpenAL.hpp>
+#include <SFML/Audio/ALCheck.hpp>
namespace sf
{
////////////////////////////////////////////////////////////
-/// Default constructor
-////////////////////////////////////////////////////////////
-Sound::Sound()
+Sound::Sound() :
+m_buffer(NULL)
{
- ALCheck(alGenSources(1, &mySource));
- ALCheck(alSourcei(mySource, AL_BUFFER, 0));
}
////////////////////////////////////////////////////////////
-/// Construct the sound from its parameters
-////////////////////////////////////////////////////////////
-Sound::Sound(const SoundBuffer& Buffer, bool Loop, float Pitch, float Volume, const Vector3f& Position) :
-myBuffer(NULL)
+Sound::Sound(const SoundBuffer& buffer) :
+m_buffer(NULL)
{
- ALCheck(alGenSources(1, &mySource));
-
- SetBuffer(Buffer);
- SetLoop(Loop);
- SetPitch(Pitch);
- SetVolume(Volume);
- SetPosition(Position);
+ setBuffer(buffer);
}
////////////////////////////////////////////////////////////
-/// Copy constructor
-////////////////////////////////////////////////////////////
-Sound::Sound(const Sound& Copy) :
-AudioResource(Copy),
-myBuffer(NULL)
+Sound::Sound(const Sound& copy) :
+SoundSource(copy),
+m_buffer (NULL)
{
- ALCheck(alGenSources(1, &mySource));
-
- if (Copy.myBuffer)
- SetBuffer(*Copy.myBuffer);
- SetLoop(Copy.GetLoop());
- SetPitch(Copy.GetPitch());
- SetVolume(Copy.GetVolume());
- SetPosition(Copy.GetPosition());
- SetRelativeToListener(Copy.IsRelativeToListener());
- SetMinDistance(Copy.GetMinDistance());
- SetAttenuation(Copy.GetAttenuation());
+ if (copy.m_buffer)
+ setBuffer(*copy.m_buffer);
+ setLoop(copy.getLoop());
}
////////////////////////////////////////////////////////////
-/// Destructor
-////////////////////////////////////////////////////////////
Sound::~Sound()
{
- if (mySource)
- {
- if (myBuffer)
- {
- Stop();
- ALCheck(alSourcei(mySource, AL_BUFFER, 0));
- myBuffer->DetachSound(this);
- }
- ALCheck(alDeleteSources(1, &mySource));
- }
+ stop();
+ if (m_buffer)
+ m_buffer->detachSound(this);
}
////////////////////////////////////////////////////////////
-/// Play the sound
-////////////////////////////////////////////////////////////
-void Sound::Play()
+void Sound::play()
{
- ALCheck(alSourcePlay(mySource));
+ alCheck(alSourcePlay(m_source));
}
////////////////////////////////////////////////////////////
-/// Pause the sound
-////////////////////////////////////////////////////////////
-void Sound::Pause()
+void Sound::pause()
{
- ALCheck(alSourcePause(mySource));
+ alCheck(alSourcePause(m_source));
}
////////////////////////////////////////////////////////////
-/// Stop the sound
-////////////////////////////////////////////////////////////
-void Sound::Stop()
+void Sound::stop()
{
- ALCheck(alSourceStop(mySource));
+ alCheck(alSourceStop(m_source));
}
////////////////////////////////////////////////////////////
-/// Set the source buffer
-////////////////////////////////////////////////////////////
-void Sound::SetBuffer(const SoundBuffer& Buffer)
+void Sound::setBuffer(const SoundBuffer& buffer)
{
// First detach from the previous buffer
- if (myBuffer)
+ if (m_buffer)
{
- Stop();
- myBuffer->DetachSound(this);
+ stop();
+ m_buffer->detachSound(this);
}
// Assign and use the new buffer
- myBuffer = &Buffer;
- myBuffer->AttachSound(this);
- ALCheck(alSourcei(mySource, AL_BUFFER, myBuffer->myBuffer));
-}
-
-
-////////////////////////////////////////////////////////////
-/// Set the sound loop state
-////////////////////////////////////////////////////////////
-void Sound::SetLoop(bool Loop)
-{
- ALCheck(alSourcei(mySource, AL_LOOPING, Loop));
-}
-
-
-////////////////////////////////////////////////////////////
-/// Set the sound pitch
-////////////////////////////////////////////////////////////
-void Sound::SetPitch(float Pitch)
-{
- ALCheck(alSourcef(mySource, AL_PITCH, Pitch));
-}
-
-
-////////////////////////////////////////////////////////////
-/// Set the sound volume
-////////////////////////////////////////////////////////////
-void Sound::SetVolume(float Volume)
-{
- ALCheck(alSourcef(mySource, AL_GAIN, Volume * 0.01f));
-}
-
-////////////////////////////////////////////////////////////
-/// Set the sound position (take 3 values).
-/// The default position is (0, 0, 0)
-////////////////////////////////////////////////////////////
-void Sound::SetPosition(float X, float Y, float Z)
-{
- ALCheck(alSource3f(mySource, AL_POSITION, X, Y, Z));
-}
-
-
-////////////////////////////////////////////////////////////
-/// Set the sound position (take a 3D vector).
-/// The default position is (0, 0, 0)
-////////////////////////////////////////////////////////////
-void Sound::SetPosition(const Vector3f& Position)
-{
- SetPosition(Position.x, Position.y, Position.z);
-}
-
-
-////////////////////////////////////////////////////////////
-/// Make the sound's position relative to the listener's
-/// position, or absolute.
-/// The default value is false (absolute)
-////////////////////////////////////////////////////////////
-void Sound::SetRelativeToListener(bool Relative)
-{
- ALCheck(alSourcei(mySource, AL_SOURCE_RELATIVE, Relative));
-}
-
-
-////////////////////////////////////////////////////////////
-/// Set the minimum distance - closer than this distance,
-/// the listener will hear the sound at its maximum volume.
-/// The default minimum distance is 1.0
-////////////////////////////////////////////////////////////
-void Sound::SetMinDistance(float MinDistance)
-{
- ALCheck(alSourcef(mySource, AL_REFERENCE_DISTANCE, MinDistance));
-}
-
-
-////////////////////////////////////////////////////////////
-/// Set the attenuation factor - the higher the attenuation, the
-/// more the sound will be attenuated with distance from listener.
-/// The default attenuation factor 1.0
-////////////////////////////////////////////////////////////
-void Sound::SetAttenuation(float Attenuation)
-{
- ALCheck(alSourcef(mySource, AL_ROLLOFF_FACTOR, Attenuation));
+ m_buffer = &buffer;
+ m_buffer->attachSound(this);
+ alCheck(alSourcei(m_source, AL_BUFFER, m_buffer->m_buffer));
}
////////////////////////////////////////////////////////////
-/// Set the current playing position of the sound
-////////////////////////////////////////////////////////////
-void Sound::SetPlayingOffset(float TimeOffset)
+void Sound::setLoop(bool Loop)
{
- ALCheck(alSourcef(mySource, AL_SEC_OFFSET, TimeOffset));
+ alCheck(alSourcei(m_source, AL_LOOPING, Loop));
}
////////////////////////////////////////////////////////////
-/// Get the source buffer
-////////////////////////////////////////////////////////////
-const SoundBuffer* Sound::GetBuffer() const
+void Sound::setPlayingOffset(Time timeOffset)
{
- return myBuffer;
+ alCheck(alSourcef(m_source, AL_SEC_OFFSET, timeOffset.asSeconds()));
}
////////////////////////////////////////////////////////////
-/// Tell whether or not the sound is looping
-////////////////////////////////////////////////////////////
-bool Sound::GetLoop() const
+const SoundBuffer* Sound::getBuffer() const
{
- ALint Loop;
- ALCheck(alGetSourcei(mySource, AL_LOOPING, &Loop));
-
- return Loop != 0;
+ return m_buffer;
}
////////////////////////////////////////////////////////////
-/// Get the pitch
-////////////////////////////////////////////////////////////
-float Sound::GetPitch() const
+bool Sound::getLoop() const
{
- ALfloat Pitch;
- ALCheck(alGetSourcef(mySource, AL_PITCH, &Pitch));
+ ALint loop;
+ alCheck(alGetSourcei(m_source, AL_LOOPING, &loop));
- return Pitch;
+ return loop != 0;
}
////////////////////////////////////////////////////////////
-/// Get the volume
-////////////////////////////////////////////////////////////
-float Sound::GetVolume() const
+Time Sound::getPlayingOffset() const
{
- ALfloat Gain;
- ALCheck(alGetSourcef(mySource, AL_GAIN, &Gain));
+ ALfloat secs = 0.f;
+ alCheck(alGetSourcef(m_source, AL_SEC_OFFSET, &secs));
- return Gain * 100.f;
+ return seconds(secs);
}
////////////////////////////////////////////////////////////
-/// Get the sound position
-////////////////////////////////////////////////////////////
-Vector3f Sound::GetPosition() const
+Sound::Status Sound::getStatus() const
{
- Vector3f Position;
- ALCheck(alGetSource3f(mySource, AL_POSITION, &Position.x, &Position.y, &Position.z));
-
- return Position;
+ return SoundSource::getStatus();
}
////////////////////////////////////////////////////////////
-/// Tell if the sound's position is relative to the listener's
-/// position, or if it's absolute
-////////////////////////////////////////////////////////////
-bool Sound::IsRelativeToListener() const
-{
- ALint Relative;
- ALCheck(alGetSourcei(mySource, AL_SOURCE_RELATIVE, &Relative));
-
- return Relative != 0;
-}
-
-
-////////////////////////////////////////////////////////////
-/// Get the minimum distance
-////////////////////////////////////////////////////////////
-float Sound::GetMinDistance() const
-{
- ALfloat MinDistance;
- ALCheck(alGetSourcef(mySource, AL_REFERENCE_DISTANCE, &MinDistance));
-
- return MinDistance;
-}
-
-
-////////////////////////////////////////////////////////////
-/// Get the attenuation factor
-////////////////////////////////////////////////////////////
-float Sound::GetAttenuation() const
-{
- ALfloat Attenuation;
- ALCheck(alGetSourcef(mySource, AL_ROLLOFF_FACTOR, &Attenuation));
-
- return Attenuation;
-}
-
-
-////////////////////////////////////////////////////////////
-/// Get the current playing position of the sound
-////////////////////////////////////////////////////////////
-float Sound::GetPlayingOffset() const
-{
- ALfloat Seconds = 0.f;
- ALCheck(alGetSourcef(mySource, AL_SEC_OFFSET, &Seconds));
-
- return Seconds;
-}
-
-
-////////////////////////////////////////////////////////////
-/// Get the status of the sound (stopped, paused, playing)
-////////////////////////////////////////////////////////////
-Sound::Status Sound::GetStatus() const
-{
- ALint State;
- ALCheck(alGetSourcei(mySource, AL_SOURCE_STATE, &State));
-
- switch (State)
- {
- case AL_INITIAL :
- case AL_STOPPED : return Stopped;
- case AL_PAUSED : return Paused;
- case AL_PLAYING : return Playing;
- }
-
- return Stopped;
-}
-
-
-////////////////////////////////////////////////////////////
-/// Assignment operator
-////////////////////////////////////////////////////////////
-Sound& Sound::operator =(const Sound& Other)
+Sound& Sound::operator =(const Sound& right)
{
// Here we don't use the copy-and-swap idiom, because it would mess up
// the list of sound instances contained in the buffers
// Detach the sound instance from the previous buffer (if any)
- if (myBuffer)
+ if (m_buffer)
{
- Stop();
- myBuffer->DetachSound(this);
- myBuffer = NULL;
+ stop();
+ m_buffer->detachSound(this);
+ m_buffer = NULL;
}
// Copy the sound attributes
- if (Other.myBuffer)
- SetBuffer(*Other.myBuffer);
- SetLoop(Other.GetLoop());
- SetPitch(Other.GetPitch());
- SetVolume(Other.GetVolume());
- SetPosition(Other.GetPosition());
- SetRelativeToListener(Other.IsRelativeToListener());
- SetMinDistance(Other.GetMinDistance());
- SetAttenuation(Other.GetAttenuation());
+ if (right.m_buffer)
+ setBuffer(*right.m_buffer);
+ setLoop(right.getLoop());
+ setPitch(right.getPitch());
+ setVolume(right.getVolume());
+ setPosition(right.getPosition());
+ setRelativeToListener(right.isRelativeToListener());
+ setMinDistance(right.getMinDistance());
+ setAttenuation(right.getAttenuation());
return *this;
}
////////////////////////////////////////////////////////////
-/// Reset the internal buffer
-////////////////////////////////////////////////////////////
-void Sound::ResetBuffer()
+void Sound::resetBuffer()
{
// First stop the sound in case it is playing
- Stop();
+ stop();
// Detach the buffer
- ALCheck(alSourcei(mySource, AL_BUFFER, 0));
- myBuffer = NULL;
+ alCheck(alSourcei(m_source, AL_BUFFER, 0));
+ m_buffer = NULL;
}
} // namespace sf
diff --git a/src/SFML/Audio/SoundBuffer.cpp b/src/SFML/Audio/SoundBuffer.cpp index e726b13..406a9fe 100755..100644 --- a/src/SFML/Audio/SoundBuffer.cpp +++ b/src/SFML/Audio/SoundBuffer.cpp @@ -1,7 +1,7 @@ ////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
@@ -29,164 +29,106 @@ #include <SFML/Audio/SoundFile.hpp>
#include <SFML/Audio/Sound.hpp>
#include <SFML/Audio/AudioDevice.hpp>
-#include <SFML/Audio/OpenAL.hpp>
-#include <iostream>
+#include <SFML/Audio/ALCheck.hpp>
+#include <SFML/System/Err.hpp>
#include <memory>
namespace sf
{
////////////////////////////////////////////////////////////
-/// Default constructor
-////////////////////////////////////////////////////////////
SoundBuffer::SoundBuffer() :
-myBuffer (0),
-myDuration(0.f)
+m_buffer (0),
+m_duration()
{
+ priv::ensureALInit();
+
// Create the buffer
- ALCheck(alGenBuffers(1, &myBuffer));
+ alCheck(alGenBuffers(1, &m_buffer));
}
////////////////////////////////////////////////////////////
-/// Copy constructor
-////////////////////////////////////////////////////////////
-SoundBuffer::SoundBuffer(const SoundBuffer& Copy) :
-AudioResource (Copy),
-Resource<SoundBuffer>(Copy),
-myBuffer (0),
-mySamples (Copy.mySamples),
-myDuration (Copy.myDuration),
-mySounds () // don't copy the attached sounds
+SoundBuffer::SoundBuffer(const SoundBuffer& copy) :
+m_buffer (0),
+m_samples (copy.m_samples),
+m_duration(copy.m_duration),
+m_sounds () // don't copy the attached sounds
{
// Create the buffer
- ALCheck(alGenBuffers(1, &myBuffer));
+ alCheck(alGenBuffers(1, &m_buffer));
// Update the internal buffer with the new samples
- Update(Copy.GetChannelsCount(), Copy.GetSampleRate());
+ update(copy.getChannelCount(), copy.getSampleRate());
}
////////////////////////////////////////////////////////////
-/// Destructor
-////////////////////////////////////////////////////////////
SoundBuffer::~SoundBuffer()
{
// First detach the buffer from the sounds that use it (to avoid OpenAL errors)
- for (SoundList::const_iterator it = mySounds.begin(); it != mySounds.end(); ++it)
- (*it)->ResetBuffer();
+ for (SoundList::const_iterator it = m_sounds.begin(); it != m_sounds.end(); ++it)
+ (*it)->resetBuffer();
// Destroy the buffer
- if (myBuffer)
- ALCheck(alDeleteBuffers(1, &myBuffer));
+ if (m_buffer)
+ alCheck(alDeleteBuffers(1, &m_buffer));
}
////////////////////////////////////////////////////////////
-/// Load the sound buffer from a file
-////////////////////////////////////////////////////////////
-bool SoundBuffer::LoadFromFile(const std::string& Filename)
+bool SoundBuffer::loadFromFile(const std::string& filename)
{
- // Create the sound file
- std::auto_ptr<priv::SoundFile> File(priv::SoundFile::CreateRead(Filename));
-
- // Open the sound file
- if (File.get())
- {
- // Get the sound parameters
- std::size_t NbSamples = File->GetSamplesCount();
- unsigned int ChannelsCount = File->GetChannelsCount();
- unsigned int SampleRate = File->GetSampleRate();
-
- // Read the samples from the opened file
- mySamples.resize(NbSamples);
- if (File->Read(&mySamples[0], NbSamples) == NbSamples)
- {
- // Update the internal buffer with the new samples
- return Update(ChannelsCount, SampleRate);
- }
- else
- {
- // Error...
- std::cerr << "Failed to read audio data from file \"" << Filename << "\"" << std::endl;
-
- return false;
- }
- }
+ priv::SoundFile file;
+ if (file.openRead(filename))
+ return initialize(file);
else
- {
- // Error...
- std::cerr << "Failed to load sound buffer from file \"" << Filename << "\"" << std::endl;
-
return false;
- }
}
////////////////////////////////////////////////////////////
-/// Load the sound buffer from a file in memory
-////////////////////////////////////////////////////////////
-bool SoundBuffer::LoadFromMemory(const char* Data, std::size_t SizeInBytes)
+bool SoundBuffer::loadFromMemory(const void* data, std::size_t sizeInBytes)
{
- // Create the sound file
- std::auto_ptr<priv::SoundFile> File(priv::SoundFile::CreateRead(Data, SizeInBytes));
-
- // Open the sound file
- if (File.get())
- {
- // Get the sound parameters
- std::size_t NbSamples = File->GetSamplesCount();
- unsigned int ChannelsCount = File->GetChannelsCount();
- unsigned int SampleRate = File->GetSampleRate();
-
- // Read the samples from the opened file
- mySamples.resize(NbSamples);
- if (File->Read(&mySamples[0], NbSamples) == NbSamples)
- {
- // Update the internal buffer with the new samples
- return Update(ChannelsCount, SampleRate);
- }
- else
- {
- // Error...
- std::cerr << "Failed to read audio data from file in memory" << std::endl;
-
- return false;
- }
- }
+ priv::SoundFile file;
+ if (file.openRead(data, sizeInBytes))
+ return initialize(file);
else
- {
- // Error...
- std::cerr << "Failed to load sound buffer from file in memory" << std::endl;
-
return false;
- }
}
////////////////////////////////////////////////////////////
-/// Load the sound buffer from an array of samples - assumed format for
-/// samples is 16 bits signed integer
+bool SoundBuffer::loadFromStream(InputStream& stream)
+{
+ priv::SoundFile file;
+ if (file.openRead(stream))
+ return initialize(file);
+ else
+ return false;
+}
+
+
////////////////////////////////////////////////////////////
-bool SoundBuffer::LoadFromSamples(const Int16* Samples, std::size_t SamplesCount, unsigned int ChannelsCount, unsigned int SampleRate)
+bool SoundBuffer::loadFromSamples(const Int16* samples, std::size_t sampleCount, unsigned int channelCount, unsigned int sampleRate)
{
- if (Samples && SamplesCount && ChannelsCount && SampleRate)
+ if (samples && sampleCount && channelCount && sampleRate)
{
// Copy the new audio samples
- mySamples.assign(Samples, Samples + SamplesCount);
+ m_samples.assign(samples, samples + sampleCount);
// Update the internal buffer with the new samples
- return Update(ChannelsCount, SampleRate);
+ return update(channelCount, sampleRate);
}
else
{
// Error...
- std::cerr << "Failed to load sound buffer from memory ("
- << "Samples : " << Samples << ", "
- << "Samples count : " << SamplesCount << ", "
- << "Channels count : " << ChannelsCount << ", "
- << "Sample rate : " << SampleRate << ")"
- << std::endl;
+ err() << "Failed to load sound buffer from samples ("
+ << "array: " << samples << ", "
+ << "count: " << sampleCount << ", "
+ << "channels: " << channelCount << ", "
+ << "samplerate: " << sampleRate << ")"
+ << std::endl;
return false;
}
@@ -194,141 +136,140 @@ bool SoundBuffer::LoadFromSamples(const Int16* Samples, std::size_t SamplesCount ////////////////////////////////////////////////////////////
-/// Save the sound buffer to a file
-////////////////////////////////////////////////////////////
-bool SoundBuffer::SaveToFile(const std::string& Filename) const
+bool SoundBuffer::saveToFile(const std::string& filename) const
{
// Create the sound file in write mode
- std::auto_ptr<priv::SoundFile> File(priv::SoundFile::CreateWrite(Filename, GetChannelsCount(), GetSampleRate()));
- if (File.get())
+ priv::SoundFile file;
+ if (file.openWrite(filename, getChannelCount(), getSampleRate()))
{
// Write the samples to the opened file
- File->Write(&mySamples[0], mySamples.size());
+ file.write(&m_samples[0], m_samples.size());
return true;
}
else
{
- // Error...
- std::cerr << "Failed to save sound buffer to file \"" << Filename << "\"" << std::endl;
-
return false;
}
}
////////////////////////////////////////////////////////////
-/// Return the sound samples
-////////////////////////////////////////////////////////////
-const Int16* SoundBuffer::GetSamples() const
+const Int16* SoundBuffer::getSamples() const
{
- return mySamples.empty() ? NULL : &mySamples[0];
+ return m_samples.empty() ? NULL : &m_samples[0];
}
////////////////////////////////////////////////////////////
-/// Return the samples count
-////////////////////////////////////////////////////////////
-std::size_t SoundBuffer::GetSamplesCount() const
+std::size_t SoundBuffer::getSampleCount() const
{
- return mySamples.size();
+ return m_samples.size();
}
////////////////////////////////////////////////////////////
-/// Get the sample rate
-////////////////////////////////////////////////////////////
-unsigned int SoundBuffer::GetSampleRate() const
+unsigned int SoundBuffer::getSampleRate() const
{
- ALint SampleRate;
- ALCheck(alGetBufferi(myBuffer, AL_FREQUENCY, &SampleRate));
+ ALint sampleRate;
+ alCheck(alGetBufferi(m_buffer, AL_FREQUENCY, &sampleRate));
- return SampleRate;
+ return sampleRate;
}
////////////////////////////////////////////////////////////
-/// Return the number of channels (1 = mono, 2 = stereo, ...)
-////////////////////////////////////////////////////////////
-unsigned int SoundBuffer::GetChannelsCount() const
+unsigned int SoundBuffer::getChannelCount() const
{
- ALint ChannelsCount;
- ALCheck(alGetBufferi(myBuffer, AL_CHANNELS, &ChannelsCount));
+ ALint channelCount;
+ alCheck(alGetBufferi(m_buffer, AL_CHANNELS, &channelCount));
- return ChannelsCount;
+ return channelCount;
}
////////////////////////////////////////////////////////////
-/// Get the sound duration
-////////////////////////////////////////////////////////////
-float SoundBuffer::GetDuration() const
+Time SoundBuffer::getDuration() const
{
- return myDuration;
+ return m_duration;
}
////////////////////////////////////////////////////////////
-/// Assignment operator
-////////////////////////////////////////////////////////////
-SoundBuffer& SoundBuffer::operator =(const SoundBuffer& Other)
+SoundBuffer& SoundBuffer::operator =(const SoundBuffer& right)
{
- SoundBuffer Temp(Other);
+ SoundBuffer temp(right);
- std::swap(mySamples, Temp.mySamples);
- std::swap(myBuffer, Temp.myBuffer);
- std::swap(myDuration, Temp.myDuration);
- std::swap(mySounds, Temp.mySounds); // swap sounds too, so that they are detached when Temp is destroyed
+ std::swap(m_samples, temp.m_samples);
+ std::swap(m_buffer, temp.m_buffer);
+ std::swap(m_duration, temp.m_duration);
+ std::swap(m_sounds, temp.m_sounds); // swap sounds too, so that they are detached when temp is destroyed
return *this;
}
////////////////////////////////////////////////////////////
-/// Update the internal buffer with the audio samples
+bool SoundBuffer::initialize(priv::SoundFile& file)
+{
+ // Retrieve the sound parameters
+ std::size_t sampleCount = file.getSampleCount();
+ unsigned int channelCount = file.getChannelCount();
+ unsigned int sampleRate = file.getSampleRate();
+
+ // Read the samples from the provided file
+ m_samples.resize(sampleCount);
+ if (file.read(&m_samples[0], sampleCount) == sampleCount)
+ {
+ // Update the internal buffer with the new samples
+ return update(channelCount, sampleRate);
+ }
+ else
+ {
+ return false;
+ }
+}
+
+
////////////////////////////////////////////////////////////
-bool SoundBuffer::Update(unsigned int ChannelsCount, unsigned int SampleRate)
+bool SoundBuffer::update(unsigned int channelCount, unsigned int sampleRate)
{
// Check parameters
- if (!SampleRate || !ChannelsCount || mySamples.empty())
+ if (!channelCount || !sampleRate || m_samples.empty())
return false;
// Find the good format according to the number of channels
- ALenum Format = priv::AudioDevice::GetInstance().GetFormatFromChannelsCount(ChannelsCount);
+ ALenum format = priv::AudioDevice::getFormatFromChannelCount(channelCount);
// Check if the format is valid
- if (Format == 0)
+ if (format == 0)
{
- std::cerr << "Unsupported number of channels (" << ChannelsCount << ")" << std::endl;
+ err() << "Failed to load sound buffer (unsupported number of channels: " << channelCount << ")" << std::endl;
return false;
}
// Fill the buffer
- ALsizei Size = static_cast<ALsizei>(mySamples.size()) * sizeof(Int16);
- ALCheck(alBufferData(myBuffer, Format, &mySamples[0], Size, SampleRate));
+ ALsizei size = static_cast<ALsizei>(m_samples.size()) * sizeof(Int16);
+ alCheck(alBufferData(m_buffer, format, &m_samples[0], size, sampleRate));
// Compute the duration
- myDuration = static_cast<float>(mySamples.size()) / SampleRate / ChannelsCount;
+ m_duration = milliseconds(1000 * m_samples.size() / sampleRate / channelCount);
return true;
}
////////////////////////////////////////////////////////////
-/// Add a sound to the list of sounds that use this buffer
-////////////////////////////////////////////////////////////
-void SoundBuffer::AttachSound(Sound* Instance) const
+void SoundBuffer::attachSound(Sound* sound) const
{
- mySounds.insert(Instance);
+ m_sounds.insert(sound);
}
////////////////////////////////////////////////////////////
-/// Remove a sound from the list of sounds that use this buffer
-////////////////////////////////////////////////////////////
-void SoundBuffer::DetachSound(Sound* Instance) const
+void SoundBuffer::detachSound(Sound* sound) const
{
- mySounds.erase(Instance);
+ m_sounds.erase(sound);
}
} // namespace sf
diff --git a/src/SFML/Audio/SoundBufferRecorder.cpp b/src/SFML/Audio/SoundBufferRecorder.cpp index c0af7bf..a8b2993 100755..100644 --- a/src/SFML/Audio/SoundBufferRecorder.cpp +++ b/src/SFML/Audio/SoundBufferRecorder.cpp @@ -1,7 +1,7 @@ ////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
@@ -33,43 +33,36 @@ namespace sf
{
////////////////////////////////////////////////////////////
-/// /see SoundBuffer::OnStart
-////////////////////////////////////////////////////////////
-bool SoundBufferRecorder::OnStart()
+bool SoundBufferRecorder::onStart()
{
- mySamples.clear();
+ m_samples.clear();
+ m_buffer = SoundBuffer();
return true;
}
////////////////////////////////////////////////////////////
-/// /see SoundBuffer::OnProcessSamples
-////////////////////////////////////////////////////////////
-bool SoundBufferRecorder::OnProcessSamples(const Int16* Samples, std::size_t SamplesCount)
+bool SoundBufferRecorder::onProcessSamples(const Int16* samples, std::size_t sampleCount)
{
- std::copy(Samples, Samples + SamplesCount, std::back_inserter(mySamples));
+ std::copy(samples, samples + sampleCount, std::back_inserter(m_samples));
return true;
}
////////////////////////////////////////////////////////////
-/// /see SoundBuffer::OnStop
-////////////////////////////////////////////////////////////
-void SoundBufferRecorder::OnStop()
+void SoundBufferRecorder::onStop()
{
- if (!mySamples.empty())
- myBuffer.LoadFromSamples(&mySamples[0], mySamples.size(), 1, GetSampleRate());
+ if (!m_samples.empty())
+ m_buffer.loadFromSamples(&m_samples[0], m_samples.size(), 1, getSampleRate());
}
////////////////////////////////////////////////////////////
-/// Get the sound buffer containing the captured audio data
-////////////////////////////////////////////////////////////
-const SoundBuffer& SoundBufferRecorder::GetBuffer() const
+const SoundBuffer& SoundBufferRecorder::getBuffer() const
{
- return myBuffer;
+ return m_buffer;
}
} // namespace sf
diff --git a/src/SFML/Audio/SoundFile.cpp b/src/SFML/Audio/SoundFile.cpp index 0554d8b..91db819 100755..100644 --- a/src/SFML/Audio/SoundFile.cpp +++ b/src/SFML/Audio/SoundFile.cpp @@ -1,7 +1,7 @@ ////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
@@ -26,9 +26,22 @@ // Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundFile.hpp>
-#include <SFML/Audio/SoundFileDefault.hpp>
-#include <SFML/Audio/SoundFileOgg.hpp>
-#include <iostream>
+#include <SFML/System/InputStream.hpp>
+#include <SFML/System/Err.hpp>
+#include <cstring>
+#include <cctype>
+
+
+namespace
+{
+ // Convert a string to lower case
+ std::string toLower(std::string str)
+ {
+ for (std::string::iterator i = str.begin(); i != str.end(); ++i)
+ *i = static_cast<char>(std::tolower(*i));
+ return str;
+ }
+}
namespace sf
@@ -36,234 +49,375 @@ namespace sf namespace priv
{
////////////////////////////////////////////////////////////
-/// Create a new sound from a file, for reading
-////////////////////////////////////////////////////////////
-SoundFile* SoundFile::CreateRead(const std::string& Filename)
+SoundFile::SoundFile() :
+m_file (NULL),
+m_sampleCount (0),
+m_channelCount(0),
+m_sampleRate (0)
{
- // Create the file according to its type
- SoundFile* File = NULL;
- if (SoundFileOgg::IsFileSupported(Filename, true)) File = new SoundFileOgg;
- else if (SoundFileDefault::IsFileSupported(Filename, true)) File = new SoundFileDefault;
- // Open it for reading
- if (File)
- {
- std::size_t SamplesCount;
- unsigned int ChannelsCount;
- unsigned int SampleRate;
+}
- if (File->OpenRead(Filename, SamplesCount, ChannelsCount, SampleRate))
- {
- File->myFilename = Filename;
- File->myData = NULL;
- File->mySize = 0;
- File->myNbSamples = SamplesCount;
- File->myChannelsCount = ChannelsCount;
- File->mySampleRate = SampleRate;
- }
- else
- {
- delete File;
- File = NULL;
- }
- }
- return File;
+////////////////////////////////////////////////////////////
+SoundFile::~SoundFile()
+{
+ if (m_file)
+ sf_close(m_file);
}
////////////////////////////////////////////////////////////
-/// Create a new sound from a file in memory, for reading
-////////////////////////////////////////////////////////////
-SoundFile* SoundFile::CreateRead(const char* Data, std::size_t SizeInMemory)
+std::size_t SoundFile::getSampleCount() const
{
- // Create the file according to its type
- SoundFile* File = NULL;
- if (SoundFileOgg::IsFileSupported(Data, SizeInMemory)) File = new SoundFileOgg;
- else if (SoundFileDefault::IsFileSupported(Data, SizeInMemory)) File = new SoundFileDefault;
-
- // Open it for reading
- if (File)
- {
- std::size_t SamplesCount;
- unsigned int ChannelsCount;
- unsigned int SampleRate;
+ return m_sampleCount;
+}
- if (File->OpenRead(Data, SizeInMemory, SamplesCount, ChannelsCount, SampleRate))
- {
- File->myFilename = "";
- File->myData = Data;
- File->mySize = SizeInMemory;
- File->myNbSamples = SamplesCount;
- File->myChannelsCount = ChannelsCount;
- File->mySampleRate = SampleRate;
- }
- else
- {
- delete File;
- File = NULL;
- }
- }
- return File;
+////////////////////////////////////////////////////////////
+unsigned int SoundFile::getChannelCount() const
+{
+ return m_channelCount;
}
////////////////////////////////////////////////////////////
-/// Create a new sound from a file, for writing
-////////////////////////////////////////////////////////////
-SoundFile* SoundFile::CreateWrite(const std::string& Filename, unsigned int ChannelsCount, unsigned int SampleRate)
+unsigned int SoundFile::getSampleRate() const
{
- // Create the file according to its type
- SoundFile* File = NULL;
- if (SoundFileOgg::IsFileSupported(Filename, false)) File = new SoundFileOgg;
- else if (SoundFileDefault::IsFileSupported(Filename, false)) File = new SoundFileDefault;
+ return m_sampleRate;
+}
+
- // Open it for writing
- if (File)
+////////////////////////////////////////////////////////////
+bool SoundFile::openRead(const std::string& filename)
+{
+ // If the file is already opened, first close it
+ if (m_file)
+ sf_close(m_file);
+
+ // Open the sound file
+ SF_INFO fileInfo;
+ fileInfo.format = 0;
+ m_file = sf_open(filename.c_str(), SFM_READ, &fileInfo);
+ if (!m_file)
{
- if (File->OpenWrite(Filename, ChannelsCount, SampleRate))
- {
- File->myFilename = "";
- File->myData = NULL;
- File->mySize = 0;
- File->myNbSamples = 0;
- File->myChannelsCount = ChannelsCount;
- File->mySampleRate = SampleRate;
- }
- else
- {
- delete File;
- File = NULL;
- }
+ err() << "Failed to open sound file \"" << filename << "\" (" << sf_strerror(m_file) << ")" << std::endl;
+ return false;
}
- return File;
+ // Initialize the internal state from the loaded information
+ initialize(fileInfo);
+
+ return true;
}
////////////////////////////////////////////////////////////
-/// Default constructor
-////////////////////////////////////////////////////////////
-SoundFile::SoundFile() :
-myNbSamples (0),
-myChannelsCount(0),
-mySampleRate (0)
+bool SoundFile::openRead(const void* data, std::size_t sizeInBytes)
{
+ // If the file is already opened, first close it
+ if (m_file)
+ sf_close(m_file);
+
+ // Prepare the memory I/O structure
+ SF_VIRTUAL_IO io;
+ io.get_filelen = &Memory::getLength;
+ io.read = &Memory::read;
+ io.seek = &Memory::seek;
+ io.tell = &Memory::tell;
+
+ // Initialize the memory data
+ m_memory.begin = static_cast<const char*>(data);
+ m_memory.current = m_memory.begin;
+ m_memory.size = sizeInBytes;
+
+ // Open the sound file
+ SF_INFO fileInfo;
+ fileInfo.format = 0;
+ m_file = sf_open_virtual(&io, SFM_READ, &fileInfo, &m_memory);
+ if (!m_file)
+ {
+ err() << "Failed to open sound file from memory (" << sf_strerror(m_file) << ")" << std::endl;
+ return false;
+ }
+
+ // Initialize the internal state from the loaded information
+ initialize(fileInfo);
+ return true;
}
////////////////////////////////////////////////////////////
-/// Virtual destructor
-////////////////////////////////////////////////////////////
-SoundFile::~SoundFile()
+bool SoundFile::openRead(InputStream& stream)
{
- // Nothing to do
+ // If the file is already opened, first close it
+ if (m_file)
+ sf_close(m_file);
+
+ // Prepare the memory I/O structure
+ SF_VIRTUAL_IO io;
+ io.get_filelen = &Stream::getLength;
+ io.read = &Stream::read;
+ io.seek = &Stream::seek;
+ io.tell = &Stream::tell;
+
+ // Initialize the stream data
+ m_stream.source = &stream;
+ m_stream.size = stream.getSize();
+
+ // Make sure that the stream's reading position is at the beginning
+ stream.seek(0);
+
+ // Open the sound file
+ SF_INFO fileInfo;
+ fileInfo.format = 0;
+ m_file = sf_open_virtual(&io, SFM_READ, &fileInfo, &m_stream);
+ if (!m_file)
+ {
+ err() << "Failed to open sound file from stream (" << sf_strerror(m_file) << ")" << std::endl;
+ return false;
+ }
+
+ // Initialize the internal state from the loaded information
+ initialize(fileInfo);
+
+ return true;
}
////////////////////////////////////////////////////////////
-/// Get the total number of samples in the file
-////////////////////////////////////////////////////////////
-std::size_t SoundFile::GetSamplesCount() const
+bool SoundFile::openWrite(const std::string& filename, unsigned int channelCount, unsigned int sampleRate)
{
- return myNbSamples;
+ // If the file is already opened, first close it
+ if (m_file)
+ sf_close(m_file);
+
+ // Find the right format according to the file extension
+ int format = getFormatFromFilename(filename);
+ if (format == -1)
+ {
+ // Error : unrecognized extension
+ err() << "Failed to create sound file \"" << filename << "\" (unknown format)" << std::endl;
+ return false;
+ }
+
+ // Fill the sound infos with parameters
+ SF_INFO fileInfos;
+ fileInfos.channels = channelCount;
+ fileInfos.samplerate = sampleRate;
+ fileInfos.format = format | (format == SF_FORMAT_OGG ? SF_FORMAT_VORBIS : SF_FORMAT_PCM_16);
+
+ // Open the sound file for writing
+ m_file = sf_open(filename.c_str(), SFM_WRITE, &fileInfos);
+ if (!m_file)
+ {
+ err() << "Failed to create sound file \"" << filename << "\" (" << sf_strerror(m_file) << ")" << std::endl;
+ return false;
+ }
+
+ // Set the sound parameters
+ m_channelCount = channelCount;
+ m_sampleRate = sampleRate;
+ m_sampleCount = 0;
+
+ return true;
}
////////////////////////////////////////////////////////////
-/// Get the number of channels used by the sound
-////////////////////////////////////////////////////////////
-unsigned int SoundFile::GetChannelsCount() const
+std::size_t SoundFile::read(Int16* data, std::size_t sampleCount)
{
- return myChannelsCount;
+ if (m_file && data && sampleCount)
+ return static_cast<std::size_t>(sf_read_short(m_file, data, sampleCount));
+ else
+ return 0;
}
////////////////////////////////////////////////////////////
-/// Get the sample rate of the sound
-////////////////////////////////////////////////////////////
-unsigned int SoundFile::GetSampleRate() const
+void SoundFile::write(const Int16* data, std::size_t sampleCount)
{
- return mySampleRate;
+ if (m_file && data && sampleCount)
+ {
+ // Write small chunks instead of everything at once,
+ // to avoid a stack overflow in libsndfile (happens only with OGG format)
+ while (sampleCount > 0)
+ {
+ std::size_t count = sampleCount > 10000 ? 10000 : sampleCount;
+ sf_write_short(m_file, data, count);
+ data += count;
+ sampleCount -= count;
+ }
+ }
}
////////////////////////////////////////////////////////////
-/// Restart the sound from the beginning
-////////////////////////////////////////////////////////////
-bool SoundFile::Restart()
+void SoundFile::seek(Time timeOffset)
{
- if (myData)
- {
- // Reopen from memory
- return OpenRead(myData, mySize, myNbSamples, myChannelsCount, mySampleRate);
- }
- else if (myFilename != "")
- {
- // Reopen from file
- return OpenRead(myFilename, myNbSamples, myChannelsCount, mySampleRate);
- }
- else
+ if (m_file)
{
- // Trying to reopen a file opened in write mode... error
- std::cerr << "Warning : trying to restart a sound opened in write mode, which is not allowed" << std::endl;
- return false;
+ sf_count_t frameOffset = static_cast<sf_count_t>(timeOffset.asSeconds() * m_sampleRate);
+ sf_seek(m_file, frameOffset, SEEK_SET);
}
}
////////////////////////////////////////////////////////////
-/// Open the sound file for reading
-////////////////////////////////////////////////////////////
-bool SoundFile::OpenRead(const std::string& Filename, std::size_t&, unsigned int&, unsigned int&)
+void SoundFile::initialize(SF_INFO fileInfo)
{
- std::cerr << "Failed to open sound file \"" << Filename << "\", format is not supported by SFML" << std::endl;
+ // Save the sound properties
+ m_channelCount = fileInfo.channels;
+ m_sampleRate = fileInfo.samplerate;
+ m_sampleCount = static_cast<std::size_t>(fileInfo.frames) * fileInfo.channels;
+
+ // Enable scaling for Vorbis files (float samples)
+ // @todo enable when it's faster (it currently has to iterate over the *whole* music)
+ //if (fileInfo.format & SF_FORMAT_VORBIS)
+ // sf_command(m_file, SFC_SET_SCALE_FLOAT_INT_READ, NULL, SF_TRUE);
+}
- return false;
+
+////////////////////////////////////////////////////////////
+int SoundFile::getFormatFromFilename(const std::string& filename)
+{
+ // Extract the extension
+ std::string ext = "wav";
+ std::string::size_type pos = filename.find_last_of(".");
+ if (pos != std::string::npos)
+ ext = filename.substr(pos + 1);
+
+ // Match every supported extension with its format constant
+ if (toLower(ext) == "wav" ) return SF_FORMAT_WAV;
+ if (toLower(ext) == "aif" ) return SF_FORMAT_AIFF;
+ if (toLower(ext) == "aiff" ) return SF_FORMAT_AIFF;
+ if (toLower(ext) == "au" ) return SF_FORMAT_AU;
+ if (toLower(ext) == "raw" ) return SF_FORMAT_RAW;
+ if (toLower(ext) == "paf" ) return SF_FORMAT_PAF;
+ if (toLower(ext) == "svx" ) return SF_FORMAT_SVX;
+ if (toLower(ext) == "nist" ) return SF_FORMAT_NIST;
+ if (toLower(ext) == "voc" ) return SF_FORMAT_VOC;
+ if (toLower(ext) == "sf" ) return SF_FORMAT_IRCAM;
+ if (toLower(ext) == "w64" ) return SF_FORMAT_W64;
+ if (toLower(ext) == "mat4" ) return SF_FORMAT_MAT4;
+ if (toLower(ext) == "mat5" ) return SF_FORMAT_MAT5;
+ if (toLower(ext) == "pvf" ) return SF_FORMAT_PVF;
+ if (toLower(ext) == "xi" ) return SF_FORMAT_XI;
+ if (toLower(ext) == "htk" ) return SF_FORMAT_HTK;
+ if (toLower(ext) == "sds" ) return SF_FORMAT_SDS;
+ if (toLower(ext) == "avr" ) return SF_FORMAT_AVR;
+ if (toLower(ext) == "sd2" ) return SF_FORMAT_SD2;
+ if (toLower(ext) == "flac" ) return SF_FORMAT_FLAC;
+ if (toLower(ext) == "caf" ) return SF_FORMAT_CAF;
+ if (toLower(ext) == "wve" ) return SF_FORMAT_WVE;
+ if (toLower(ext) == "ogg" ) return SF_FORMAT_OGG;
+ if (toLower(ext) == "mpc2k") return SF_FORMAT_MPC2K;
+ if (toLower(ext) == "rf64" ) return SF_FORMAT_RF64;
+
+ return -1;
}
////////////////////////////////////////////////////////////
-/// Open the sound file in memory for reading
+sf_count_t SoundFile::Memory::getLength(void* user)
+{
+ Memory* memory = static_cast<Memory*>(user);
+ return memory->size;
+}
+
+
////////////////////////////////////////////////////////////
-bool SoundFile::OpenRead(const char*, std::size_t, std::size_t&, unsigned int&, unsigned int&)
+sf_count_t SoundFile::Memory::read(void* ptr, sf_count_t count, void* user)
{
- std::cerr << "Failed to open sound file from memory, format is not supported by SFML" << std::endl;
+ Memory* memory = static_cast<Memory*>(user);
- return false;
+ sf_count_t position = tell(user);
+ if (position + count >= memory->size)
+ count = memory->size - position;
+
+ std::memcpy(ptr, memory->current, static_cast<std::size_t>(count));
+ memory->current += count;
+ return count;
}
////////////////////////////////////////////////////////////
-/// Open the sound file for writing
-////////////////////////////////////////////////////////////
-bool SoundFile::OpenWrite(const std::string& Filename, unsigned int, unsigned int)
+sf_count_t SoundFile::Memory::seek(sf_count_t offset, int whence, void* user)
{
- std::cerr << "Failed to open sound file \"" << Filename << "\", format is not supported by SFML" << std::endl;
+ Memory* memory = static_cast<Memory*>(user);
+ sf_count_t position = 0;
+ switch (whence)
+ {
+ case SEEK_SET : position = offset; break;
+ case SEEK_CUR : position = memory->current - memory->begin + offset; break;
+ case SEEK_END : position = memory->size - offset; break;
+ default : position = 0; break;
+ }
+
+ if (position >= memory->size)
+ position = memory->size - 1;
+ else if (position < 0)
+ position = 0;
- return false;
+ memory->current = memory->begin + position;
+ return position;
}
////////////////////////////////////////////////////////////
-/// Read samples from the loaded sound
+sf_count_t SoundFile::Memory::tell(void* user)
+{
+ Memory* memory = static_cast<Memory*>(user);
+ return memory->current - memory->begin;
+}
+
+
////////////////////////////////////////////////////////////
-std::size_t SoundFile::Read(Int16*, std::size_t)
+sf_count_t SoundFile::Stream::getLength(void* userData)
{
- std::cerr << "Failed to read from sound file (not supported)" << std::endl;
+ Stream* stream = static_cast<Stream*>(userData);
+ return stream->size;
+}
- return 0;
+
+////////////////////////////////////////////////////////////
+sf_count_t SoundFile::Stream::read(void* ptr, sf_count_t count, void* userData)
+{
+ Stream* stream = static_cast<Stream*>(userData);
+ Int64 position = stream->source->tell();
+ if (position != -1)
+ {
+ if (count > stream->size - position)
+ count = stream->size - position;
+ return stream->source->read(reinterpret_cast<char*>(ptr), count);
+ }
+ else
+ {
+ return -1;
+ }
}
////////////////////////////////////////////////////////////
-/// Write samples to the file
+sf_count_t SoundFile::Stream::seek(sf_count_t offset, int whence, void* userData)
+{
+ Stream* stream = static_cast<Stream*>(userData);
+ switch (whence)
+ {
+ case SEEK_SET : return stream->source->seek(offset);
+ case SEEK_CUR : return stream->source->seek(stream->source->tell() + offset);
+ case SEEK_END : return stream->source->seek(stream->size - offset);
+ default : return stream->source->seek(0);
+ }
+}
+
+
////////////////////////////////////////////////////////////
-void SoundFile::Write(const Int16*, std::size_t)
+sf_count_t SoundFile::Stream::tell(void* userData)
{
- std::cerr << "Failed to write to sound file (not supported)" << std::endl;
+ Stream* stream = static_cast<Stream*>(userData);
+ return stream->source->tell();
}
} // namespace priv
diff --git a/src/SFML/Audio/SoundFile.hpp b/src/SFML/Audio/SoundFile.hpp index fb36bdd..105a951 100755..100644 --- a/src/SFML/Audio/SoundFile.hpp +++ b/src/SFML/Audio/SoundFile.hpp @@ -1,7 +1,7 @@ ////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
@@ -29,176 +29,195 @@ // Headers
////////////////////////////////////////////////////////////
#include <SFML/System/NonCopyable.hpp>
+#include <SFML/System/Time.hpp>
+#include <sndfile.h>
#include <string>
namespace sf
{
+class InputStream;
+
namespace priv
{
////////////////////////////////////////////////////////////
-/// SoundFile is the abstract base class for loading
-/// and saving different sound file formats
+/// \brief Provide read and write access to sound files
+///
////////////////////////////////////////////////////////////
class SoundFile : NonCopyable
{
public :
////////////////////////////////////////////////////////////
- /// Create a new sound from a file, for reading
- ///
- /// \param Filename : Path of sound file
- /// \param NbSamples : Number of samples in the file
- /// \param ChannelsCount : Number of channels in the loaded sound
- /// \param SampleRate : Sample rate of the loaded sound
- ///
- /// \return Pointer to the new sound file (NULL if failed)
+ /// \brief Default constructor
///
////////////////////////////////////////////////////////////
- static SoundFile* CreateRead(const std::string& Filename);
+ SoundFile();
////////////////////////////////////////////////////////////
- /// Create a new sound from a file in memory, for reading
+ /// \brief Destructor
///
- /// \param Data : Pointer to the file data in memory
- /// \param SizeInBytes : Size of the data to load, in bytes
- /// \param NbSamples : Number of samples in the file
- /// \param ChannelsCount : Number of channels in the loaded sound
- /// \param SampleRate : Sample rate of the loaded sound
+ ////////////////////////////////////////////////////////////
+ ~SoundFile();
+
+ ////////////////////////////////////////////////////////////
+ /// \brief Get the total number of audio samples in the file
///
- /// \return Pointer to the new sound file (NULL if failed)
+ /// \return Number of samples
///
////////////////////////////////////////////////////////////
- static SoundFile* CreateRead(const char* Data, std::size_t SizeInBytes);
+ std::size_t getSampleCount() const;
////////////////////////////////////////////////////////////
- /// Create a new sound from a file, for writing
- ///
- /// \param Filename : Path of sound file
- /// \param ChannelsCount : Number of channels in the sound
- /// \param SampleRate : Sample rate of the sound
+ /// \brief Get the number of channels used by the sound
///
- /// \return Pointer to the new sound file (NULL if failed)
+ /// \return Number of channels (1 = mono, 2 = stereo)
///
////////////////////////////////////////////////////////////
- static SoundFile* CreateWrite(const std::string& Filename, unsigned int ChannelsCount, unsigned int SampleRate);
+ unsigned int getChannelCount() const;
////////////////////////////////////////////////////////////
- /// Virtual destructor
+ /// \brief Get the sample rate of the sound
+ ///
+ /// \return Sample rate, in samples per second
///
////////////////////////////////////////////////////////////
- virtual ~SoundFile();
+ unsigned int getSampleRate() const;
////////////////////////////////////////////////////////////
- /// Get the total number of samples in the file
+ /// \brief Open a sound file for reading
///
- /// \return Number of samples
+ /// \param filename Path of the sound file to load
+ ///
+ /// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
- std::size_t GetSamplesCount() const;
+ bool openRead(const std::string& filename);
////////////////////////////////////////////////////////////
- /// Get the number of channels used by the sound
+ /// \brief Open a sound file in memory for reading
///
- /// \return Number of channels (1 = mono, 2 = stereo)
+ /// \param data Pointer to the file data in memory
+ /// \param sizeInBytes Size of the data to load, in bytes
+ ///
+ /// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
- unsigned int GetChannelsCount() const;
+ bool openRead(const void* data, std::size_t sizeInBytes);
////////////////////////////////////////////////////////////
- /// Get the sample rate of the sound
+ /// \brief Open a sound file from a custom stream for reading
///
- /// \return Sample rate, in samples / sec
+ /// \param stream Source stream to read from
+ ///
+ /// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
- unsigned int GetSampleRate() const;
+ bool openRead(InputStream& stream);
////////////////////////////////////////////////////////////
- /// Restart the sound from the beginning
+ /// \brief a the sound file for writing
///
- /// \return True if restart was successful
+ /// \param filename Path of the sound file to write
+ /// \param channelCount Number of channels in the sound
+ /// \param sampleRate Sample rate of the sound
+ ///
+ /// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
- bool Restart();
+ bool openWrite(const std::string& filename, unsigned int channelCount, unsigned int sampleRate);
////////////////////////////////////////////////////////////
- /// Read samples from the loaded sound
+ /// \brief Read audio samples from the loaded sound
///
- /// \param Data : Pointer to the samples array to fill
- /// \param NbSamples : Number of samples to read
+ /// \param data Pointer to the sample array to fill
+ /// \param sampleCount Number of samples to read
///
- /// \return Number of samples read
+ /// \return Number of samples actually read (may be less than \a sampleCount)
///
////////////////////////////////////////////////////////////
- virtual std::size_t Read(Int16* Data, std::size_t NbSamples);
+ std::size_t read(Int16* data, std::size_t sampleCount);
////////////////////////////////////////////////////////////
- /// Write samples to the file
+ /// \brief Write audio samples to the file
///
- /// \param Data : Pointer to the samples array to write
- /// \param NbSamples : Number of samples to write
+ /// \param data Pointer to the sample array to write
+ /// \param sampleCount Number of samples to write
///
////////////////////////////////////////////////////////////
- virtual void Write(const Int16* Data, std::size_t NbSamples);
-
-protected :
+ void write(const Int16* data, std::size_t sampleCount);
////////////////////////////////////////////////////////////
- /// Default constructor
+ /// \brief Change the current read position in the file
+ ///
+ /// \param timeOffset New playing position, from the beginning of the file
///
////////////////////////////////////////////////////////////
- SoundFile();
+ void seek(Time timeOffset);
private :
////////////////////////////////////////////////////////////
- /// Open the sound file for reading
+ /// \brief Initialize the internal state of the sound file
///
- /// \param Filename : Path of sound file to load
- /// \param NbSamples : Number of samples in the file
- /// \param ChannelsCount : Number of channels in the loaded sound
- /// \param SampleRate : Sample rate of the loaded sound
+ /// This function is called by all the openRead functions.
///
- /// \return True if the file was successfully opened
+ /// \param fileInfo Information about the loaded sound file
///
////////////////////////////////////////////////////////////
- virtual bool OpenRead(const std::string& Filename, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate);
+ void initialize(SF_INFO fileInfo);
////////////////////////////////////////////////////////////
- /// Open the sound file in memory for reading
+ /// \brief Get the internal format of an audio file according to
+ /// its filename extension
///
- /// \param Data : Pointer to the file data in memory
- /// \param SizeInBytes : Size of the data to load, in bytes
- /// \param NbSamples : Number of samples in the file
- /// \param ChannelsCount : Number of channels in the loaded sound
- /// \param SampleRate : Sample rate of the loaded sound
+ /// \param filename Filename to check
///
- /// \return True if the file was successfully opened
+ /// \return Internal format matching the filename (-1 if no match)
///
////////////////////////////////////////////////////////////
- virtual bool OpenRead(const char* Data, std::size_t SizeInBytes, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate);
+ static int getFormatFromFilename(const std::string& filename);
////////////////////////////////////////////////////////////
- /// Open the sound file for writing
- ///
- /// \param Filename : Path of sound file to write
- /// \param ChannelsCount : Number of channels in the sound
- /// \param SampleRate : Sample rate of the sound
+ /// \brief Data and callbacks for opening from memory
///
- /// \return True if the file was successfully opened
+ ////////////////////////////////////////////////////////////
+ struct Memory
+ {
+ const char* begin;
+ const char* current;
+ sf_count_t size;
+
+ static sf_count_t getLength(void* user);
+ static sf_count_t read(void* ptr, sf_count_t count, void* user);
+ static sf_count_t seek(sf_count_t offset, int whence, void* user);
+ static sf_count_t tell(void* user);
+ };
+
+ ////////////////////////////////////////////////////////////
+ /// \brief Data and callbacks for opening from stream
///
////////////////////////////////////////////////////////////
- virtual bool OpenWrite(const std::string& Filename, unsigned int ChannelsCount, unsigned int SampleRate);
+ struct Stream
+ {
+ InputStream* source;
+ Int64 size;
+
+ static sf_count_t getLength(void* user);
+ static sf_count_t read(void* ptr, sf_count_t count, void* user);
+ static sf_count_t seek(sf_count_t offset, int whence, void* user);
+ static sf_count_t tell(void* user);
+ };
////////////////////////////////////////////////////////////
// Member data
////////////////////////////////////////////////////////////
- std::size_t myNbSamples; ///< Total number of samples in the file
- unsigned int myChannelsCount; ///< Number of channels used by the sound
- unsigned int mySampleRate; ///< Number of samples per second
- std::string myFilename; ///< Path of the file (valid if loaded from a file)
- const char* myData; ///< Pointer to the file in memory (valid if loaded from memory)
- std::size_t mySize; ///< Size of the file in memory (valid if loaded from memory)
+ SNDFILE* m_file; ///< File descriptor
+ Memory m_memory; ///< Memory reading info
+ Stream m_stream; ///< Stream reading info
+ std::size_t m_sampleCount; ///< Total number of samples in the file
+ unsigned int m_channelCount; ///< Number of channels used by the sound
+ unsigned int m_sampleRate; ///< Number of samples per second
};
} // namespace priv
diff --git a/src/SFML/Audio/SoundFileDefault.cpp b/src/SFML/Audio/SoundFileDefault.cpp deleted file mode 100755 index 4413f6b..0000000 --- a/src/SFML/Audio/SoundFileDefault.cpp +++ /dev/null @@ -1,352 +0,0 @@ -////////////////////////////////////////////////////////////
-//
-// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
-//
-// This software is provided 'as-is', without any express or implied warranty.
-// In no event will the authors be held liable for any damages arising from the use of this software.
-//
-// Permission is granted to anyone to use this software for any purpose,
-// including commercial applications, and to alter it and redistribute it freely,
-// subject to the following restrictions:
-//
-// 1. The origin of this software must not be misrepresented;
-// you must not claim that you wrote the original software.
-// If you use this software in a product, an acknowledgment
-// in the product documentation would be appreciated but is not required.
-//
-// 2. Altered source versions must be plainly marked as such,
-// and must not be misrepresented as being the original software.
-//
-// 3. This notice may not be removed or altered from any source distribution.
-//
-////////////////////////////////////////////////////////////
-
-////////////////////////////////////////////////////////////
-// Headers
-////////////////////////////////////////////////////////////
-#include <SFML/Audio/SoundFileDefault.hpp>
-#include <iostream>
-#include <string.h>
-
-
-namespace sf
-{
-namespace priv
-{
-////////////////////////////////////////////////////////////
-/// Default constructor
-////////////////////////////////////////////////////////////
-SoundFileDefault::SoundFileDefault() :
-myFile(NULL)
-{
-
-}
-
-
-////////////////////////////////////////////////////////////
-/// Destructor
-////////////////////////////////////////////////////////////
-SoundFileDefault::~SoundFileDefault()
-{
- if (myFile)
- sf_close(myFile);
-}
-
-
-////////////////////////////////////////////////////////////
-/// Check if a given file is supported by this loader
-////////////////////////////////////////////////////////////
-bool SoundFileDefault::IsFileSupported(const std::string& Filename, bool Read)
-{
- if (Read)
- {
- // Open the sound file
- SF_INFO FileInfos;
- SNDFILE* File = sf_open(Filename.c_str(), SFM_READ, &FileInfos);
-
- if (File)
- {
- sf_close(File);
- return true;
- }
- else
- {
- return false;
- }
- }
- else
- {
- // Check the extension
- return GetFormatFromFilename(Filename) != -1;
- }
-}
-
-
-////////////////////////////////////////////////////////////
-/// Check if a given file in memory is supported by this loader
-////////////////////////////////////////////////////////////
-bool SoundFileDefault::IsFileSupported(const char* Data, std::size_t SizeInBytes)
-{
- // Define the I/O custom functions for reading from memory
- SF_VIRTUAL_IO VirtualIO;
- VirtualIO.get_filelen = &SoundFileDefault::MemoryGetLength;
- VirtualIO.read = &SoundFileDefault::MemoryRead;
- VirtualIO.seek = &SoundFileDefault::MemorySeek;
- VirtualIO.tell = &SoundFileDefault::MemoryTell;
- VirtualIO.write = &SoundFileDefault::MemoryWrite;
-
- // Initialize the memory data
- MemoryInfos Memory;
- Memory.DataStart = Data;
- Memory.DataPtr = Data;
- Memory.TotalSize = SizeInBytes;
-
- // Open the sound file
- SF_INFO FileInfos;
- SNDFILE* File = sf_open_virtual(&VirtualIO, SFM_READ, &FileInfos, &Memory);
-
- if (File)
- {
- sf_close(File);
- return true;
- }
- else
- {
- return false;
- }
-}
-
-
-////////////////////////////////////////////////////////////
-/// Open the sound file for reading
-////////////////////////////////////////////////////////////
-bool SoundFileDefault::OpenRead(const std::string& Filename, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate)
-{
- // If the file is already opened, first close it
- if (myFile)
- sf_close(myFile);
-
- // Open the sound file
- SF_INFO FileInfos;
- myFile = sf_open(Filename.c_str(), SFM_READ, &FileInfos);
- if (!myFile)
- {
- std::cerr << "Failed to read sound file \"" << Filename << "\"" << std::endl;
- return false;
- }
-
- // Set the sound parameters
- ChannelsCount = FileInfos.channels;
- SampleRate = FileInfos.samplerate;
- NbSamples = static_cast<std::size_t>(FileInfos.frames) * ChannelsCount;
-
- return true;
-}
-
-
-////////////////////////////////////////////////////////////
-/// /see sf::SoundFile::OpenRead
-////////////////////////////////////////////////////////////
-bool SoundFileDefault::OpenRead(const char* Data, std::size_t SizeInBytes, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate)
-{
- // If the file is already opened, first close it
- if (myFile)
- sf_close(myFile);
-
- // Define the I/O custom functions for reading from memory
- SF_VIRTUAL_IO VirtualIO;
- VirtualIO.get_filelen = &SoundFileDefault::MemoryGetLength;
- VirtualIO.read = &SoundFileDefault::MemoryRead;
- VirtualIO.seek = &SoundFileDefault::MemorySeek;
- VirtualIO.tell = &SoundFileDefault::MemoryTell;
- VirtualIO.write = &SoundFileDefault::MemoryWrite;
-
- // Initialize the memory data
- myMemory.DataStart = Data;
- myMemory.DataPtr = Data;
- myMemory.TotalSize = SizeInBytes;
-
- // Open the sound file
- SF_INFO FileInfos;
- myFile = sf_open_virtual(&VirtualIO, SFM_READ, &FileInfos, &myMemory);
- if (!myFile)
- {
- std::cerr << "Failed to read sound file from memory" << std::endl;
- return false;
- }
-
- // Set the sound parameters
- ChannelsCount = FileInfos.channels;
- SampleRate = FileInfos.samplerate;
- NbSamples = static_cast<std::size_t>(FileInfos.frames) * ChannelsCount;
-
- return true;
-}
-
-
-////////////////////////////////////////////////////////////
-/// Open the sound file for writing
-////////////////////////////////////////////////////////////
-bool SoundFileDefault::OpenWrite(const std::string& Filename, unsigned int ChannelsCount, unsigned int SampleRate)
-{
- // If the file is already opened, first close it
- if (myFile)
- sf_close(myFile);
-
- // Find the right format according to the file extension
- int Format = GetFormatFromFilename(Filename);
- if (Format == -1)
- {
- // Error : unrecognized extension
- std::cerr << "Failed to create sound file \"" << Filename << "\" : unknown format" << std::endl;
- return false;
- }
-
- // Fill the sound infos with parameters
- SF_INFO FileInfos;
- FileInfos.channels = ChannelsCount;
- FileInfos.samplerate = SampleRate;
- FileInfos.format = Format | SF_FORMAT_PCM_16;
-
- // Open the sound file for writing
- myFile = sf_open(Filename.c_str(), SFM_WRITE, &FileInfos);
- if (!myFile)
- {
- std::cerr << "Failed to create sound file \"" << Filename << "\"" << std::endl;
- return false;
- }
-
- return true;
-}
-
-
-////////////////////////////////////////////////////////////
-/// Read samples from the loaded sound
-////////////////////////////////////////////////////////////
-std::size_t SoundFileDefault::Read(Int16* Data, std::size_t NbSamples)
-{
- if (myFile && Data && NbSamples)
- return static_cast<std::size_t>(sf_read_short(myFile, Data, NbSamples));
- else
- return 0;
-}
-
-
-////////////////////////////////////////////////////////////
-/// Write samples to the file
-////////////////////////////////////////////////////////////
-void SoundFileDefault::Write(const Int16* Data, std::size_t NbSamples)
-{
- if (myFile && Data && NbSamples)
- sf_write_short(myFile, Data, NbSamples);
-}
-
-
-////////////////////////////////////////////////////////////
-/// Get the internal format of an audio file according to
-/// its filename extension
-////////////////////////////////////////////////////////////
-int SoundFileDefault::GetFormatFromFilename(const std::string& Filename)
-{
- // Extract the extension
- std::string Ext = "wav";
- std::string::size_type Pos = Filename.find_last_of(".");
- if (Pos != std::string::npos)
- Ext = Filename.substr(Pos + 1);
-
- // Match every supported extension with its format constant
- if (Ext == "wav" || Ext == "WAV" ) return SF_FORMAT_WAV;
- if (Ext == "aif" || Ext == "AIF" ) return SF_FORMAT_AIFF;
- if (Ext == "aiff" || Ext == "AIFF") return SF_FORMAT_AIFF;
- if (Ext == "au" || Ext == "AU" ) return SF_FORMAT_AU;
- if (Ext == "raw" || Ext == "RAW" ) return SF_FORMAT_RAW;
- if (Ext == "paf" || Ext == "PAF" ) return SF_FORMAT_PAF;
- if (Ext == "svx" || Ext == "SVX" ) return SF_FORMAT_SVX;
- if (Ext == "voc" || Ext == "VOC" ) return SF_FORMAT_VOC;
- if (Ext == "sf" || Ext == "SF" ) return SF_FORMAT_IRCAM;
- if (Ext == "w64" || Ext == "W64" ) return SF_FORMAT_W64;
- if (Ext == "mat4" || Ext == "MAT4") return SF_FORMAT_MAT4;
- if (Ext == "mat5" || Ext == "MAT5") return SF_FORMAT_MAT5;
- if (Ext == "pvf" || Ext == "PVF" ) return SF_FORMAT_PVF;
- if (Ext == "htk" || Ext == "HTK" ) return SF_FORMAT_HTK;
- if (Ext == "caf" || Ext == "CAF" ) return SF_FORMAT_CAF;
- if (Ext == "nist" || Ext == "NIST") return SF_FORMAT_NIST; // SUPPORTED ?
- if (Ext == "sds" || Ext == "SDS" ) return SF_FORMAT_SDS; // SUPPORTED ?
- if (Ext == "avr" || Ext == "AVR" ) return SF_FORMAT_AVR; // SUPPORTED ?
- if (Ext == "sd2" || Ext == "SD2" ) return SF_FORMAT_SD2; // SUPPORTED ?
- if (Ext == "flac" || Ext == "FLAC") return SF_FORMAT_FLAC; // SUPPORTED ?
-
- return -1;
-}
-
-
-////////////////////////////////////////////////////////////
-/// Functions for implementing custom read and write to memory files
-///
-////////////////////////////////////////////////////////////
-sf_count_t SoundFileDefault::MemoryGetLength(void* UserData)
-{
- MemoryInfos* Memory = static_cast<MemoryInfos*>(UserData);
-
- return Memory->TotalSize;
-}
-sf_count_t SoundFileDefault::MemoryRead(void* Ptr, sf_count_t Count, void* UserData)
-{
- MemoryInfos* Memory = static_cast<MemoryInfos*>(UserData);
-
- sf_count_t Position = Memory->DataPtr - Memory->DataStart;
- if (Position + Count >= Memory->TotalSize)
- Count = Memory->TotalSize - Position;
-
- memcpy(Ptr, Memory->DataPtr, static_cast<std::size_t>(Count));
-
- Memory->DataPtr += Count;
-
- return Count;
-}
-sf_count_t SoundFileDefault::MemorySeek(sf_count_t Offset, int Whence, void* UserData)
-{
- MemoryInfos* Memory = static_cast<MemoryInfos*>(UserData);
-
- sf_count_t Position = 0;
- switch (Whence)
- {
- case SEEK_SET :
- Position = Offset;
- break;
- case SEEK_CUR :
- Position = Memory->DataPtr - Memory->DataStart + Offset;
- break;
- case SEEK_END :
- Position = Memory->TotalSize - Offset;
- break;
- default :
- Position = 0;
- break;
- }
-
- if (Position >= Memory->TotalSize)
- Position = Memory->TotalSize - 1;
- else if (Position < 0)
- Position = 0;
-
- Memory->DataPtr = Memory->DataStart + Position;
-
- return Position;
-}
-sf_count_t SoundFileDefault::MemoryTell(void* UserData)
-{
- MemoryInfos* Memory = static_cast<MemoryInfos*>(UserData);
-
- return Memory->DataPtr - Memory->DataStart;
-}
-sf_count_t SoundFileDefault::MemoryWrite(const void*, sf_count_t, void*)
-{
- return 0;
-}
-
-
-} // namespace priv
-
-} // namespace sf
diff --git a/src/SFML/Audio/SoundFileDefault.hpp b/src/SFML/Audio/SoundFileDefault.hpp deleted file mode 100755 index 8317d54..0000000 --- a/src/SFML/Audio/SoundFileDefault.hpp +++ /dev/null @@ -1,156 +0,0 @@ -////////////////////////////////////////////////////////////
-//
-// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
-//
-// This software is provided 'as-is', without any express or implied warranty.
-// In no event will the authors be held liable for any damages arising from the use of this software.
-//
-// Permission is granted to anyone to use this software for any purpose,
-// including commercial applications, and to alter it and redistribute it freely,
-// subject to the following restrictions:
-//
-// 1. The origin of this software must not be misrepresented;
-// you must not claim that you wrote the original software.
-// If you use this software in a product, an acknowledgment
-// in the product documentation would be appreciated but is not required.
-//
-// 2. Altered source versions must be plainly marked as such,
-// and must not be misrepresented as being the original software.
-//
-// 3. This notice may not be removed or altered from any source distribution.
-//
-////////////////////////////////////////////////////////////
-
-#ifndef SFML_SOUNDFILEDEFAULT_HPP
-#define SFML_SOUNDFILEDEFAULT_HPP
-
-////////////////////////////////////////////////////////////
-// Headers
-////////////////////////////////////////////////////////////
-#include <SFML/Audio/SoundFile.hpp>
-#include <sndfile.h>
-
-
-namespace sf
-{
-namespace priv
-{
-////////////////////////////////////////////////////////////
-/// Specialization of SoundFile that can handle a lot of
-/// sound formats (see libsndfile homepage for a complete list)
-////////////////////////////////////////////////////////////
-class SoundFileDefault : public SoundFile
-{
-public :
-
- ////////////////////////////////////////////////////////////
- /// Default constructor
- ///
- ////////////////////////////////////////////////////////////
- SoundFileDefault();
-
- ////////////////////////////////////////////////////////////
- /// Destructor
- ///
- ////////////////////////////////////////////////////////////
- ~SoundFileDefault();
-
- ////////////////////////////////////////////////////////////
- /// Check if a given file is supported by this loader
- ///
- /// \param Filename : Path of the file to check
- /// \param Read : Is the file opened for reading or writing ?
- ///
- /// \param return True if the loader can handle this file
- ///
- ////////////////////////////////////////////////////////////
- static bool IsFileSupported(const std::string& Filename, bool Read);
-
- ////////////////////////////////////////////////////////////
- /// Check if a given file in memory is supported by this loader
- ///
- /// \param Data : Pointer to the file data in memory
- /// \param SizeInBytes : Size of the data to load, in bytes
- ///
- /// \param return True if the loader can handle this file
- ///
- ////////////////////////////////////////////////////////////
- static bool IsFileSupported(const char* Data, std::size_t SizeInBytes);
-
- ////////////////////////////////////////////////////////////
- /// /see sf::SoundFile::Read
- ///
- ////////////////////////////////////////////////////////////
- virtual std::size_t Read(Int16* Data, std::size_t NbSamples);
-
- ////////////////////////////////////////////////////////////
- /// /see sf::SoundFile::Write
- ///
- ////////////////////////////////////////////////////////////
- virtual void Write(const Int16* Data, std::size_t NbSamples);
-
-private :
-
- ////////////////////////////////////////////////////////////
- /// /see sf::SoundFile::OpenRead
- ///
- ////////////////////////////////////////////////////////////
- virtual bool OpenRead(const std::string& Filename, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate);
-
- ////////////////////////////////////////////////////////////
- /// /see sf::SoundFile::OpenRead
- ///
- ////////////////////////////////////////////////////////////
- virtual bool OpenRead(const char* Data, std::size_t SizeInBytes, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate);
-
- ////////////////////////////////////////////////////////////
- /// /see sf::SoundFile::OpenWrite
- ///
- ////////////////////////////////////////////////////////////
- virtual bool OpenWrite(const std::string& Filename, unsigned int ChannelsCount, unsigned int SampleRate);
-
- ////////////////////////////////////////////////////////////
- /// Get the internal format of an audio file according to
- /// its filename extension
- ///
- /// \param Filename : Filename to check
- ///
- /// \return Internal format matching the filename (-1 if no match)
- ///
- ////////////////////////////////////////////////////////////
- static int GetFormatFromFilename(const std::string& Filename);
-
- ////////////////////////////////////////////////////////////
- /// Functions for implementing custom read and write to memory files
- ///
- ////////////////////////////////////////////////////////////
- static sf_count_t MemoryGetLength(void* UserData);
- static sf_count_t MemoryRead(void* Ptr, sf_count_t Count, void* UserData);
- static sf_count_t MemorySeek(sf_count_t Offset, int Whence, void* UserData);
- static sf_count_t MemoryTell(void* UserData);
- static sf_count_t MemoryWrite(const void* Ptr, sf_count_t Count, void* UserData);
-
- ////////////////////////////////////////////////////////////
- /// Structure holding data related to memory operations
- ////////////////////////////////////////////////////////////
- struct MemoryInfos
- {
- const char* DataStart; ///< Pointer to the begining of the data
- const char* DataPtr; ///< Pointer to the current read / write position
- sf_count_t TotalSize; ///< Total size of the data, in bytes
- };
-
- ////////////////////////////////////////////////////////////
- // Member data
- ////////////////////////////////////////////////////////////
- SNDFILE* myFile; ///< File descriptor
- MemoryInfos myMemory; ///< Memory read / write data
-};
-
-} // namespace priv
-
-} // namespace sf
-
-
-#endif // SFML_SOUNDFILEDEFAULT_HPP
diff --git a/src/SFML/Audio/SoundFileOgg.cpp b/src/SFML/Audio/SoundFileOgg.cpp deleted file mode 100755 index 8aa94f8..0000000 --- a/src/SFML/Audio/SoundFileOgg.cpp +++ /dev/null @@ -1,182 +0,0 @@ -////////////////////////////////////////////////////////////
-//
-// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
-//
-// This software is provided 'as-is', without any express or implied warranty.
-// In no event will the authors be held liable for any damages arising from the use of this software.
-//
-// Permission is granted to anyone to use this software for any purpose,
-// including commercial applications, and to alter it and redistribute it freely,
-// subject to the following restrictions:
-//
-// 1. The origin of this software must not be misrepresented;
-// you must not claim that you wrote the original software.
-// If you use this software in a product, an acknowledgment
-// in the product documentation would be appreciated but is not required.
-//
-// 2. Altered source versions must be plainly marked as such,
-// and must not be misrepresented as being the original software.
-//
-// 3. This notice may not be removed or altered from any source distribution.
-//
-////////////////////////////////////////////////////////////
-
-////////////////////////////////////////////////////////////
-// Headers
-////////////////////////////////////////////////////////////
-#include <SFML/Audio/SoundFileOgg.hpp>
-#include <SFML/Audio/stb_vorbis/stb_vorbis.h>
-#include <iostream>
-
-
-namespace sf
-{
-namespace priv
-{
-////////////////////////////////////////////////////////////
-/// Default constructor
-////////////////////////////////////////////////////////////
-SoundFileOgg::SoundFileOgg() :
-myStream (NULL),
-myChannelsCount(0)
-{
-
-}
-
-
-////////////////////////////////////////////////////////////
-/// Destructor
-////////////////////////////////////////////////////////////
-SoundFileOgg::~SoundFileOgg()
-{
- if (myStream)
- stb_vorbis_close(myStream);
-}
-
-
-////////////////////////////////////////////////////////////
-/// Check if a given file is supported by this loader
-////////////////////////////////////////////////////////////
-bool SoundFileOgg::IsFileSupported(const std::string& Filename, bool Read)
-{
- if (Read)
- {
- // Open the vorbis stream
- stb_vorbis* Stream = stb_vorbis_open_filename(const_cast<char*>(Filename.c_str()), NULL, NULL);
-
- if (Stream)
- {
- stb_vorbis_close(Stream);
- return true;
- }
- else
- {
- return false;
- }
- }
- else
- {
- // No support for writing ogg files yet...
- return false;
- }
-}
-
-
-////////////////////////////////////////////////////////////
-/// Check if a given file in memory is supported by this loader
-////////////////////////////////////////////////////////////
-bool SoundFileOgg::IsFileSupported(const char* Data, std::size_t SizeInBytes)
-{
- // Open the vorbis stream
- unsigned char* Buffer = reinterpret_cast<unsigned char*>(const_cast<char*>(Data));
- int Length = static_cast<int>(SizeInBytes);
- stb_vorbis* Stream = stb_vorbis_open_memory(Buffer, Length, NULL, NULL);
-
- if (Stream)
- {
- stb_vorbis_close(Stream);
- return true;
- }
- else
- {
- return false;
- }
-}
-
-
-////////////////////////////////////////////////////////////
-/// Open the sound file for reading
-////////////////////////////////////////////////////////////
-bool SoundFileOgg::OpenRead(const std::string& Filename, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate)
-{
- // Close the file if already opened
- if (myStream)
- stb_vorbis_close(myStream);
-
- // Open the vorbis stream
- myStream = stb_vorbis_open_filename(const_cast<char*>(Filename.c_str()), NULL, NULL);
- if (myStream == NULL)
- {
- std::cerr << "Failed to read sound file \"" << Filename << "\" (cannot open the file)" << std::endl;
- return false;
- }
-
- // Get the music parameters
- stb_vorbis_info Infos = stb_vorbis_get_info(myStream);
- ChannelsCount = myChannelsCount = Infos.channels;
- SampleRate = Infos.sample_rate;
- NbSamples = static_cast<std::size_t>(stb_vorbis_stream_length_in_samples(myStream) * ChannelsCount);
-
- return true;
-}
-
-
-////////////////////////////////////////////////////////////
-/// /see sf::SoundFile::OpenRead
-////////////////////////////////////////////////////////////
-bool SoundFileOgg::OpenRead(const char* Data, std::size_t SizeInBytes, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate)
-{
- // Close the file if already opened
- if (myStream)
- stb_vorbis_close(myStream);
-
- // Open the vorbis stream
- unsigned char* Buffer = reinterpret_cast<unsigned char*>(const_cast<char*>(Data));
- int Length = static_cast<int>(SizeInBytes);
- myStream = stb_vorbis_open_memory(Buffer, Length, NULL, NULL);
- if (myStream == NULL)
- {
- std::cerr << "Failed to read sound file from memory (cannot open the file)" << std::endl;
- return false;
- }
-
- // Get the music parameters
- stb_vorbis_info Infos = stb_vorbis_get_info(myStream);
- ChannelsCount = myChannelsCount = Infos.channels;
- SampleRate = Infos.sample_rate;
- NbSamples = static_cast<std::size_t>(stb_vorbis_stream_length_in_samples(myStream) * ChannelsCount);
-
- return true;
-}
-
-
-////////////////////////////////////////////////////////////
-/// Read samples from the loaded sound
-////////////////////////////////////////////////////////////
-std::size_t SoundFileOgg::Read(Int16* Data, std::size_t NbSamples)
-{
- if (myStream && Data && NbSamples)
- {
- int Read = stb_vorbis_get_samples_short_interleaved(myStream, myChannelsCount, Data, static_cast<int>(NbSamples));
- return static_cast<std::size_t>(Read * myChannelsCount);
- }
- else
- {
- return 0;
- }
-}
-
-} // namespace priv
-
-} // namespace sf
diff --git a/src/SFML/Audio/SoundFileOgg.hpp b/src/SFML/Audio/SoundFileOgg.hpp deleted file mode 100755 index 98f4799..0000000 --- a/src/SFML/Audio/SoundFileOgg.hpp +++ /dev/null @@ -1,114 +0,0 @@ -////////////////////////////////////////////////////////////
-//
-// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
-//
-// This software is provided 'as-is', without any express or implied warranty.
-// In no event will the authors be held liable for any damages arising from the use of this software.
-//
-// Permission is granted to anyone to use this software for any purpose,
-// including commercial applications, and to alter it and redistribute it freely,
-// subject to the following restrictions:
-//
-// 1. The origin of this software must not be misrepresented;
-// you must not claim that you wrote the original software.
-// If you use this software in a product, an acknowledgment
-// in the product documentation would be appreciated but is not required.
-//
-// 2. Altered source versions must be plainly marked as such,
-// and must not be misrepresented as being the original software.
-//
-// 3. This notice may not be removed or altered from any source distribution.
-//
-////////////////////////////////////////////////////////////
-
-#ifndef SFML_SOUNDFILEOGG_HPP
-#define SFML_SOUNDFILEOGG_HPP
-
-////////////////////////////////////////////////////////////
-// Headers
-////////////////////////////////////////////////////////////
-#include <SFML/Audio/SoundFile.hpp>
-
-struct stb_vorbis;
-
-
-namespace sf
-{
-namespace priv
-{
-////////////////////////////////////////////////////////////
-/// Specialization of SoundFile that handles ogg-vorbis files (.ogg)
-/// (does not support variable bitrate / channels and writing)
-////////////////////////////////////////////////////////////
-class SoundFileOgg : public SoundFile
-{
-public :
-
- ////////////////////////////////////////////////////////////
- /// Default constructor
- ///
- ////////////////////////////////////////////////////////////
- SoundFileOgg();
-
- ////////////////////////////////////////////////////////////
- /// Destructor
- ///
- ////////////////////////////////////////////////////////////
- ~SoundFileOgg();
-
- ////////////////////////////////////////////////////////////
- /// Check if a given file is supported by this loader
- ///
- /// \param Filename : Path of the file to check
- /// \param Read : Is the file opened for reading or writing ?
- ///
- /// \param return True if the loader can handle this file
- ///
- ////////////////////////////////////////////////////////////
- static bool IsFileSupported(const std::string& Filename, bool Read);
-
- ////////////////////////////////////////////////////////////
- /// Check if a given file in memory is supported by this loader
- ///
- /// \param Data : Pointer to the file data in memory
- /// \param SizeInBytes : Size of the data to load, in bytes
- ///
- /// \param return True if the loader can handle this file
- ///
- ////////////////////////////////////////////////////////////
- static bool IsFileSupported(const char* Data, std::size_t SizeInBytes);
-
- ////////////////////////////////////////////////////////////
- /// /see sf::SoundFile::Read
- ///
- ////////////////////////////////////////////////////////////
- virtual std::size_t Read(Int16* Data, std::size_t NbSamples);
-
-private :
-
- ////////////////////////////////////////////////////////////
- /// /see sf::SoundFile::OpenRead
- ///
- ////////////////////////////////////////////////////////////
- virtual bool OpenRead(const std::string& Filename, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate);
-
- ////////////////////////////////////////////////////////////
- /// /see sf::SoundFile::OpenRead
- ///
- ////////////////////////////////////////////////////////////
- virtual bool OpenRead(const char* Data, std::size_t SizeInBytes, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate);
-
- ////////////////////////////////////////////////////////////
- // Member data
- ////////////////////////////////////////////////////////////
- stb_vorbis* myStream; ///< Vorbis stream
- unsigned int myChannelsCount; ///< Number of channels (1 = mono, 2 = stereo)
-};
-
-} // namespace priv
-
-} // namespace sf
-
-
-#endif // SFML_SOUNDFILEOGG_HPP
diff --git a/src/SFML/Audio/SoundRecorder.cpp b/src/SFML/Audio/SoundRecorder.cpp index c12a49d..bf7b03f 100755..100644 --- a/src/SFML/Audio/SoundRecorder.cpp +++ b/src/SFML/Audio/SoundRecorder.cpp @@ -1,7 +1,7 @@ ////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
@@ -27,35 +27,33 @@ ////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundRecorder.hpp>
#include <SFML/Audio/AudioDevice.hpp>
-#include <SFML/Audio/OpenAL.hpp>
+#include <SFML/Audio/ALCheck.hpp>
#include <SFML/System/Sleep.hpp>
-#include <iostream>
+#include <SFML/System/Err.hpp>
+
+#ifdef _MSC_VER
+ #pragma warning(disable : 4355) // 'this' used in base member initializer list
+#endif
-////////////////////////////////////////////////////////////
-// Private data
-////////////////////////////////////////////////////////////
namespace
{
- ALCdevice* CaptureDevice = NULL;
+ ALCdevice* captureDevice = NULL;
}
namespace sf
{
////////////////////////////////////////////////////////////
-/// Default constructor
-////////////////////////////////////////////////////////////
SoundRecorder::SoundRecorder() :
-mySampleRate (0),
-myIsCapturing(false)
+m_thread (&SoundRecorder::record, this),
+m_sampleRate (0),
+m_isCapturing(false)
{
-
+ priv::ensureALInit();
}
////////////////////////////////////////////////////////////
-/// Virtual destructor
-////////////////////////////////////////////////////////////
SoundRecorder::~SoundRecorder()
{
// Nothing to do
@@ -63,89 +61,75 @@ SoundRecorder::~SoundRecorder() ////////////////////////////////////////////////////////////
-/// Start the capture.
-/// Warning : only one capture can happen at the same time
-////////////////////////////////////////////////////////////
-void SoundRecorder::Start(unsigned int SampleRate)
+void SoundRecorder::start(unsigned int sampleRate)
{
// Check if the device can do audio capture
- if (!CanCapture())
+ if (!isAvailable())
{
- std::cerr << "Failed to start capture : your system cannot capture audio data (call SoundRecorder::CanCapture to check it)" << std::endl;
+ err() << "Failed to start capture : your system cannot capture audio data (call SoundRecorder::IsAvailable to check it)" << std::endl;
return;
}
// Check that another capture is not already running
- if (CaptureDevice)
+ if (captureDevice)
{
- std::cerr << "Trying to start audio capture, but another capture is already running" << std::endl;
+ err() << "Trying to start audio capture, but another capture is already running" << std::endl;
return;
}
// Open the capture device for capturing 16 bits mono samples
- CaptureDevice = alcCaptureOpenDevice(NULL, SampleRate, AL_FORMAT_MONO16, SampleRate);
- if (!CaptureDevice)
+ captureDevice = alcCaptureOpenDevice(NULL, sampleRate, AL_FORMAT_MONO16, sampleRate);
+ if (!captureDevice)
{
- std::cerr << "Failed to open the audio capture device" << std::endl;
+ err() << "Failed to open the audio capture device" << std::endl;
return;
}
- // Clear the sample array
- mySamples.clear();
+ // Clear the array of samples
+ m_samples.clear();
// Store the sample rate
- mySampleRate = SampleRate;
+ m_sampleRate = sampleRate;
// Notify derived class
- if (OnStart())
+ if (onStart())
{
// Start the capture
- alcCaptureStart(CaptureDevice);
+ alcCaptureStart(captureDevice);
// Start the capture in a new thread, to avoid blocking the main thread
- myIsCapturing = true;
- Launch();
+ m_isCapturing = true;
+ m_thread.launch();
}
}
////////////////////////////////////////////////////////////
-/// Stop the capture
-////////////////////////////////////////////////////////////
-void SoundRecorder::Stop()
+void SoundRecorder::stop()
{
// Stop the capturing thread
- myIsCapturing = false;
- Wait();
+ m_isCapturing = false;
+ m_thread.wait();
}
////////////////////////////////////////////////////////////
-/// Get the sample rate
-////////////////////////////////////////////////////////////
-unsigned int SoundRecorder::GetSampleRate() const
+unsigned int SoundRecorder::getSampleRate() const
{
- return mySampleRate;
+ return m_sampleRate;
}
////////////////////////////////////////////////////////////
-/// Tell if the system supports sound capture.
-/// If not, this class won't be usable
-////////////////////////////////////////////////////////////
-bool SoundRecorder::CanCapture()
+bool SoundRecorder::isAvailable()
{
- ALCdevice* Device = priv::AudioDevice::GetInstance().GetDevice();
-
- return (alcIsExtensionPresent(Device, "ALC_EXT_CAPTURE") != AL_FALSE) ||
- (alcIsExtensionPresent(Device, "ALC_EXT_capture") != AL_FALSE); // "bug" in Mac OS X 10.5 and 10.6
+ return (priv::AudioDevice::isExtensionSupported("ALC_EXT_CAPTURE") != AL_FALSE) ||
+ (priv::AudioDevice::isExtensionSupported("ALC_EXT_capture") != AL_FALSE); // "bug" in Mac OS X 10.5 and 10.6
}
////////////////////////////////////////////////////////////
-/// Start recording audio data
-////////////////////////////////////////////////////////////
-bool SoundRecorder::OnStart()
+bool SoundRecorder::onStart()
{
// Nothing to do
return true;
@@ -153,75 +137,67 @@ bool SoundRecorder::OnStart() ////////////////////////////////////////////////////////////
-/// Stop recording audio data
-////////////////////////////////////////////////////////////
-void SoundRecorder::OnStop()
+void SoundRecorder::onStop()
{
// Nothing to do
}
////////////////////////////////////////////////////////////
-/// /see Thread::Run
-////////////////////////////////////////////////////////////
-void SoundRecorder::Run()
+void SoundRecorder::record()
{
- while (myIsCapturing)
+ while (m_isCapturing)
{
// Process available samples
- ProcessCapturedSamples();
+ processCapturedSamples();
// Don't bother the CPU while waiting for more captured data
- Sleep(0.1f);
+ sleep(milliseconds(100));
}
// Capture is finished : clean up everything
- CleanUp();
+ cleanup();
// Notify derived class
- OnStop();
+ onStop();
}
////////////////////////////////////////////////////////////
-/// Get available captured samples and process them
-////////////////////////////////////////////////////////////
-void SoundRecorder::ProcessCapturedSamples()
+void SoundRecorder::processCapturedSamples()
{
// Get the number of samples available
- ALCint SamplesAvailable;
- alcGetIntegerv(CaptureDevice, ALC_CAPTURE_SAMPLES, 1, &SamplesAvailable);
+ ALCint samplesAvailable;
+ alcGetIntegerv(captureDevice, ALC_CAPTURE_SAMPLES, 1, &samplesAvailable);
- if (SamplesAvailable > 0)
+ if (samplesAvailable > 0)
{
// Get the recorded samples
- mySamples.resize(SamplesAvailable);
- alcCaptureSamples(CaptureDevice, &mySamples[0], SamplesAvailable);
+ m_samples.resize(samplesAvailable);
+ alcCaptureSamples(captureDevice, &m_samples[0], samplesAvailable);
// Forward them to the derived class
- if (!OnProcessSamples(&mySamples[0], mySamples.size()))
+ if (!onProcessSamples(&m_samples[0], m_samples.size()))
{
// The user wants to stop the capture
- myIsCapturing = false;
+ m_isCapturing = false;
}
}
}
////////////////////////////////////////////////////////////
-/// Clean up recorder internal resources
-////////////////////////////////////////////////////////////
-void SoundRecorder::CleanUp()
+void SoundRecorder::cleanup()
{
// Stop the capture
- alcCaptureStop(CaptureDevice);
+ alcCaptureStop(captureDevice);
// Get the samples left in the buffer
- ProcessCapturedSamples();
+ processCapturedSamples();
// Close the device
- alcCaptureCloseDevice(CaptureDevice);
- CaptureDevice = NULL;
+ alcCaptureCloseDevice(captureDevice);
+ captureDevice = NULL;
}
} // namespace sf
diff --git a/src/SFML/Audio/SoundSource.cpp b/src/SFML/Audio/SoundSource.cpp new file mode 100644 index 0000000..c5a6524 --- /dev/null +++ b/src/SFML/Audio/SoundSource.cpp @@ -0,0 +1,194 @@ +////////////////////////////////////////////////////////////
+//
+// SFML - Simple and Fast Multimedia Library
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
+//
+// This software is provided 'as-is', without any express or implied warranty.
+// In no event will the authors be held liable for any damages arising from the use of this software.
+//
+// Permission is granted to anyone to use this software for any purpose,
+// including commercial applications, and to alter it and redistribute it freely,
+// subject to the following restrictions:
+//
+// 1. The origin of this software must not be misrepresented;
+// you must not claim that you wrote the original software.
+// If you use this software in a product, an acknowledgment
+// in the product documentation would be appreciated but is not required.
+//
+// 2. Altered source versions must be plainly marked as such,
+// and must not be misrepresented as being the original software.
+//
+// 3. This notice may not be removed or altered from any source distribution.
+//
+////////////////////////////////////////////////////////////
+
+////////////////////////////////////////////////////////////
+// Headers
+////////////////////////////////////////////////////////////
+#include <SFML/Audio/SoundSource.hpp>
+#include <SFML/Audio/ALCheck.hpp>
+
+
+namespace sf
+{
+////////////////////////////////////////////////////////////
+SoundSource::SoundSource()
+{
+ priv::ensureALInit();
+
+ alCheck(alGenSources(1, &m_source));
+ alCheck(alSourcei(m_source, AL_BUFFER, 0));
+}
+
+
+////////////////////////////////////////////////////////////
+SoundSource::SoundSource(const SoundSource& copy)
+{
+ priv::ensureALInit();
+
+ alCheck(alGenSources(1, &m_source));
+ alCheck(alSourcei(m_source, AL_BUFFER, 0));
+
+ setPitch(copy.getPitch());
+ setVolume(copy.getVolume());
+ setPosition(copy.getPosition());
+ setRelativeToListener(copy.isRelativeToListener());
+ setMinDistance(copy.getMinDistance());
+ setAttenuation(copy.getAttenuation());
+}
+
+
+////////////////////////////////////////////////////////////
+SoundSource::~SoundSource()
+{
+ alCheck(alSourcei(m_source, AL_BUFFER, 0));
+ alCheck(alDeleteSources(1, &m_source));
+}
+
+
+////////////////////////////////////////////////////////////
+void SoundSource::setPitch(float pitch)
+{
+ alCheck(alSourcef(m_source, AL_PITCH, pitch));
+}
+
+
+////////////////////////////////////////////////////////////
+void SoundSource::setVolume(float volume)
+{
+ alCheck(alSourcef(m_source, AL_GAIN, volume * 0.01f));
+}
+
+////////////////////////////////////////////////////////////
+void SoundSource::setPosition(float x, float y, float z)
+{
+ alCheck(alSource3f(m_source, AL_POSITION, x, y, z));
+}
+
+
+////////////////////////////////////////////////////////////
+void SoundSource::setPosition(const Vector3f& position)
+{
+ setPosition(position.x, position.y, position.z);
+}
+
+
+////////////////////////////////////////////////////////////
+void SoundSource::setRelativeToListener(bool relative)
+{
+ alCheck(alSourcei(m_source, AL_SOURCE_RELATIVE, relative));
+}
+
+
+////////////////////////////////////////////////////////////
+void SoundSource::setMinDistance(float distance)
+{
+ alCheck(alSourcef(m_source, AL_REFERENCE_DISTANCE, distance));
+}
+
+
+////////////////////////////////////////////////////////////
+void SoundSource::setAttenuation(float attenuation)
+{
+ alCheck(alSourcef(m_source, AL_ROLLOFF_FACTOR, attenuation));
+}
+
+
+////////////////////////////////////////////////////////////
+float SoundSource::getPitch() const
+{
+ ALfloat pitch;
+ alCheck(alGetSourcef(m_source, AL_PITCH, &pitch));
+
+ return pitch;
+}
+
+
+////////////////////////////////////////////////////////////
+float SoundSource::getVolume() const
+{
+ ALfloat gain;
+ alCheck(alGetSourcef(m_source, AL_GAIN, &gain));
+
+ return gain * 100.f;
+}
+
+
+////////////////////////////////////////////////////////////
+Vector3f SoundSource::getPosition() const
+{
+ Vector3f position;
+ alCheck(alGetSource3f(m_source, AL_POSITION, &position.x, &position.y, &position.z));
+
+ return position;
+}
+
+
+////////////////////////////////////////////////////////////
+bool SoundSource::isRelativeToListener() const
+{
+ ALint relative;
+ alCheck(alGetSourcei(m_source, AL_SOURCE_RELATIVE, &relative));
+
+ return relative != 0;
+}
+
+
+////////////////////////////////////////////////////////////
+float SoundSource::getMinDistance() const
+{
+ ALfloat distance;
+ alCheck(alGetSourcef(m_source, AL_REFERENCE_DISTANCE, &distance));
+
+ return distance;
+}
+
+
+////////////////////////////////////////////////////////////
+float SoundSource::getAttenuation() const
+{
+ ALfloat attenuation;
+ alCheck(alGetSourcef(m_source, AL_ROLLOFF_FACTOR, &attenuation));
+
+ return attenuation;
+}
+
+
+////////////////////////////////////////////////////////////
+SoundSource::Status SoundSource::getStatus() const
+{
+ ALint status;
+ alCheck(alGetSourcei(m_source, AL_SOURCE_STATE, &status));
+
+ switch (status)
+ {
+ case AL_INITIAL :
+ case AL_STOPPED : return Stopped;
+ case AL_PAUSED : return Paused;
+ case AL_PLAYING : return Playing;
+ }
+
+ return Stopped;
+}
+
+} // namespace sf
diff --git a/src/SFML/Audio/SoundStream.cpp b/src/SFML/Audio/SoundStream.cpp index 9530d78..31df188 100755..100644 --- a/src/SFML/Audio/SoundStream.cpp +++ b/src/SFML/Audio/SoundStream.cpp @@ -1,7 +1,7 @@ ////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
-// Copyright (C) 2007-2009 Laurent Gomila (laurent.gom@gmail.com)
+// Copyright (C) 2007-2013 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
@@ -27,342 +27,339 @@ ////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundStream.hpp>
#include <SFML/Audio/AudioDevice.hpp>
-#include <SFML/Audio/OpenAL.hpp>
+#include <SFML/Audio/ALCheck.hpp>
#include <SFML/System/Sleep.hpp>
+#include <SFML/System/Err.hpp>
+
+#ifdef _MSC_VER
+ #pragma warning(disable : 4355) // 'this' used in base member initializer list
+#endif
namespace sf
{
////////////////////////////////////////////////////////////
-/// Default constructor
-////////////////////////////////////////////////////////////
SoundStream::SoundStream() :
-myIsStreaming (false),
-myChannelsCount (0),
-mySampleRate (0),
-myFormat (0),
-myLoop (false),
-mySamplesProcessed(0)
+m_thread (&SoundStream::streamData, this),
+m_isStreaming (false),
+m_channelCount (0),
+m_sampleRate (0),
+m_format (0),
+m_loop (false),
+m_samplesProcessed(0)
{
}
////////////////////////////////////////////////////////////
-/// Virtual destructor
-////////////////////////////////////////////////////////////
SoundStream::~SoundStream()
{
// Stop the sound if it was playing
- Stop();
+ stop();
}
////////////////////////////////////////////////////////////
-/// Set the audio stream parameters, you must call it before Play()
-////////////////////////////////////////////////////////////
-void SoundStream::Initialize(unsigned int ChannelsCount, unsigned int SampleRate)
+void SoundStream::initialize(unsigned int channelCount, unsigned int sampleRate)
{
- myChannelsCount = ChannelsCount;
- mySampleRate = SampleRate;
+ m_channelCount = channelCount;
+ m_sampleRate = sampleRate;
// Deduce the format from the number of channels
- myFormat = priv::AudioDevice::GetInstance().GetFormatFromChannelsCount(ChannelsCount);
+ m_format = priv::AudioDevice::getFormatFromChannelCount(channelCount);
// Check if the format is valid
- if (myFormat == 0)
+ if (m_format == 0)
{
- myChannelsCount = 0;
- mySampleRate = 0;
- std::cerr << "Unsupported number of channels (" << myChannelsCount << ")" << std::endl;
+ m_channelCount = 0;
+ m_sampleRate = 0;
+ err() << "Unsupported number of channels (" << m_channelCount << ")" << std::endl;
}
}
////////////////////////////////////////////////////////////
-/// Start playing the audio stream
-////////////////////////////////////////////////////////////
-void SoundStream::Play()
+void SoundStream::play()
{
// Check if the sound parameters have been set
- if (myFormat == 0)
+ if (m_format == 0)
{
- std::cerr << "Failed to play audio stream : sound parameters have not been initialized (call Initialize first)" << std::endl;
+ err() << "Failed to play audio stream: sound parameters have not been initialized (call Initialize first)" << std::endl;
return;
}
// If the sound is already playing (probably paused), just resume it
- if (myIsStreaming)
+ if (m_isStreaming)
{
- Sound::Play();
+ alCheck(alSourcePlay(m_source));
return;
}
- // Notify the derived class
- if (OnStart())
- {
- // Start updating the stream in a separate thread to avoid blocking the application
- mySamplesProcessed = 0;
- myIsStreaming = true;
- Launch();
- }
+ // Move to the beginning
+ onSeek(Time::Zero);
+
+ // Start updating the stream in a separate thread to avoid blocking the application
+ m_samplesProcessed = 0;
+ m_isStreaming = true;
+ m_thread.launch();
}
////////////////////////////////////////////////////////////
-/// Stop playing the audio stream
+void SoundStream::pause()
+{
+ alCheck(alSourcePause(m_source));
+}
+
+
////////////////////////////////////////////////////////////
-void SoundStream::Stop()
+void SoundStream::stop()
{
// Wait for the thread to terminate
- myIsStreaming = false;
- Wait();
+ m_isStreaming = false;
+ m_thread.wait();
}
////////////////////////////////////////////////////////////
-/// Return the number of channels (1 = mono, 2 = stereo, ...)
-////////////////////////////////////////////////////////////
-unsigned int SoundStream::GetChannelsCount() const
+unsigned int SoundStream::getChannelCount() const
{
- return myChannelsCount;
+ return m_channelCount;
}
////////////////////////////////////////////////////////////
-/// Get the sound frequency (sample rate)
-////////////////////////////////////////////////////////////
-unsigned int SoundStream::GetSampleRate() const
+unsigned int SoundStream::getSampleRate() const
{
- return mySampleRate;
+ return m_sampleRate;
}
////////////////////////////////////////////////////////////
-/// Get the status of the sound (stopped, paused, playing)
-////////////////////////////////////////////////////////////
-Sound::Status SoundStream::GetStatus() const
+SoundStream::Status SoundStream::getStatus() const
{
- Status Status = Sound::GetStatus();
+ Status status = SoundSource::getStatus();
- // To compensate for the lag between Play() and alSourcePlay()
- if ((Status == Stopped) && myIsStreaming)
- Status = Playing;
+ // To compensate for the lag between play() and alSourceplay()
+ if ((status == Stopped) && m_isStreaming)
+ status = Playing;
- return Status;
+ return status;
}
////////////////////////////////////////////////////////////
-/// Get the current playing position of the stream
-///
-/// \return Current playing position, expressed in seconds
-///
-////////////////////////////////////////////////////////////
-float SoundStream::GetPlayingOffset() const
+void SoundStream::setPlayingOffset(Time timeOffset)
{
- return Sound::GetPlayingOffset() + static_cast<float>(mySamplesProcessed) / mySampleRate / myChannelsCount;
+ // Stop the stream
+ stop();
+
+ // Let the derived class update the current position
+ onSeek(timeOffset);
+
+ // Restart streaming
+ m_samplesProcessed = static_cast<Uint64>(timeOffset.asSeconds() * m_sampleRate * m_channelCount);
+ m_isStreaming = true;
+ m_thread.launch();
}
////////////////////////////////////////////////////////////
-/// Set the music loop state
-////////////////////////////////////////////////////////////
-void SoundStream::SetLoop(bool Loop)
+Time SoundStream::getPlayingOffset() const
{
- myLoop = Loop;
+ if (m_sampleRate && m_channelCount)
+ {
+ ALfloat secs = 0.f;
+ alCheck(alGetSourcef(m_source, AL_SEC_OFFSET, &secs));
+
+ return seconds(secs + static_cast<float>(m_samplesProcessed) / m_sampleRate / m_channelCount);
+ }
+ else
+ {
+ return Time::Zero;
+ }
}
////////////////////////////////////////////////////////////
-/// Tell whether or not the music is looping
-////////////////////////////////////////////////////////////
-bool SoundStream::GetLoop() const
+void SoundStream::setLoop(bool loop)
{
- return myLoop;
+ m_loop = loop;
}
////////////////////////////////////////////////////////////
-/// /see Thread::Run
+bool SoundStream::getLoop() const
+{
+ return m_loop;
+}
+
+
////////////////////////////////////////////////////////////
-void SoundStream::Run()
+void SoundStream::streamData()
{
// Create the buffers
- ALCheck(alGenBuffers(BuffersCount, myBuffers));
- for (int i = 0; i < BuffersCount; ++i)
- myEndBuffers[i] = false;
+ alCheck(alGenBuffers(BufferCount, m_buffers));
+ for (int i = 0; i < BufferCount; ++i)
+ m_endBuffers[i] = false;
// Fill the queue
- bool RequestStop = FillQueue();
+ bool requestStop = fillQueue();
// Play the sound
- Sound::Play();
+ alCheck(alSourcePlay(m_source));
- while (myIsStreaming)
+ while (m_isStreaming)
{
- // The stream has been interrupted !
- if (Sound::GetStatus() == Stopped)
+ // The stream has been interrupted!
+ if (SoundSource::getStatus() == Stopped)
{
- if (!RequestStop)
+ if (!requestStop)
{
// Just continue
- Sound::Play();
+ alCheck(alSourcePlay(m_source));
}
else
{
// End streaming
- myIsStreaming = false;
+ m_isStreaming = false;
}
}
// Get the number of buffers that have been processed (ie. ready for reuse)
- ALint NbProcessed;
- ALCheck(alGetSourcei(Sound::mySource, AL_BUFFERS_PROCESSED, &NbProcessed));
+ ALint nbProcessed = 0;
+ alCheck(alGetSourcei(m_source, AL_BUFFERS_PROCESSED, &nbProcessed));
- while (NbProcessed--)
+ while (nbProcessed--)
{
// Pop the first unused buffer from the queue
- ALuint Buffer;
- ALCheck(alSourceUnqueueBuffers(Sound::mySource, 1, &Buffer));
+ ALuint buffer;
+ alCheck(alSourceUnqueueBuffers(m_source, 1, &buffer));
// Find its number
- unsigned int BufferNum = 0;
- for (int i = 0; i < BuffersCount; ++i)
- if (myBuffers[i] == Buffer)
+ unsigned int bufferNum = 0;
+ for (int i = 0; i < BufferCount; ++i)
+ if (m_buffers[i] == buffer)
{
- BufferNum = i;
+ bufferNum = i;
break;
}
// Retrieve its size and add it to the samples count
- if (myEndBuffers[BufferNum])
+ if (m_endBuffers[bufferNum])
{
// This was the last buffer: reset the sample count
- mySamplesProcessed = 0;
- myEndBuffers[BufferNum] = false;
+ m_samplesProcessed = 0;
+ m_endBuffers[bufferNum] = false;
}
else
{
- ALint Size;
- ALCheck(alGetBufferi(Buffer, AL_SIZE, &Size));
- mySamplesProcessed += Size / sizeof(Int16);
+ ALint size, bits;
+ alCheck(alGetBufferi(buffer, AL_SIZE, &size));
+ alCheck(alGetBufferi(buffer, AL_BITS, &bits));
+ m_samplesProcessed += size / (bits / 8);
}
// Fill it and push it back into the playing queue
- if (!RequestStop)
+ if (!requestStop)
{
- if (FillAndPushBuffer(BufferNum))
- RequestStop = true;
+ if (fillAndPushBuffer(bufferNum))
+ requestStop = true;
}
}
// Leave some time for the other threads if the stream is still playing
- if (Sound::GetStatus() != Stopped)
- Sleep(0.1f);
+ if (SoundSource::getStatus() != Stopped)
+ sleep(milliseconds(10));
}
// Stop the playback
- Sound::Stop();
+ alCheck(alSourceStop(m_source));
// Unqueue any buffer left in the queue
- ClearQueue();
+ clearQueue();
// Delete the buffers
- ALCheck(alSourcei(Sound::mySource, AL_BUFFER, 0));
- ALCheck(alDeleteBuffers(BuffersCount, myBuffers));
+ alCheck(alSourcei(m_source, AL_BUFFER, 0));
+ alCheck(alDeleteBuffers(BufferCount, m_buffers));
}
////////////////////////////////////////////////////////////
-/// Fill a new buffer with audio data, and push it to the
-/// playing queue
-////////////////////////////////////////////////////////////
-bool SoundStream::FillAndPushBuffer(unsigned int BufferNum)
+bool SoundStream::fillAndPushBuffer(unsigned int bufferNum)
{
- bool RequestStop = false;
+ bool requestStop = false;
// Acquire audio data
- Chunk Data = {NULL, 0};
- if (!OnGetData(Data))
+ Chunk data = {NULL, 0};
+ if (!onGetData(data))
{
// Mark the buffer as the last one (so that we know when to reset the playing position)
- myEndBuffers[BufferNum] = true;
+ m_endBuffers[bufferNum] = true;
// Check if the stream must loop or stop
- if (myLoop && OnStart())
+ if (m_loop)
{
- // If we succeeded to restart and we previously had no data, try to fill the buffer once again
- if (!Data.Samples || (Data.NbSamples == 0))
+ // Return to the beginning of the stream source
+ onSeek(Time::Zero);
+
+ // If we previously had no data, try to fill the buffer once again
+ if (!data.samples || (data.sampleCount == 0))
{
- return FillAndPushBuffer(BufferNum);
+ return fillAndPushBuffer(bufferNum);
}
}
else
{
- // Not looping or restart failed: request stop
- RequestStop = true;
+ // Not looping: request stop
+ requestStop = true;
}
}
// Fill the buffer if some data was returned
- if (Data.Samples && Data.NbSamples)
+ if (data.samples && data.sampleCount)
{
- unsigned int Buffer = myBuffers[BufferNum];
+ unsigned int buffer = m_buffers[bufferNum];
// Fill the buffer
- ALsizei Size = static_cast<ALsizei>(Data.NbSamples) * sizeof(Int16);
- ALCheck(alBufferData(Buffer, myFormat, Data.Samples, Size, mySampleRate));
+ ALsizei size = static_cast<ALsizei>(data.sampleCount) * sizeof(Int16);
+ alCheck(alBufferData(buffer, m_format, data.samples, size, m_sampleRate));
// Push it into the sound queue
- ALCheck(alSourceQueueBuffers(Sound::mySource, 1, &Buffer));
+ alCheck(alSourceQueueBuffers(m_source, 1, &buffer));
}
- return RequestStop;
+ return requestStop;
}
////////////////////////////////////////////////////////////
-/// Fill the buffers queue with all available buffers
-////////////////////////////////////////////////////////////
-bool SoundStream::FillQueue()
+bool SoundStream::fillQueue()
{
// Fill and enqueue all the available buffers
- bool RequestStop = false;
- for (int i = 0; (i < BuffersCount) && !RequestStop; ++i)
+ bool requestStop = false;
+ for (int i = 0; (i < BufferCount) && !requestStop; ++i)
{
- if (FillAndPushBuffer(i))
- RequestStop = true;
+ if (fillAndPushBuffer(i))
+ requestStop = true;
}
- return RequestStop;
+ return requestStop;
}
////////////////////////////////////////////////////////////
-/// Clear the queue of any remaining buffers
-////////////////////////////////////////////////////////////
-void SoundStream::ClearQueue()
+void SoundStream::clearQueue()
{
// Get the number of buffers still in the queue
- ALint NbQueued;
- ALCheck(alGetSourcei(Sound::mySource, AL_BUFFERS_QUEUED, &NbQueued));
+ ALint nbQueued;
+ alCheck(alGetSourcei(m_source, AL_BUFFERS_QUEUED, &nbQueued));
// Unqueue them all
- ALuint Buffer;
- for (ALint i = 0; i < NbQueued; ++i)
- ALCheck(alSourceUnqueueBuffers(Sound::mySource, 1, &Buffer));
-}
-
-
-////////////////////////////////////////////////////////////
-/// Called when the sound restarts
-////////////////////////////////////////////////////////////
-bool SoundStream::OnStart()
-{
- // Does nothing by default
-
- return true;
+ ALuint buffer;
+ for (ALint i = 0; i < nbQueued; ++i)
+ alCheck(alSourceUnqueueBuffers(m_source, 1, &buffer));
}
} // namespace sf
diff --git a/src/SFML/Audio/stb_vorbis/stb_vorbis.c b/src/SFML/Audio/stb_vorbis/stb_vorbis.c deleted file mode 100755 index 902761a..0000000 --- a/src/SFML/Audio/stb_vorbis/stb_vorbis.c +++ /dev/null @@ -1,5039 +0,0 @@ -// Ogg Vorbis I audio decoder -- version 0.99994
-//
-// Written in April 2007 by Sean Barrett, sponsored by RAD Game Tools.
-//
-// Placed in the public domain April 2007 by the author: no copyright is
-// claimed, and you may use it for any purpose you like.
-//
-// No warranty for any purpose is expressed or implied by the author (nor
-// by RAD Game Tools). Report bugs and send enhancements to the author.
-//
-// Get the latest version and other information at:
-// http://nothings.org/stb_vorbis/
-
-
-// Todo:
-//
-// - seeking (note you can seek yourself using the pushdata API)
-//
-// Limitations:
-//
-// - floor 0 not supported (used in old ogg vorbis files)
-// - lossless sample-truncation at beginning ignored
-// - cannot concatenate multiple vorbis streams
-// - sample positions are 32-bit, limiting seekable 192Khz
-// files to around 6 hours (Ogg supports 64-bit)
-//
-// All of these limitations may be removed in future versions.
-
-#include "stb_vorbis.h"
-
-#ifndef STB_VORBIS_HEADER_ONLY
-
-// global configuration settings (e.g. set these in the project/makefile),
-// or just set them in this file at the top (although ideally the first few
-// should be visible when the header file is compiled too, although it's not
-// crucial)
-
-// STB_VORBIS_NO_PUSHDATA_API
-// does not compile the code for the various stb_vorbis_*_pushdata()
-// functions
-// #define STB_VORBIS_NO_PUSHDATA_API
-
-// STB_VORBIS_NO_PULLDATA_API
-// does not compile the code for the non-pushdata APIs
-// #define STB_VORBIS_NO_PULLDATA_API
-
-// STB_VORBIS_NO_STDIO
-// does not compile the code for the APIs that use FILE *s internally
-// or externally (implied by STB_VORBIS_NO_PULLDATA_API)
-// #define STB_VORBIS_NO_STDIO
-
-// STB_VORBIS_NO_INTEGER_CONVERSION
-// does not compile the code for converting audio sample data from
-// float to integer (implied by STB_VORBIS_NO_PULLDATA_API)
-// #define STB_VORBIS_NO_INTEGER_CONVERSION
-
-// STB_VORBIS_NO_FAST_SCALED_FLOAT
-// does not use a fast float-to-int trick to accelerate float-to-int on
-// most platforms which requires endianness be defined correctly.
-#define STB_VORBIS_NO_FAST_SCALED_FLOAT
-
-
-// STB_VORBIS_MAX_CHANNELS [number]
-// globally define this to the maximum number of channels you need.
-// The spec does not put a restriction on channels except that
-// the count is stored in a byte, so 255 is the hard limit.
-// Reducing this saves about 16 bytes per value, so using 16 saves
-// (255-16)*16 or around 4KB. Plus anything other memory usage
-// I forgot to account for. Can probably go as low as 8 (7.1 audio),
-// 6 (5.1 audio), or 2 (stereo only).
-#ifndef STB_VORBIS_MAX_CHANNELS
-#define STB_VORBIS_MAX_CHANNELS 16 // enough for anyone?
-#endif
-
-// STB_VORBIS_PUSHDATA_CRC_COUNT [number]
-// after a flush_pushdata(), stb_vorbis begins scanning for the
-// next valid page, without backtracking. when it finds something
-// that looks like a page, it streams through it and verifies its
-// CRC32. Should that validation fail, it keeps scanning. But it's
-// possible that _while_ streaming through to check the CRC32 of
-// one candidate page, it sees another candidate page. This #define
-// determines how many "overlapping" candidate pages it can search
-// at once. Note that "real" pages are typically ~4KB to ~8KB, whereas
-// garbage pages could be as big as 64KB, but probably average ~16KB.
-// So don't hose ourselves by scanning an apparent 64KB page and
-// missing a ton of real ones in the interim; so minimum of 2
-#ifndef STB_VORBIS_PUSHDATA_CRC_COUNT
-#define STB_VORBIS_PUSHDATA_CRC_COUNT 4
-#endif
-
-// STB_VORBIS_FAST_HUFFMAN_LENGTH [number]
-// sets the log size of the huffman-acceleration table. Maximum
-// supported value is 24. with larger numbers, more decodings are O(1),
-// but the table size is larger so worse cache missing, so you'll have
-// to probe (and try multiple ogg vorbis files) to find the sweet spot.
-#ifndef STB_VORBIS_FAST_HUFFMAN_LENGTH
-#define STB_VORBIS_FAST_HUFFMAN_LENGTH 10
-#endif
-
-// STB_VORBIS_FAST_BINARY_LENGTH [number]
-// sets the log size of the binary-search acceleration table. this
-// is used in similar fashion to the fast-huffman size to set initial
-// parameters for the binary search
-
-// STB_VORBIS_FAST_HUFFMAN_INT
-// The fast huffman tables are much more efficient if they can be
-// stored as 16-bit results instead of 32-bit results. This restricts
-// the codebooks to having only 65535 possible outcomes, though.
-// (At least, accelerated by the huffman table.)
-#ifndef STB_VORBIS_FAST_HUFFMAN_INT
-#define STB_VORBIS_FAST_HUFFMAN_SHORT
-#endif
-
-// STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
-// If the 'fast huffman' search doesn't succeed, then stb_vorbis falls
-// back on binary searching for the correct one. This requires storing
-// extra tables with the huffman codes in sorted order. Defining this
-// symbol trades off space for speed by forcing a linear search in the
-// non-fast case, except for "sparse" codebooks.
-// #define STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
-
-// STB_VORBIS_DIVIDES_IN_RESIDUE
-// stb_vorbis precomputes the result of the scalar residue decoding
-// that would otherwise require a divide per chunk. you can trade off
-// space for time by defining this symbol.
-// #define STB_VORBIS_DIVIDES_IN_RESIDUE
-
-// STB_VORBIS_DIVIDES_IN_CODEBOOK
-// vorbis VQ codebooks can be encoded two ways: with every case explicitly
-// stored, or with all elements being chosen from a small range of values,
-// and all values possible in all elements. By default, stb_vorbis expands
-// this latter kind out to look like the former kind for ease of decoding,
-// because otherwise an integer divide-per-vector-element is required to
-// unpack the index. If you define STB_VORBIS_DIVIDES_IN_CODEBOOK, you can
-// trade off storage for speed.
-//#define STB_VORBIS_DIVIDES_IN_CODEBOOK
-
-// STB_VORBIS_CODEBOOK_SHORTS
-// The vorbis file format encodes VQ codebook floats as ax+b where a and
-// b are floating point per-codebook constants, and x is a 16-bit int.
-// Normally, stb_vorbis decodes them to floats rather than leaving them
-// as 16-bit ints and computing ax+b while decoding. This is a speed/space
-// tradeoff; you can save space by defining this flag.
-#ifndef STB_VORBIS_CODEBOOK_SHORTS
-#define STB_VORBIS_CODEBOOK_FLOATS
-#endif
-
-// STB_VORBIS_DIVIDE_TABLE
-// this replaces small integer divides in the floor decode loop with
-// table lookups. made less than 1% difference, so disabled by default.
-
-// STB_VORBIS_NO_INLINE_DECODE
-// disables the inlining of the scalar codebook fast-huffman decode.
-// might save a little codespace; useful for debugging
-// #define STB_VORBIS_NO_INLINE_DECODE
-
-// STB_VORBIS_NO_DEFER_FLOOR
-// Normally we only decode the floor without synthesizing the actual
-// full curve. We can instead synthesize the curve immediately. This
-// requires more memory and is very likely slower, so I don't think
-// you'd ever want to do it except for debugging.
-// #define STB_VORBIS_NO_DEFER_FLOOR
-
-
-
-
-//////////////////////////////////////////////////////////////////////////////
-
-#ifdef STB_VORBIS_NO_PULLDATA_API
- #define STB_VORBIS_NO_INTEGER_CONVERSION
- #define STB_VORBIS_NO_STDIO
-#endif
-
-#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
- #define STB_VORBIS_NO_STDIO 1
-#endif
-
-#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
-#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
-
- // only need endianness for fast-float-to-int, which we don't
- // use for pushdata
-
- #ifndef STB_VORBIS_BIG_ENDIAN
- #define STB_VORBIS_ENDIAN 0
- #else
- #define STB_VORBIS_ENDIAN 1
- #endif
-
-#endif
-#endif
-
-
-#ifndef STB_VORBIS_NO_STDIO
-#include <stdio.h>
-#endif
-
-#ifndef STB_VORBIS_NO_CRT
-#include <stdlib.h>
-#include <string.h>
-#include <assert.h>
-#include <math.h>
-
-#if !defined(__APPLE__) && !defined(MACOSX) && !defined(macintosh) && !defined(Macintosh) &&!defined(__FreeBSD__)
-#include <malloc.h>
-#endif
-
-#else
-#define NULL 0
-#endif
-
-#ifndef _MSC_VER
- #if __GNUC__
- #define __forceinline inline
- #else
- #define __forceinline
- #endif
-#endif
-
-#if STB_VORBIS_MAX_CHANNELS > 256
-#error "Value of STB_VORBIS_MAX_CHANNELS outside of allowed range"
-#endif
-
-#if STB_VORBIS_FAST_HUFFMAN_LENGTH > 24
-#error "Value of STB_VORBIS_FAST_HUFFMAN_LENGTH outside of allowed range"
-#endif
-
-
-#define MAX_BLOCKSIZE_LOG 13 // from specification
-#define MAX_BLOCKSIZE (1 << MAX_BLOCKSIZE_LOG)
-
-
-typedef unsigned char uint8;
-typedef signed char int8;
-typedef unsigned short uint16;
-typedef signed short int16;
-typedef unsigned int uint32;
-typedef signed int int32;
-
-#ifndef TRUE
-#define TRUE 1
-#define FALSE 0
-#endif
-
-#ifdef STB_VORBIS_CODEBOOK_FLOATS
-typedef float codetype;
-#else
-typedef uint16 codetype;
-#endif
-
-// @NOTE
-//
-// Some arrays below are tagged "//varies", which means it's actually
-// a variable-sized piece of data, but rather than malloc I assume it's
-// small enough it's better to just allocate it all together with the
-// main thing
-//
-// Most of the variables are specified with the smallest size I could pack
-// them into. It might give better performance to make them all full-sized
-// integers. It should be safe to freely rearrange the structures or change
-// the sizes larger--nothing relies on silently truncating etc., nor the
-// order of variables.
-
-#define FAST_HUFFMAN_TABLE_SIZE (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH)
-#define FAST_HUFFMAN_TABLE_MASK (FAST_HUFFMAN_TABLE_SIZE - 1)
-
-typedef struct
-{
- int dimensions, entries;
- uint8 *codeword_lengths;
- float minimum_value;
- float delta_value;
- uint8 value_bits;
- uint8 lookup_type;
- uint8 sequence_p;
- uint8 sparse;
- uint32 lookup_values;
- codetype *multiplicands;
- uint32 *codewords;
- #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
- int16 fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
- #else
- int32 fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
- #endif
- uint32 *sorted_codewords;
- int *sorted_values;
- int sorted_entries;
-} Codebook;
-
-typedef struct
-{
- uint8 order;
- uint16 rate;
- uint16 bark_map_size;
- uint8 amplitude_bits;
- uint8 amplitude_offset;
- uint8 number_of_books;
- uint8 book_list[16]; // varies
-} Floor0;
-
-typedef struct
-{
- uint8 partitions;
- uint8 partition_class_list[32]; // varies
- uint8 class_dimensions[16]; // varies
- uint8 class_subclasses[16]; // varies
- uint8 class_masterbooks[16]; // varies
- int16 subclass_books[16][8]; // varies
- uint16 Xlist[31*8+2]; // varies
- uint8 sorted_order[31*8+2];
- uint8 neighbors[31*8+2][2];
- uint8 floor1_multiplier;
- uint8 rangebits;
- int values;
-} Floor1;
-
-typedef union
-{
- Floor0 floor0;
- Floor1 floor1;
-} Floor;
-
-typedef struct
-{
- uint32 begin, end;
- uint32 part_size;
- uint8 classifications;
- uint8 classbook;
- uint8 **classdata;
- int16 (*residue_books)[8];
-} Residue;
-
-typedef struct
-{
- uint8 magnitude;
- uint8 angle;
- uint8 mux;
-} MappingChannel;
-
-typedef struct
-{
- uint16 coupling_steps;
- MappingChannel *chan;
- uint8 submaps;
- uint8 submap_floor[15]; // varies
- uint8 submap_residue[15]; // varies
-} Mapping;
-
-typedef struct
-{
- uint8 blockflag;
- uint8 mapping;
- uint16 windowtype;
- uint16 transformtype;
-} Mode;
-
-typedef struct
-{
- uint32 goal_crc; // expected crc if match
- int bytes_left; // bytes left in packet
- uint32 crc_so_far; // running crc
- int bytes_done; // bytes processed in _current_ chunk
- uint32 sample_loc; // granule pos encoded in page
-} CRCscan;
-
-typedef struct
-{
- uint32 page_start, page_end;
- uint32 after_previous_page_start;
- uint32 first_decoded_sample;
- uint32 last_decoded_sample;
-} ProbedPage;
-
-struct stb_vorbis
-{
- // user-accessible info
- unsigned int sample_rate;
- int channels;
-
- unsigned int setup_memory_required;
- unsigned int temp_memory_required;
- unsigned int setup_temp_memory_required;
-
- // input config
-#ifndef STB_VORBIS_NO_STDIO
- FILE *f;
- uint32 f_start;
- int close_on_free;
-#endif
-
- uint8 *stream;
- uint8 *stream_start;
- uint8 *stream_end;
-
- uint32 stream_len;
-
- uint8 push_mode;
-
- uint32 first_audio_page_offset;
-
- ProbedPage p_first, p_last;
-
- // memory management
- stb_vorbis_alloc alloc;
- int setup_offset;
- int temp_offset;
-
- // run-time results
- int eof;
- enum STBVorbisError error;
-
- // user-useful data
-
- // header info
- int blocksize[2];
- int blocksize_0, blocksize_1;
- int codebook_count;
- Codebook *codebooks;
- int floor_count;
- uint16 floor_types[64]; // varies
- Floor *floor_config;
- int residue_count;
- uint16 residue_types[64]; // varies
- Residue *residue_config;
- int mapping_count;
- Mapping *mapping;
- int mode_count;
- Mode mode_config[64]; // varies
-
- uint32 total_samples;
-
- // decode buffer
- float *channel_buffers[STB_VORBIS_MAX_CHANNELS];
- float *outputs [STB_VORBIS_MAX_CHANNELS];
-
- float *previous_window[STB_VORBIS_MAX_CHANNELS];
- int previous_length;
-
- #ifndef STB_VORBIS_NO_DEFER_FLOOR
- int16 *finalY[STB_VORBIS_MAX_CHANNELS];
- #else
- float *floor_buffers[STB_VORBIS_MAX_CHANNELS];
- #endif
-
- uint32 current_loc; // sample location of next frame to decode
- int current_loc_valid;
-
- // per-blocksize precomputed data
-
- // twiddle factors
- float *A[2],*B[2],*C[2];
- float *window[2];
- uint16 *bit_reverse[2];
-
- // current page/packet/segment streaming info
- uint32 serial; // stream serial number for verification
- int last_page;
- int segment_count;
- uint8 segments[255];
- uint8 page_flag;
- uint8 bytes_in_seg;
- uint8 first_decode;
- int next_seg;
- int last_seg; // flag that we're on the last segment
- int last_seg_which; // what was the segment number of the last seg?
- uint32 acc;
- int valid_bits;
- int packet_bytes;
- int end_seg_with_known_loc;
- uint32 known_loc_for_packet;
- int discard_samples_deferred;
- uint32 samples_output;
-
- // push mode scanning
- int page_crc_tests; // only in push_mode: number of tests active; -1 if not searching
-#ifndef STB_VORBIS_NO_PUSHDATA_API
- CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT];
-#endif
-
- // sample-access
- int channel_buffer_start;
- int channel_buffer_end;
-};
-
-extern int my_prof(int slot);
-//#define stb_prof my_prof
-
-#ifndef stb_prof
-#define stb_prof(x) 0
-#endif
-
-#if defined(STB_VORBIS_NO_PUSHDATA_API)
- #define IS_PUSH_MODE(f) FALSE
-#elif defined(STB_VORBIS_NO_PULLDATA_API)
- #define IS_PUSH_MODE(f) TRUE
-#else
- #define IS_PUSH_MODE(f) ((f)->push_mode)
-#endif
-
-typedef struct stb_vorbis vorb;
-
-static int error(vorb *f, enum STBVorbisError e)
-{
- f->error = e;
- if (!f->eof && e != VORBIS_need_more_data) {
- f->error=e; // breakpoint for debugging
- }
- return 0;
-}
-
-
-// these functions are used for allocating temporary memory
-// while decoding. if you can afford the stack space, use
-// alloca(); otherwise, provide a temp buffer and it will
-// allocate out of those.
-
-#define array_size_required(count,size) (count*(sizeof(void *)+(size)))
-
-#define temp_alloc(f,size) (f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : alloca(size))
-#ifdef dealloca
-#define temp_free(f,p) (f->alloc.alloc_buffer ? 0 : dealloca(size))
-#else
-#define temp_free(f,p) 0
-#endif
-#define temp_alloc_save(f) ((f)->temp_offset)
-#define temp_alloc_restore(f,p) ((f)->temp_offset = (p))
-
-#define temp_block_array(f,count,size) make_block_array(temp_alloc(f,array_size_required(count,size)), count, size)
-
-// given a sufficiently large block of memory, make an array of pointers to subblocks of it
-static void *make_block_array(void *mem, int count, int size)
-{
- int i;
- void ** p = (void **) mem;
- char *q = (char *) (p + count);
- for (i=0; i < count; ++i) {
- p[i] = q;
- q += size;
- }
- return p;
-}
-
-static void *setup_malloc(vorb *f, int sz)
-{
- sz = (sz+3) & ~3;
- f->setup_memory_required += sz;
- if (f->alloc.alloc_buffer) {
- void *p = (char *) f->alloc.alloc_buffer + f->setup_offset;
- if (f->setup_offset + sz > f->temp_offset) return NULL;
- f->setup_offset += sz;
- return p;
- }
- return sz ? malloc(sz) : NULL;
-}
-
-static void setup_free(vorb *f, void *p)
-{
- if (f->alloc.alloc_buffer) return; // do nothing; setup mem is not a stack
- free(p);
-}
-
-static void *setup_temp_malloc(vorb *f, int sz)
-{
- sz = (sz+3) & ~3;
- if (f->alloc.alloc_buffer) {
- if (f->temp_offset - sz < f->setup_offset) return NULL;
- f->temp_offset -= sz;
- return (char *) f->alloc.alloc_buffer + f->temp_offset;
- }
- return malloc(sz);
-}
-
-static void setup_temp_free(vorb *f, void *p, size_t sz)
-{
- if (f->alloc.alloc_buffer) {
- f->temp_offset += (sz+3)&~3;
- return;
- }
- free(p);
-}
-
-#define CRC32_POLY 0x04c11db7 // from spec
-
-static uint32 crc_table[256];
-static void crc32_init(void)
-{
- int i,j;
- uint32 s;
- for(i=0; i < 256; i++) {
- for (s=i<<24, j=0; j < 8; ++j)
- s = (s << 1) ^ (s >= (1<<31) ? CRC32_POLY : 0);
- crc_table[i] = s;
- }
-}
-
-static __forceinline uint32 crc32_update(uint32 crc, uint8 byte)
-{
- return (crc << 8) ^ crc_table[byte ^ (crc >> 24)];
-}
-
-
-// used in setup, and for huffman that doesn't go fast path
-static unsigned int bit_reverse(unsigned int n)
-{
- n = ((n & 0xAAAAAAAA) >> 1) | ((n & 0x55555555) << 1);
- n = ((n & 0xCCCCCCCC) >> 2) | ((n & 0x33333333) << 2);
- n = ((n & 0xF0F0F0F0) >> 4) | ((n & 0x0F0F0F0F) << 4);
- n = ((n & 0xFF00FF00) >> 8) | ((n & 0x00FF00FF) << 8);
- return (n >> 16) | (n << 16);
-}
-
-static float square(float x)
-{
- return x*x;
-}
-
-// this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3
-// as required by the specification. fast(?) implementation from stb.h
-// @OPTIMIZE: called multiple times per-packet with "constants"; move to setup
-static int ilog(int32 n)
-{
- static signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 };
-
- // 2 compares if n < 16, 3 compares otherwise (4 if signed or n > 1<<29)
- if (n < (1U << 14))
- if (n < (1U << 4)) return 0 + log2_4[n ];
- else if (n < (1U << 9)) return 5 + log2_4[n >> 5];
- else return 10 + log2_4[n >> 10];
- else if (n < (1U << 24))
- if (n < (1U << 19)) return 15 + log2_4[n >> 15];
- else return 20 + log2_4[n >> 20];
- else if (n < (1U << 29)) return 25 + log2_4[n >> 25];
- else if (n < (1U << 31)) return 30 + log2_4[n >> 30];
- else return 0; // signed n returns 0
-}
-
-#ifndef M_PI
- #define M_PI 3.14159265358979323846264f // from CRC
-#endif
-
-// code length assigned to a value with no huffman encoding
-#define NO_CODE 255
-
-/////////////////////// LEAF SETUP FUNCTIONS //////////////////////////
-//
-// these functions are only called at setup, and only a few times
-// per file
-
-static float float32_unpack(uint32 x)
-{
- // from the specification
- uint32 mantissa = x & 0x1fffff;
- uint32 sign = x & 0x80000000;
- uint32 exp = (x & 0x7fe00000) >> 21;
- double res = sign ? -(double)mantissa : (double)mantissa;
- return (float) ldexp((float)res, exp-788);
-}
-
-
-// zlib & jpeg huffman tables assume that the output symbols
-// can either be arbitrarily arranged, or have monotonically
-// increasing frequencies--they rely on the lengths being sorted;
-// this makes for a very simple generation algorithm.
-// vorbis allows a huffman table with non-sorted lengths. This
-// requires a more sophisticated construction, since symbols in
-// order do not map to huffman codes "in order".
-static void add_entry(Codebook *c, uint32 huff_code, int symbol, int count, int len, uint32 *values)
-{
- if (!c->sparse) {
- c->codewords [symbol] = huff_code;
- } else {
- c->codewords [count] = huff_code;
- c->codeword_lengths[count] = len;
- values [count] = symbol;
- }
-}
-
-static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
-{
- int i,k,m=0;
- uint32 available[32];
-
- memset(available, 0, sizeof(available));
- // find the first entry
- for (k=0; k < n; ++k) if (len[k] < NO_CODE) break;
- if (k == n) { assert(c->sorted_entries == 0); return TRUE; }
- // add to the list
- add_entry(c, 0, k, m++, len[k], values);
- // add all available leaves
- for (i=1; i <= len[k]; ++i)
- available[i] = 1 << (32-i);
- // note that the above code treats the first case specially,
- // but it's really the same as the following code, so they
- // could probably be combined (except the initial code is 0,
- // and I use 0 in available[] to mean 'empty')
- for (i=k+1; i < n; ++i) {
- uint32 res;
- int z = len[i], y;
- if (z == NO_CODE) continue;
- // find lowest available leaf (should always be earliest,
- // which is what the specification calls for)
- // note that this property, and the fact we can never have
- // more than one free leaf at a given level, isn't totally
- // trivial to prove, but it seems true and the assert never
- // fires, so!
- while (z > 0 && !available[z]) --z;
- if (z == 0) { assert(0); return FALSE; }
- res = available[z];
- available[z] = 0;
- add_entry(c, bit_reverse(res), i, m++, len[i], values);
- // propogate availability up the tree
- if (z != len[i]) {
- for (y=len[i]; y > z; --y) {
- assert(available[y] == 0);
- available[y] = res + (1 << (32-y));
- }
- }
- }
- return TRUE;
-}
-
-// accelerated huffman table allows fast O(1) match of all symbols
-// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH
-static void compute_accelerated_huffman(Codebook *c)
-{
- int i, len;
- for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i)
- c->fast_huffman[i] = -1;
-
- len = c->sparse ? c->sorted_entries : c->entries;
- #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
- if (len > 32767) len = 32767; // largest possible value we can encode!
- #endif
- for (i=0; i < len; ++i) {
- if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) {
- uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i];
- // set table entries for all bit combinations in the higher bits
- while (z < FAST_HUFFMAN_TABLE_SIZE) {
- c->fast_huffman[z] = i;
- z += 1 << c->codeword_lengths[i];
- }
- }
- }
-}
-
-static int uint32_compare(const void *p, const void *q)
-{
- uint32 x = * (uint32 *) p;
- uint32 y = * (uint32 *) q;
- return x < y ? -1 : x > y;
-}
-
-static int include_in_sort(Codebook *c, uint8 len)
-{
- if (c->sparse) { assert(len != NO_CODE); return TRUE; }
- if (len == NO_CODE) return FALSE;
- if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE;
- return FALSE;
-}
-
-// if the fast table above doesn't work, we want to binary
-// search them... need to reverse the bits
-static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values)
-{
- int i, len;
- // build a list of all the entries
- // OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN.
- // this is kind of a frivolous optimization--I don't see any performance improvement,
- // but it's like 4 extra lines of code, so.
- if (!c->sparse) {
- int k = 0;
- for (i=0; i < c->entries; ++i)
- if (include_in_sort(c, lengths[i]))
- c->sorted_codewords[k++] = bit_reverse(c->codewords[i]);
- assert(k == c->sorted_entries);
- } else {
- for (i=0; i < c->sorted_entries; ++i)
- c->sorted_codewords[i] = bit_reverse(c->codewords[i]);
- }
-
- qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare);
- c->sorted_codewords[c->sorted_entries] = 0xffffffff;
-
- len = c->sparse ? c->sorted_entries : c->entries;
- // now we need to indicate how they correspond; we could either
- // #1: sort a different data structure that says who they correspond to
- // #2: for each sorted entry, search the original list to find who corresponds
- // #3: for each original entry, find the sorted entry
- // #1 requires extra storage, #2 is slow, #3 can use binary search!
- for (i=0; i < len; ++i) {
- int huff_len = c->sparse ? lengths[values[i]] : lengths[i];
- if (include_in_sort(c,huff_len)) {
- uint32 code = bit_reverse(c->codewords[i]);
- int x=0, n=c->sorted_entries;
- while (n > 1) {
- // invariant: sc[x] <= code < sc[x+n]
- int m = x + (n >> 1);
- if (c->sorted_codewords[m] <= code) {
- x = m;
- n -= (n>>1);
- } else {
- n >>= 1;
- }
- }
- assert(c->sorted_codewords[x] == code);
- if (c->sparse) {
- c->sorted_values[x] = values[i];
- c->codeword_lengths[x] = huff_len;
- } else {
- c->sorted_values[x] = i;
- }
- }
- }
-}
-
-// only run while parsing the header (3 times)
-static int vorbis_validate(uint8 *data)
-{
- static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' };
- return memcmp(data, vorbis, 6) == 0;
-}
-
-// called from setup only, once per code book
-// (formula implied by specification)
-static int lookup1_values(int entries, int dim)
-{
- int r = (int) floor(exp((float) log((float) entries) / dim));
- if ((int) floor(pow((float) r+1, dim)) <= entries) // (int) cast for MinGW warning;
- ++r; // floor() to avoid _ftol() when non-CRT
- assert(pow((float) r+1, dim) > entries);
- assert((int) floor(pow((float) r, dim)) <= entries); // (int),floor() as above
- return r;
-}
-
-// called twice per file
-static void compute_twiddle_factors(int n, float *A, float *B, float *C)
-{
- int n4 = n >> 2, n8 = n >> 3;
- int k,k2;
-
- for (k=k2=0; k < n4; ++k,k2+=2) {
- A[k2 ] = (float) cos(4*k*M_PI/n);
- A[k2+1] = (float) -sin(4*k*M_PI/n);
- B[k2 ] = (float) cos((k2+1)*M_PI/n/2) * 0.5f;
- B[k2+1] = (float) sin((k2+1)*M_PI/n/2) * 0.5f;
- }
- for (k=k2=0; k < n8; ++k,k2+=2) {
- C[k2 ] = (float) cos(2*(k2+1)*M_PI/n);
- C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n);
- }
-}
-
-static void compute_window(int n, float *window)
-{
- int n2 = n >> 1, i;
- for (i=0; i < n2; ++i)
- window[i] = (float) sin(0.5 * M_PI * square((float) sin((i - 0 + 0.5) / n2 * 0.5 * M_PI)));
-}
-
-static void compute_bitreverse(int n, uint16 *rev)
-{
- int ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
- int i, n8 = n >> 3;
- for (i=0; i < n8; ++i)
- rev[i] = (bit_reverse(i) >> (32-ld+3)) << 2;
-}
-
-static int init_blocksize(vorb *f, int b, int n)
-{
- int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3;
- f->A[b] = (float *) setup_malloc(f, sizeof(float) * n2);
- f->B[b] = (float *) setup_malloc(f, sizeof(float) * n2);
- f->C[b] = (float *) setup_malloc(f, sizeof(float) * n4);
- if (!f->A[b] || !f->B[b] || !f->C[b]) return error(f, VORBIS_outofmem);
- compute_twiddle_factors(n, f->A[b], f->B[b], f->C[b]);
- f->window[b] = (float *) setup_malloc(f, sizeof(float) * n2);
- if (!f->window[b]) return error(f, VORBIS_outofmem);
- compute_window(n, f->window[b]);
- f->bit_reverse[b] = (uint16 *) setup_malloc(f, sizeof(uint16) * n8);
- if (!f->bit_reverse[b]) return error(f, VORBIS_outofmem);
- compute_bitreverse(n, f->bit_reverse[b]);
- return TRUE;
-}
-
-static void neighbors(uint16 *x, int n, int *plow, int *phigh)
-{
- int low = -1;
- int high = 65536;
- int i;
- for (i=0; i < n; ++i) {
- if (x[i] > low && x[i] < x[n]) { *plow = i; low = x[i]; }
- if (x[i] < high && x[i] > x[n]) { *phigh = i; high = x[i]; }
- }
-}
-
-// this has been repurposed so y is now the original index instead of y
-typedef struct
-{
- uint16 x,y;
-} Point;
-
-int point_compare(const void *p, const void *q)
-{
- Point *a = (Point *) p;
- Point *b = (Point *) q;
- return a->x < b->x ? -1 : a->x > b->x;
-}
-
-//
-/////////////////////// END LEAF SETUP FUNCTIONS //////////////////////////
-
-
-#if defined(STB_VORBIS_NO_STDIO)
- #define USE_MEMORY(z) TRUE
-#else
- #define USE_MEMORY(z) ((z)->stream)
-#endif
-
-static uint8 get8(vorb *z)
-{
- if (USE_MEMORY(z)) {
- if (z->stream >= z->stream_end) { z->eof = TRUE; return 0; }
- return *z->stream++;
- }
-
- #ifndef STB_VORBIS_NO_STDIO
- {
- int c = fgetc(z->f);
- if (c == EOF) { z->eof = TRUE; return 0; }
- return c;
- }
- #endif
-}
-
-static uint32 get32(vorb *f)
-{
- uint32 x;
- x = get8(f);
- x += get8(f) << 8;
- x += get8(f) << 16;
- x += get8(f) << 24;
- return x;
-}
-
-static int getn(vorb *z, uint8 *data, int n)
-{
- if (USE_MEMORY(z)) {
- if (z->stream+n > z->stream_end) { z->eof = 1; return 0; }
- memcpy(data, z->stream, n);
- z->stream += n;
- return 1;
- }
-
- #ifndef STB_VORBIS_NO_STDIO
- if (fread(data, n, 1, z->f) == 1)
- return 1;
- else {
- z->eof = 1;
- return 0;
- }
- #endif
-}
-
-static void skip(vorb *z, int n)
-{
- if (USE_MEMORY(z)) {
- z->stream += n;
- if (z->stream >= z->stream_end) z->eof = 1;
- return;
- }
- #ifndef STB_VORBIS_NO_STDIO
- {
- long x = ftell(z->f);
- fseek(z->f, x+n, SEEK_SET);
- }
- #endif
-}
-
-static int set_file_offset(stb_vorbis *f, unsigned int loc)
-{
- #ifndef STB_VORBIS_NO_PUSHDATA_API
- if (f->push_mode) return 0;
- #endif
- f->eof = 0;
- if (USE_MEMORY(f)) {
- if (f->stream_start + loc >= f->stream_end || f->stream_start + loc < f->stream_start) {
- f->stream = f->stream_end;
- f->eof = 1;
- return 0;
- } else {
- f->stream = f->stream_start + loc;
- return 1;
- }
- }
- #ifndef STB_VORBIS_NO_STDIO
- if (loc + f->f_start < loc || loc >= 0x80000000) {
- loc = 0x7fffffff;
- f->eof = 1;
- } else {
- loc += f->f_start;
- }
- if (!fseek(f->f, loc, SEEK_SET))
- return 1;
- f->eof = 1;
- fseek(f->f, f->f_start, SEEK_END);
- return 0;
- #endif
-}
-
-
-static uint8 ogg_page_header[4] = { 0x4f, 0x67, 0x67, 0x53 };
-
-static int capture_pattern(vorb *f)
-{
- if (0x4f != get8(f)) return FALSE;
- if (0x67 != get8(f)) return FALSE;
- if (0x67 != get8(f)) return FALSE;
- if (0x53 != get8(f)) return FALSE;
- return TRUE;
-}
-
-#define PAGEFLAG_continued_packet 1
-#define PAGEFLAG_first_page 2
-#define PAGEFLAG_last_page 4
-
-static int start_page_no_capturepattern(vorb *f)
-{
- uint32 loc0,loc1,n,i;
- // stream structure version
- if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version);
- // header flag
- f->page_flag = get8(f);
- // absolute granule position
- loc0 = get32(f);
- loc1 = get32(f);
- // @TODO: validate loc0,loc1 as valid positions?
- // stream serial number -- vorbis doesn't interleave, so discard
- get32(f);
- //if (f->serial != get32(f)) return error(f, VORBIS_incorrect_stream_serial_number);
- // page sequence number
- n = get32(f);
- f->last_page = n;
- // CRC32
- get32(f);
- // page_segments
- f->segment_count = get8(f);
- if (!getn(f, f->segments, f->segment_count))
- return error(f, VORBIS_unexpected_eof);
- // assume we _don't_ know any the sample position of any segments
- f->end_seg_with_known_loc = -2;
- if (loc0 != ~0 || loc1 != ~0) {
- // determine which packet is the last one that will complete
- for (i=f->segment_count-1; i >= 0; --i)
- if (f->segments[i] < 255)
- break;
- // 'i' is now the index of the _last_ segment of a packet that ends
- if (i >= 0) {
- f->end_seg_with_known_loc = i;
- f->known_loc_for_packet = loc0;
- }
- }
- if (f->first_decode) {
- int i,len;
- ProbedPage p;
- len = 0;
- for (i=0; i < f->segment_count; ++i)
- len += f->segments[i];
- len += 27 + f->segment_count;
- p.page_start = f->first_audio_page_offset;
- p.page_end = p.page_start + len;
- p.after_previous_page_start = p.page_start;
- p.first_decoded_sample = 0;
- p.last_decoded_sample = loc0;
- f->p_first = p;
- }
- f->next_seg = 0;
- return TRUE;
-}
-
-static int start_page(vorb *f)
-{
- if (!capture_pattern(f)) return error(f, VORBIS_missing_capture_pattern);
- return start_page_no_capturepattern(f);
-}
-
-static int start_packet(vorb *f)
-{
- while (f->next_seg == -1) {
- if (!start_page(f)) return FALSE;
- if (f->page_flag & PAGEFLAG_continued_packet)
- return error(f, VORBIS_continued_packet_flag_invalid);
- }
- f->last_seg = FALSE;
- f->valid_bits = 0;
- f->packet_bytes = 0;
- f->bytes_in_seg = 0;
- // f->next_seg is now valid
- return TRUE;
-}
-
-static int maybe_start_packet(vorb *f)
-{
- if (f->next_seg == -1) {
- int x = get8(f);
- if (f->eof) return FALSE; // EOF at page boundary is not an error!
- if (0x4f != x ) return error(f, VORBIS_missing_capture_pattern);
- if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
- if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
- if (0x53 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
- if (!start_page_no_capturepattern(f)) return FALSE;
- if (f->page_flag & PAGEFLAG_continued_packet) {
- // set up enough state that we can read this packet if we want,
- // e.g. during recovery
- f->last_seg = FALSE;
- f->bytes_in_seg = 0;
- return error(f, VORBIS_continued_packet_flag_invalid);
- }
- }
- return start_packet(f);
-}
-
-static int next_segment(vorb *f)
-{
- int len;
- if (f->last_seg) return 0;
- if (f->next_seg == -1) {
- f->last_seg_which = f->segment_count-1; // in case start_page fails
- if (!start_page(f)) { f->last_seg = 1; return 0; }
- if (!(f->page_flag & PAGEFLAG_continued_packet)) return error(f, VORBIS_continued_packet_flag_invalid);
- }
- len = f->segments[f->next_seg++];
- if (len < 255) {
- f->last_seg = TRUE;
- f->last_seg_which = f->next_seg-1;
- }
- if (f->next_seg >= f->segment_count)
- f->next_seg = -1;
- assert(f->bytes_in_seg == 0);
- f->bytes_in_seg = len;
- return len;
-}
-
-#define EOP (-1)
-#define INVALID_BITS (-1)
-
-static int get8_packet_raw(vorb *f)
-{
- if (!f->bytes_in_seg)
- if (f->last_seg) return EOP;
- else if (!next_segment(f)) return EOP;
- assert(f->bytes_in_seg > 0);
- --f->bytes_in_seg;
- ++f->packet_bytes;
- return get8(f);
-}
-
-static int get8_packet(vorb *f)
-{
- int x = get8_packet_raw(f);
- f->valid_bits = 0;
- return x;
-}
-
-static void flush_packet(vorb *f)
-{
- while (get8_packet_raw(f) != EOP);
-}
-
-// @OPTIMIZE: this is the secondary bit decoder, so it's probably not as important
-// as the huffman decoder?
-static uint32 get_bits(vorb *f, int n)
-{
- uint32 z;
-
- if (f->valid_bits < 0) return 0;
- if (f->valid_bits < n) {
- if (n > 24) {
- // the accumulator technique below would not work correctly in this case
- z = get_bits(f, 24);
- z += get_bits(f, n-24) << 24;
- return z;
- }
- if (f->valid_bits == 0) f->acc = 0;
- while (f->valid_bits < n) {
- int z = get8_packet_raw(f);
- if (z == EOP) {
- f->valid_bits = INVALID_BITS;
- return 0;
- }
- f->acc += z << f->valid_bits;
- f->valid_bits += 8;
- }
- }
- if (f->valid_bits < 0) return 0;
- z = f->acc & ((1 << n)-1);
- f->acc >>= n;
- f->valid_bits -= n;
- return z;
-}
-
-static int32 get_bits_signed(vorb *f, int n)
-{
- uint32 z = get_bits(f, n);
- if (z & (1 << (n-1)))
- z += ~((1 << n) - 1);
- return (int32) z;
-}
-
-// @OPTIMIZE: primary accumulator for huffman
-// expand the buffer to as many bits as possible without reading off end of packet
-// it might be nice to allow f->valid_bits and f->acc to be stored in registers,
-// e.g. cache them locally and decode locally
-static __forceinline void prep_huffman(vorb *f)
-{
- if (f->valid_bits <= 24) {
- if (f->valid_bits == 0) f->acc = 0;
- do {
- int z;
- if (f->last_seg && !f->bytes_in_seg) return;
- z = get8_packet_raw(f);
- if (z == EOP) return;
- f->acc += z << f->valid_bits;
- f->valid_bits += 8;
- } while (f->valid_bits <= 24);
- }
-}
-
-enum
-{
- VORBIS_packet_id = 1,
- VORBIS_packet_comment = 3,
- VORBIS_packet_setup = 5,
-};
-
-static int codebook_decode_scalar_raw(vorb *f, Codebook *c)
-{
- int i;
- prep_huffman(f);
-
- assert(c->sorted_codewords || c->codewords);
- // cases to use binary search: sorted_codewords && !c->codewords
- // sorted_codewords && c->entries > 8
- if (c->entries > 8 ? c->sorted_codewords!=NULL : !c->codewords) {
- // binary search
- uint32 code = bit_reverse(f->acc);
- int x=0, n=c->sorted_entries, len;
-
- while (n > 1) {
- // invariant: sc[x] <= code < sc[x+n]
- int m = x + (n >> 1);
- if (c->sorted_codewords[m] <= code) {
- x = m;
- n -= (n>>1);
- } else {
- n >>= 1;
- }
- }
- // x is now the sorted index
- if (!c->sparse) x = c->sorted_values[x];
- // x is now sorted index if sparse, or symbol otherwise
- len = c->codeword_lengths[x];
- if (f->valid_bits >= len) {
- f->acc >>= len;
- f->valid_bits -= len;
- return x;
- }
-
- f->valid_bits = 0;
- return -1;
- }
-
- // if small, linear search
- assert(!c->sparse);
- for (i=0; i < c->entries; ++i) {
- if (c->codeword_lengths[i] == NO_CODE) continue;
- if (c->codewords[i] == (f->acc & ((1 << c->codeword_lengths[i])-1))) {
- if (f->valid_bits >= c->codeword_lengths[i]) {
- f->acc >>= c->codeword_lengths[i];
- f->valid_bits -= c->codeword_lengths[i];
- return i;
- }
- f->valid_bits = 0;
- return -1;
- }
- }
-
- error(f, VORBIS_invalid_stream);
- f->valid_bits = 0;
- return -1;
-}
-
-static int codebook_decode_scalar(vorb *f, Codebook *c)
-{
- int i;
- if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH)
- prep_huffman(f);
- // fast huffman table lookup
- i = f->acc & FAST_HUFFMAN_TABLE_MASK;
- i = c->fast_huffman[i];
- if (i >= 0) {
- f->acc >>= c->codeword_lengths[i];
- f->valid_bits -= c->codeword_lengths[i];
- if (f->valid_bits < 0) { f->valid_bits = 0; return -1; }
- return i;
- }
- return codebook_decode_scalar_raw(f,c);
-}
-
-#ifndef STB_VORBIS_NO_INLINE_DECODE
-
-#define DECODE_RAW(var, f,c) \
- if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH) \
- prep_huffman(f); \
- var = f->acc & FAST_HUFFMAN_TABLE_MASK; \
- var = c->fast_huffman[var]; \
- if (var >= 0) { \
- int n = c->codeword_lengths[var]; \
- f->acc >>= n; \
- f->valid_bits -= n; \
- if (f->valid_bits < 0) { f->valid_bits = 0; var = -1; } \
- } else { \
- var = codebook_decode_scalar_raw(f,c); \
- }
-
-#else
-
-#define DECODE_RAW(var,f,c) var = codebook_decode_scalar(f,c);
-
-#endif
-
-#define DECODE(var,f,c) \
- DECODE_RAW(var,f,c) \
- if (c->sparse) var = c->sorted_values[var];
-
-#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
- #define DECODE_VQ(var,f,c) DECODE_RAW(var,f,c)
-#else
- #define DECODE_VQ(var,f,c) DECODE(var,f,c)
-#endif
-
-
-
-
-
-
-// CODEBOOK_ELEMENT_FAST is an optimization for the CODEBOOK_FLOATS case
-// where we avoid one addition
-#ifndef STB_VORBIS_CODEBOOK_FLOATS
- #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off] * c->delta_value + c->minimum_value)
- #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off] * c->delta_value)
- #define CODEBOOK_ELEMENT_BASE(c) (c->minimum_value)
-#else
- #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off])
- #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off])
- #define CODEBOOK_ELEMENT_BASE(c) (0)
-#endif
-
-static int codebook_decode_start(vorb *f, Codebook *c, int len)
-{
- int z = -1;
-
- // type 0 is only legal in a scalar context
- if (c->lookup_type == 0)
- error(f, VORBIS_invalid_stream);
- else {
- DECODE_VQ(z,f,c);
- if (c->sparse) assert(z < c->sorted_entries);
- if (z < 0) { // check for EOP
- if (!f->bytes_in_seg)
- if (f->last_seg)
- return z;
- error(f, VORBIS_invalid_stream);
- }
- }
- return z;
-}
-
-static int codebook_decode(vorb *f, Codebook *c, float *output, int len)
-{
- int i,z = codebook_decode_start(f,c,len);
- if (z < 0) return FALSE;
- if (len > c->dimensions) len = c->dimensions;
-
-#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
- if (c->lookup_type == 1) {
- float last = CODEBOOK_ELEMENT_BASE(c);
- int div = 1;
- for (i=0; i < len; ++i) {
- int off = (z / div) % c->lookup_values;
- float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
- output[i] += val;
- if (c->sequence_p) last = val + c->minimum_value;
- div *= c->lookup_values;
- }
- return TRUE;
- }
-#endif
-
- z *= c->dimensions;
- if (c->sequence_p) {
- float last = CODEBOOK_ELEMENT_BASE(c);
- for (i=0; i < len; ++i) {
- float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
- output[i] += val;
- last = val + c->minimum_value;
- }
- } else {
- float last = CODEBOOK_ELEMENT_BASE(c);
- for (i=0; i < len; ++i) {
- output[i] += CODEBOOK_ELEMENT_FAST(c,z+i) + last;
- }
- }
-
- return TRUE;
-}
-
-static int codebook_decode_step(vorb *f, Codebook *c, float *output, int len, int step)
-{
- int i,z = codebook_decode_start(f,c,len);
- float last = CODEBOOK_ELEMENT_BASE(c);
- if (z < 0) return FALSE;
- if (len > c->dimensions) len = c->dimensions;
-
-#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
- if (c->lookup_type == 1) {
- int div = 1;
- for (i=0; i < len; ++i) {
- int off = (z / div) % c->lookup_values;
- float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
- output[i*step] += val;
- if (c->sequence_p) last = val;
- div *= c->lookup_values;
- }
- return TRUE;
- }
-#endif
-
- z *= c->dimensions;
- for (i=0; i < len; ++i) {
- float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
- output[i*step] += val;
- if (c->sequence_p) last = val;
- }
-
- return TRUE;
-}
-
-static int codebook_decode_deinterleave_repeat(vorb *f, Codebook *c, float **outputs, int ch, int *c_inter_p, int *p_inter_p, int len, int total_decode)
-{
- int c_inter = *c_inter_p;
- int p_inter = *p_inter_p;
- int i,z, effective = c->dimensions;
-
- // type 0 is only legal in a scalar context
- if (c->lookup_type == 0) return error(f, VORBIS_invalid_stream);
-
- while (total_decode > 0) {
- float last = CODEBOOK_ELEMENT_BASE(c);
- DECODE_VQ(z,f,c);
- #ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
- assert(!c->sparse || z < c->sorted_entries);
- #endif
- if (z < 0) {
- if (!f->bytes_in_seg)
- if (f->last_seg) return FALSE;
- return error(f, VORBIS_invalid_stream);
- }
-
- // if this will take us off the end of the buffers, stop short!
- // we check by computing the length of the virtual interleaved
- // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter),
- // and the length we'll be using (effective)
- if (c_inter + p_inter*ch + effective > len * ch) {
- effective = len*ch - (p_inter*ch - c_inter);
- }
-
- #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
- if (c->lookup_type == 1) {
- int div = 1;
- for (i=0; i < effective; ++i) {
- int off = (z / div) % c->lookup_values;
- float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
- outputs[c_inter][p_inter] += val;
- if (++c_inter == ch) { c_inter = 0; ++p_inter; }
- if (c->sequence_p) last = val;
- div *= c->lookup_values;
- }
- } else
- #endif
- {
- z *= c->dimensions;
- if (c->sequence_p) {
- for (i=0; i < effective; ++i) {
- float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
- outputs[c_inter][p_inter] += val;
- if (++c_inter == ch) { c_inter = 0; ++p_inter; }
- last = val;
- }
- } else {
- for (i=0; i < effective; ++i) {
- float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
- outputs[c_inter][p_inter] += val;
- if (++c_inter == ch) { c_inter = 0; ++p_inter; }
- }
- }
- }
-
- total_decode -= effective;
- }
- *c_inter_p = c_inter;
- *p_inter_p = p_inter;
- return TRUE;
-}
-
-#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
-static int codebook_decode_deinterleave_repeat_2(vorb *f, Codebook *c, float **outputs, int *c_inter_p, int *p_inter_p, int len, int total_decode)
-{
- int c_inter = *c_inter_p;
- int p_inter = *p_inter_p;
- int i,z, effective = c->dimensions;
-
- // type 0 is only legal in a scalar context
- if (c->lookup_type == 0) return error(f, VORBIS_invalid_stream);
-
- while (total_decode > 0) {
- float last = CODEBOOK_ELEMENT_BASE(c);
- DECODE_VQ(z,f,c);
-
- if (z < 0) {
- if (!f->bytes_in_seg)
- if (f->last_seg) return FALSE;
- return error(f, VORBIS_invalid_stream);
- }
-
- // if this will take us off the end of the buffers, stop short!
- // we check by computing the length of the virtual interleaved
- // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter),
- // and the length we'll be using (effective)
- if (c_inter + p_inter*2 + effective > len * 2) {
- effective = len*2 - (p_inter*2 - c_inter);
- }
-
- {
- z *= c->dimensions;
- stb_prof(11);
- if (c->sequence_p) {
- // haven't optimized this case because I don't have any examples
- for (i=0; i < effective; ++i) {
- float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
- outputs[c_inter][p_inter] += val;
- if (++c_inter == 2) { c_inter = 0; ++p_inter; }
- last = val;
- }
- } else {
- i=0;
- if (c_inter == 1) {
- float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
- outputs[c_inter][p_inter] += val;
- c_inter = 0; ++p_inter;
- ++i;
- }
- {
- float *z0 = outputs[0];
- float *z1 = outputs[1];
- for (; i+1 < effective;) {
- z0[p_inter] += CODEBOOK_ELEMENT_FAST(c,z+i) + last;
- z1[p_inter] += CODEBOOK_ELEMENT_FAST(c,z+i+1) + last;
- ++p_inter;
- i += 2;
- }
- }
- if (i < effective) {
- float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
- outputs[c_inter][p_inter] += val;
- if (++c_inter == 2) { c_inter = 0; ++p_inter; }
- }
- }
- }
-
- total_decode -= effective;
- }
- *c_inter_p = c_inter;
- *p_inter_p = p_inter;
- return TRUE;
-}
-#endif
-
-static int predict_point(int x, int x0, int x1, int y0, int y1)
-{
- int dy = y1 - y0;
- int adx = x1 - x0;
- // @OPTIMIZE: force int division to round in the right direction... is this necessary on x86?
- int err = abs(dy) * (x - x0);
- int off = err / adx;
- return dy < 0 ? y0 - off : y0 + off;
-}
-
-// the following table is block-copied from the specification
-static float inverse_db_table[256] =
-{
- 1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f,
- 1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f,
- 1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f,
- 2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f,
- 2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f,
- 3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f,
- 4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f,
- 6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f,
- 7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f,
- 1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f,
- 1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f,
- 1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f,
- 2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f,
- 2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f,
- 3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f,
- 4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f,
- 5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f,
- 7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f,
- 9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f,
- 1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f,
- 1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f,
- 2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f,
- 2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f,
- 3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f,
- 4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f,
- 5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f,
- 7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f,
- 9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f,
- 0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f,
- 0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f,
- 0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f,
- 0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f,
- 0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f,
- 0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f,
- 0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f,
- 0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f,
- 0.00092223983f, 0.00098217216f, 0.0010459992f, 0.0011139742f,
- 0.0011863665f, 0.0012634633f, 0.0013455702f, 0.0014330129f,
- 0.0015261382f, 0.0016253153f, 0.0017309374f, 0.0018434235f,
- 0.0019632195f, 0.0020908006f, 0.0022266726f, 0.0023713743f,
- 0.0025254795f, 0.0026895994f, 0.0028643847f, 0.0030505286f,
- 0.0032487691f, 0.0034598925f, 0.0036847358f, 0.0039241906f,
- 0.0041792066f, 0.0044507950f, 0.0047400328f, 0.0050480668f,
- 0.0053761186f, 0.0057254891f, 0.0060975636f, 0.0064938176f,
- 0.0069158225f, 0.0073652516f, 0.0078438871f, 0.0083536271f,
- 0.0088964928f, 0.009474637f, 0.010090352f, 0.010746080f,
- 0.011444421f, 0.012188144f, 0.012980198f, 0.013823725f,
- 0.014722068f, 0.015678791f, 0.016697687f, 0.017782797f,
- 0.018938423f, 0.020169149f, 0.021479854f, 0.022875735f,
- 0.024362330f, 0.025945531f, 0.027631618f, 0.029427276f,
- 0.031339626f, 0.033376252f, 0.035545228f, 0.037855157f,
- 0.040315199f, 0.042935108f, 0.045725273f, 0.048696758f,
- 0.051861348f, 0.055231591f, 0.058820850f, 0.062643361f,
- 0.066714279f, 0.071049749f, 0.075666962f, 0.080584227f,
- 0.085821044f, 0.091398179f, 0.097337747f, 0.10366330f,
- 0.11039993f, 0.11757434f, 0.12521498f, 0.13335215f,
- 0.14201813f, 0.15124727f, 0.16107617f, 0.17154380f,
- 0.18269168f, 0.19456402f, 0.20720788f, 0.22067342f,
- 0.23501402f, 0.25028656f, 0.26655159f, 0.28387361f,
- 0.30232132f, 0.32196786f, 0.34289114f, 0.36517414f,
- 0.38890521f, 0.41417847f, 0.44109412f, 0.46975890f,
- 0.50028648f, 0.53279791f, 0.56742212f, 0.60429640f,
- 0.64356699f, 0.68538959f, 0.72993007f, 0.77736504f,
- 0.82788260f, 0.88168307f, 0.9389798f, 1.0f
-};
-
-
-// @OPTIMIZE: if you want to replace this bresenham line-drawing routine,
-// note that you must produce bit-identical output to decode correctly;
-// this specific sequence of operations is specified in the spec (it's
-// drawing integer-quantized frequency-space lines that the encoder
-// expects to be exactly the same)
-// ... also, isn't the whole point of Bresenham's algorithm to NOT
-// have to divide in the setup? sigh.
-#ifndef STB_VORBIS_NO_DEFER_FLOOR
-#define LINE_OP(a,b) a *= b
-#else
-#define LINE_OP(a,b) a = b
-#endif
-
-#ifdef STB_VORBIS_DIVIDE_TABLE
-#define DIVTAB_NUMER 32
-#define DIVTAB_DENOM 64
-int8 integer_divide_table[DIVTAB_NUMER][DIVTAB_DENOM]; // 2KB
-#endif
-
-static __forceinline void draw_line(float *output, int x0, int y0, int x1, int y1, int n)
-{
- int dy = y1 - y0;
- int adx = x1 - x0;
- int ady = abs(dy);
- int base;
- int x=x0,y=y0;
- int err = 0;
- int sy;
-
-#ifdef STB_VORBIS_DIVIDE_TABLE
- if (adx < DIVTAB_DENOM && ady < DIVTAB_NUMER) {
- if (dy < 0) {
- base = -integer_divide_table[ady][adx];
- sy = base-1;
- } else {
- base = integer_divide_table[ady][adx];
- sy = base+1;
- }
- } else {
- base = dy / adx;
- if (dy < 0)
- sy = base - 1;
- else
- sy = base+1;
- }
-#else
- base = dy / adx;
- if (dy < 0)
- sy = base - 1;
- else
- sy = base+1;
-#endif
- ady -= abs(base) * adx;
- if (x1 > n) x1 = n;
- LINE_OP(output[x], inverse_db_table[y]);
- for (++x; x < x1; ++x) {
- err += ady;
- if (err >= adx) {
- err -= adx;
- y += sy;
- } else
- y += base;
- LINE_OP(output[x], inverse_db_table[y]);
- }
-}
-
-static int residue_decode(vorb *f, Codebook *book, float *target, int offset, int n, int rtype)
-{
- int k;
- if (rtype == 0) {
- int step = n / book->dimensions;
- for (k=0; k < step; ++k)
- if (!codebook_decode_step(f, book, target+offset+k, n-offset-k, step))
- return FALSE;
- } else {
- for (k=0; k < n; ) {
- if (!codebook_decode(f, book, target+offset, n-k))
- return FALSE;
- k += book->dimensions;
- offset += book->dimensions;
- }
- }
- return TRUE;
-}
-
-static void decode_residue(vorb *f, float *residue_buffers[], int ch, int n, int rn, uint8 *do_not_decode)
-{
- int i,j,pass;
- Residue *r = f->residue_config + rn;
- int rtype = f->residue_types[rn];
- int c = r->classbook;
- int classwords = f->codebooks[c].dimensions;
- int n_read = r->end - r->begin;
- int part_read = n_read / r->part_size;
- int temp_alloc_point = temp_alloc_save(f);
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- uint8 ***part_classdata = (uint8 ***) temp_block_array(f,f->channels, part_read * sizeof(**part_classdata));
- #else
- int **classifications = (int **) temp_block_array(f,f->channels, part_read * sizeof(**classifications));
- #endif
-
- stb_prof(2);
- for (i=0; i < ch; ++i)
- if (!do_not_decode[i])
- memset(residue_buffers[i], 0, sizeof(float) * n);
-
- if (rtype == 2 && ch != 1) {
- int len = ch * n;
- for (j=0; j < ch; ++j)
- if (!do_not_decode[j])
- break;
- if (j == ch)
- goto done;
-
- stb_prof(3);
- for (pass=0; pass < 8; ++pass) {
- int pcount = 0, class_set = 0;
- if (ch == 2) {
- stb_prof(13);
- while (pcount < part_read) {
- int z = r->begin + pcount*r->part_size;
- int c_inter = (z & 1), p_inter = z>>1;
- if (pass == 0) {
- Codebook *c = f->codebooks+r->classbook;
- int q;
- DECODE(q,f,c);
- if (q == EOP) goto done;
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- part_classdata[0][class_set] = r->classdata[q];
- #else
- for (i=classwords-1; i >= 0; --i) {
- classifications[0][i+pcount] = q % r->classifications;
- q /= r->classifications;
- }
- #endif
- }
- stb_prof(5);
- for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
- int z = r->begin + pcount*r->part_size;
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- int c = part_classdata[0][class_set][i];
- #else
- int c = classifications[0][pcount];
- #endif
- int b = r->residue_books[c][pass];
- if (b >= 0) {
- Codebook *book = f->codebooks + b;
- stb_prof(20); // accounts for X time
- #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
- if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
- goto done;
- #else
- // saves 1%
- if (!codebook_decode_deinterleave_repeat_2(f, book, residue_buffers, &c_inter, &p_inter, n, r->part_size))
- goto done;
- #endif
- stb_prof(7);
- } else {
- z += r->part_size;
- c_inter = z & 1;
- p_inter = z >> 1;
- }
- }
- stb_prof(8);
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- ++class_set;
- #endif
- }
- } else if (ch == 1) {
- while (pcount < part_read) {
- int z = r->begin + pcount*r->part_size;
- int c_inter = 0, p_inter = z;
- if (pass == 0) {
- Codebook *c = f->codebooks+r->classbook;
- int q;
- DECODE(q,f,c);
- if (q == EOP) goto done;
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- part_classdata[0][class_set] = r->classdata[q];
- #else
- for (i=classwords-1; i >= 0; --i) {
- classifications[0][i+pcount] = q % r->classifications;
- q /= r->classifications;
- }
- #endif
- }
- for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
- int z = r->begin + pcount*r->part_size;
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- int c = part_classdata[0][class_set][i];
- #else
- int c = classifications[0][pcount];
- #endif
- int b = r->residue_books[c][pass];
- if (b >= 0) {
- Codebook *book = f->codebooks + b;
- stb_prof(22);
- if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
- goto done;
- stb_prof(3);
- } else {
- z += r->part_size;
- c_inter = 0;
- p_inter = z;
- }
- }
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- ++class_set;
- #endif
- }
- } else {
- while (pcount < part_read) {
- int z = r->begin + pcount*r->part_size;
- int c_inter = z % ch, p_inter = z/ch;
- if (pass == 0) {
- Codebook *c = f->codebooks+r->classbook;
- int q;
- DECODE(q,f,c);
- if (q == EOP) goto done;
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- part_classdata[0][class_set] = r->classdata[q];
- #else
- for (i=classwords-1; i >= 0; --i) {
- classifications[0][i+pcount] = q % r->classifications;
- q /= r->classifications;
- }
- #endif
- }
- for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
- int z = r->begin + pcount*r->part_size;
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- int c = part_classdata[0][class_set][i];
- #else
- int c = classifications[0][pcount];
- #endif
- int b = r->residue_books[c][pass];
- if (b >= 0) {
- Codebook *book = f->codebooks + b;
- stb_prof(22);
- if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
- goto done;
- stb_prof(3);
- } else {
- z += r->part_size;
- c_inter = z % ch;
- p_inter = z / ch;
- }
- }
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- ++class_set;
- #endif
- }
- }
- }
- goto done;
- }
- stb_prof(9);
-
- for (pass=0; pass < 8; ++pass) {
- int pcount = 0, class_set=0;
- while (pcount < part_read) {
- if (pass == 0) {
- for (j=0; j < ch; ++j) {
- if (!do_not_decode[j]) {
- Codebook *c = f->codebooks+r->classbook;
- int temp;
- DECODE(temp,f,c);
- if (temp == EOP) goto done;
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- part_classdata[j][class_set] = r->classdata[temp];
- #else
- for (i=classwords-1; i >= 0; --i) {
- classifications[j][i+pcount] = temp % r->classifications;
- temp /= r->classifications;
- }
- #endif
- }
- }
- }
- for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
- for (j=0; j < ch; ++j) {
- if (!do_not_decode[j]) {
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- int c = part_classdata[j][class_set][i];
- #else
- int c = classifications[j][pcount];
- #endif
- int b = r->residue_books[c][pass];
- if (b >= 0) {
- float *target = residue_buffers[j];
- int offset = r->begin + pcount * r->part_size;
- int n = r->part_size;
- Codebook *book = f->codebooks + b;
- if (!residue_decode(f, book, target, offset, n, rtype))
- goto done;
- }
- }
- }
- }
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- ++class_set;
- #endif
- }
- }
- done:
- stb_prof(0);
- temp_alloc_restore(f,temp_alloc_point);
-}
-
-
-#if 0
-// slow way for debugging
-void inverse_mdct_slow(float *buffer, int n)
-{
- int i,j;
- int n2 = n >> 1;
- float *x = (float *) malloc(sizeof(*x) * n2);
- memcpy(x, buffer, sizeof(*x) * n2);
- for (i=0; i < n; ++i) {
- float acc = 0;
- for (j=0; j < n2; ++j)
- // formula from paper:
- //acc += n/4.0f * x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1));
- // formula from wikipedia
- //acc += 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5));
- // these are equivalent, except the formula from the paper inverts the multiplier!
- // however, what actually works is NO MULTIPLIER!?!
- //acc += 64 * 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5));
- acc += x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1));
- buffer[i] = acc;
- }
- free(x);
-}
-#elif 0
-// same as above, but just barely able to run in real time on modern machines
-void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
-{
- float mcos[16384];
- int i,j;
- int n2 = n >> 1, nmask = (n << 2) -1;
- float *x = (float *) malloc(sizeof(*x) * n2);
- memcpy(x, buffer, sizeof(*x) * n2);
- for (i=0; i < 4*n; ++i)
- mcos[i] = (float) cos(M_PI / 2 * i / n);
-
- for (i=0; i < n; ++i) {
- float acc = 0;
- for (j=0; j < n2; ++j)
- acc += x[j] * mcos[(2 * i + 1 + n2)*(2*j+1) & nmask];
- buffer[i] = acc;
- }
- free(x);
-}
-#else
-// transform to use a slow dct-iv; this is STILL basically trivial,
-// but only requires half as many ops
-void dct_iv_slow(float *buffer, int n)
-{
- float mcos[16384];
- float x[2048];
- int i,j;
- int n2 = n >> 1, nmask = (n << 3) - 1;
- memcpy(x, buffer, sizeof(*x) * n);
- for (i=0; i < 8*n; ++i)
- mcos[i] = (float) cos(M_PI / 4 * i / n);
- for (i=0; i < n; ++i) {
- float acc = 0;
- for (j=0; j < n; ++j)
- acc += x[j] * mcos[((2 * i + 1)*(2*j+1)) & nmask];
- //acc += x[j] * cos(M_PI / n * (i + 0.5) * (j + 0.5));
- buffer[i] = acc;
- }
-}
-
-void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
-{
- int i, n4 = n >> 2, n2 = n >> 1, n3_4 = n - n4;
- float temp[4096];
-
- memcpy(temp, buffer, n2 * sizeof(float));
- dct_iv_slow(temp, n2); // returns -c'-d, a-b'
-
- for (i=0; i < n4 ; ++i) buffer[i] = temp[i+n4]; // a-b'
- for ( ; i < n3_4; ++i) buffer[i] = -temp[n3_4 - i - 1]; // b-a', c+d'
- for ( ; i < n ; ++i) buffer[i] = -temp[i - n3_4]; // c'+d
-}
-#endif
-
-#ifndef LIBVORBIS_MDCT
-#define LIBVORBIS_MDCT 0
-#endif
-
-#if LIBVORBIS_MDCT
-// directly call the vorbis MDCT using an interface documented
-// by Jeff Roberts... useful for performance comparison
-typedef struct
-{
- int n;
- int log2n;
-
- float *trig;
- int *bitrev;
-
- float scale;
-} mdct_lookup;
-
-extern void mdct_init(mdct_lookup *lookup, int n);
-extern void mdct_clear(mdct_lookup *l);
-extern void mdct_backward(mdct_lookup *init, float *in, float *out);
-
-mdct_lookup M1,M2;
-
-void inverse_mdct(float *buffer, int n, vorb *f, int blocktype)
-{
- mdct_lookup *M;
- if (M1.n == n) M = &M1;
- else if (M2.n == n) M = &M2;
- else if (M1.n == 0) { mdct_init(&M1, n); M = &M1; }
- else {
- if (M2.n) __asm int 3;
- mdct_init(&M2, n);
- M = &M2;
- }
-
- mdct_backward(M, buffer, buffer);
-}
-#endif
-
-
-// the following were split out into separate functions while optimizing;
-// they could be pushed back up but eh. __forceinline showed no change;
-// they're probably already being inlined.
-static void imdct_step3_iter0_loop(int n, float *e, int i_off, int k_off, float *A)
-{
- float *ee0 = e + i_off;
- float *ee2 = ee0 + k_off;
- int i;
-
- assert((n & 3) == 0);
- for (i=(n>>2); i > 0; --i) {
- float k00_20, k01_21;
- k00_20 = ee0[ 0] - ee2[ 0];
- k01_21 = ee0[-1] - ee2[-1];
- ee0[ 0] += ee2[ 0];//ee0[ 0] = ee0[ 0] + ee2[ 0];
- ee0[-1] += ee2[-1];//ee0[-1] = ee0[-1] + ee2[-1];
- ee2[ 0] = k00_20 * A[0] - k01_21 * A[1];
- ee2[-1] = k01_21 * A[0] + k00_20 * A[1];
- A += 8;
-
- k00_20 = ee0[-2] - ee2[-2];
- k01_21 = ee0[-3] - ee2[-3];
- ee0[-2] += ee2[-2];//ee0[-2] = ee0[-2] + ee2[-2];
- ee0[-3] += ee2[-3];//ee0[-3] = ee0[-3] + ee2[-3];
- ee2[-2] = k00_20 * A[0] - k01_21 * A[1];
- ee2[-3] = k01_21 * A[0] + k00_20 * A[1];
- A += 8;
-
- k00_20 = ee0[-4] - ee2[-4];
- k01_21 = ee0[-5] - ee2[-5];
- ee0[-4] += ee2[-4];//ee0[-4] = ee0[-4] + ee2[-4];
- ee0[-5] += ee2[-5];//ee0[-5] = ee0[-5] + ee2[-5];
- ee2[-4] = k00_20 * A[0] - k01_21 * A[1];
- ee2[-5] = k01_21 * A[0] + k00_20 * A[1];
- A += 8;
-
- k00_20 = ee0[-6] - ee2[-6];
- k01_21 = ee0[-7] - ee2[-7];
- ee0[-6] += ee2[-6];//ee0[-6] = ee0[-6] + ee2[-6];
- ee0[-7] += ee2[-7];//ee0[-7] = ee0[-7] + ee2[-7];
- ee2[-6] = k00_20 * A[0] - k01_21 * A[1];
- ee2[-7] = k01_21 * A[0] + k00_20 * A[1];
- A += 8;
- ee0 -= 8;
- ee2 -= 8;
- }
-}
-
-static void imdct_step3_inner_r_loop(int lim, float *e, int d0, int k_off, float *A, int k1)
-{
- int i;
- float k00_20, k01_21;
-
- float *e0 = e + d0;
- float *e2 = e0 + k_off;
-
- for (i=lim >> 2; i > 0; --i) {
- k00_20 = e0[-0] - e2[-0];
- k01_21 = e0[-1] - e2[-1];
- e0[-0] += e2[-0];//e0[-0] = e0[-0] + e2[-0];
- e0[-1] += e2[-1];//e0[-1] = e0[-1] + e2[-1];
- e2[-0] = (k00_20)*A[0] - (k01_21) * A[1];
- e2[-1] = (k01_21)*A[0] + (k00_20) * A[1];
-
- A += k1;
-
- k00_20 = e0[-2] - e2[-2];
- k01_21 = e0[-3] - e2[-3];
- e0[-2] += e2[-2];//e0[-2] = e0[-2] + e2[-2];
- e0[-3] += e2[-3];//e0[-3] = e0[-3] + e2[-3];
- e2[-2] = (k00_20)*A[0] - (k01_21) * A[1];
- e2[-3] = (k01_21)*A[0] + (k00_20) * A[1];
-
- A += k1;
-
- k00_20 = e0[-4] - e2[-4];
- k01_21 = e0[-5] - e2[-5];
- e0[-4] += e2[-4];//e0[-4] = e0[-4] + e2[-4];
- e0[-5] += e2[-5];//e0[-5] = e0[-5] + e2[-5];
- e2[-4] = (k00_20)*A[0] - (k01_21) * A[1];
- e2[-5] = (k01_21)*A[0] + (k00_20) * A[1];
-
- A += k1;
-
- k00_20 = e0[-6] - e2[-6];
- k01_21 = e0[-7] - e2[-7];
- e0[-6] += e2[-6];//e0[-6] = e0[-6] + e2[-6];
- e0[-7] += e2[-7];//e0[-7] = e0[-7] + e2[-7];
- e2[-6] = (k00_20)*A[0] - (k01_21) * A[1];
- e2[-7] = (k01_21)*A[0] + (k00_20) * A[1];
-
- e0 -= 8;
- e2 -= 8;
-
- A += k1;
- }
-}
-
-static void imdct_step3_inner_s_loop(int n, float *e, int i_off, int k_off, float *A, int a_off, int k0)
-{
- int i;
- float A0 = A[0];
- float A1 = A[0+1];
- float A2 = A[0+a_off];
- float A3 = A[0+a_off+1];
- float A4 = A[0+a_off*2+0];
- float A5 = A[0+a_off*2+1];
- float A6 = A[0+a_off*3+0];
- float A7 = A[0+a_off*3+1];
-
- float k00,k11;
-
- float *ee0 = e +i_off;
- float *ee2 = ee0+k_off;
-
- for (i=n; i > 0; --i) {
- k00 = ee0[ 0] - ee2[ 0];
- k11 = ee0[-1] - ee2[-1];
- ee0[ 0] = ee0[ 0] + ee2[ 0];
- ee0[-1] = ee0[-1] + ee2[-1];
- ee2[ 0] = (k00) * A0 - (k11) * A1;
- ee2[-1] = (k11) * A0 + (k00) * A1;
-
- k00 = ee0[-2] - ee2[-2];
- k11 = ee0[-3] - ee2[-3];
- ee0[-2] = ee0[-2] + ee2[-2];
- ee0[-3] = ee0[-3] + ee2[-3];
- ee2[-2] = (k00) * A2 - (k11) * A3;
- ee2[-3] = (k11) * A2 + (k00) * A3;
-
- k00 = ee0[-4] - ee2[-4];
- k11 = ee0[-5] - ee2[-5];
- ee0[-4] = ee0[-4] + ee2[-4];
- ee0[-5] = ee0[-5] + ee2[-5];
- ee2[-4] = (k00) * A4 - (k11) * A5;
- ee2[-5] = (k11) * A4 + (k00) * A5;
-
- k00 = ee0[-6] - ee2[-6];
- k11 = ee0[-7] - ee2[-7];
- ee0[-6] = ee0[-6] + ee2[-6];
- ee0[-7] = ee0[-7] + ee2[-7];
- ee2[-6] = (k00) * A6 - (k11) * A7;
- ee2[-7] = (k11) * A6 + (k00) * A7;
-
- ee0 -= k0;
- ee2 -= k0;
- }
-}
-
-static __forceinline void iter_54(float *z)
-{
- float k00,k11,k22,k33;
- float y0,y1,y2,y3;
-
- k00 = z[ 0] - z[-4];
- y0 = z[ 0] + z[-4];
- y2 = z[-2] + z[-6];
- k22 = z[-2] - z[-6];
-
- z[-0] = y0 + y2; // z0 + z4 + z2 + z6
- z[-2] = y0 - y2; // z0 + z4 - z2 - z6
-
- // done with y0,y2
-
- k33 = z[-3] - z[-7];
-
- z[-4] = k00 + k33; // z0 - z4 + z3 - z7
- z[-6] = k00 - k33; // z0 - z4 - z3 + z7
-
- // done with k33
-
- k11 = z[-1] - z[-5];
- y1 = z[-1] + z[-5];
- y3 = z[-3] + z[-7];
-
- z[-1] = y1 + y3; // z1 + z5 + z3 + z7
- z[-3] = y1 - y3; // z1 + z5 - z3 - z7
- z[-5] = k11 - k22; // z1 - z5 + z2 - z6
- z[-7] = k11 + k22; // z1 - z5 - z2 + z6
-}
-
-static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A, int base_n)
-{
- int k_off = -8;
- int a_off = base_n >> 3;
- float A2 = A[0+a_off];
- float *z = e + i_off;
- float *base = z - 16 * n;
-
- while (z > base) {
- float k00,k11;
-
- k00 = z[-0] - z[-8];
- k11 = z[-1] - z[-9];
- z[-0] = z[-0] + z[-8];
- z[-1] = z[-1] + z[-9];
- z[-8] = k00;
- z[-9] = k11 ;
-
- k00 = z[ -2] - z[-10];
- k11 = z[ -3] - z[-11];
- z[ -2] = z[ -2] + z[-10];
- z[ -3] = z[ -3] + z[-11];
- z[-10] = (k00+k11) * A2;
- z[-11] = (k11-k00) * A2;
-
- k00 = z[-12] - z[ -4]; // reverse to avoid a unary negation
- k11 = z[ -5] - z[-13];
- z[ -4] = z[ -4] + z[-12];
- z[ -5] = z[ -5] + z[-13];
- z[-12] = k11;
- z[-13] = k00;
-
- k00 = z[-14] - z[ -6]; // reverse to avoid a unary negation
- k11 = z[ -7] - z[-15];
- z[ -6] = z[ -6] + z[-14];
- z[ -7] = z[ -7] + z[-15];
- z[-14] = (k00+k11) * A2;
- z[-15] = (k00-k11) * A2;
-
- iter_54(z);
- iter_54(z-8);
- z -= 16;
- }
-}
-
-static void inverse_mdct(float *buffer, int n, vorb *f, int blocktype)
-{
- int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l;
- int n3_4 = n - n4, ld;
- // @OPTIMIZE: reduce register pressure by using fewer variables?
- int save_point = temp_alloc_save(f);
- float *buf2 = (float *) temp_alloc(f, n2 * sizeof(*buf2));
- float *u=NULL,*v=NULL;
- // twiddle factors
- float *A = f->A[blocktype];
-
- // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio"
- // See notes about bugs in that paper in less-optimal implementation 'inverse_mdct_old' after this function.
-
- // kernel from paper
-
-
- // merged:
- // copy and reflect spectral data
- // step 0
-
- // note that it turns out that the items added together during
- // this step are, in fact, being added to themselves (as reflected
- // by step 0). inexplicable inefficiency! this became obvious
- // once I combined the passes.
-
- // so there's a missing 'times 2' here (for adding X to itself).
- // this propogates through linearly to the end, where the numbers
- // are 1/2 too small, and need to be compensated for.
-
- {
- float *d,*e, *AA, *e_stop;
- d = &buf2[n2-2];
- AA = A;
- e = &buffer[0];
- e_stop = &buffer[n2];
- while (e != e_stop) {
- d[1] = (e[0] * AA[0] - e[2]*AA[1]);
- d[0] = (e[0] * AA[1] + e[2]*AA[0]);
- d -= 2;
- AA += 2;
- e += 4;
- }
-
- e = &buffer[n2-3];
- while (d >= buf2) {
- d[1] = (-e[2] * AA[0] - -e[0]*AA[1]);
- d[0] = (-e[2] * AA[1] + -e[0]*AA[0]);
- d -= 2;
- AA += 2;
- e -= 4;
- }
- }
-
- // now we use symbolic names for these, so that we can
- // possibly swap their meaning as we change which operations
- // are in place
-
- u = buffer;
- v = buf2;
-
- // step 2 (paper output is w, now u)
- // this could be in place, but the data ends up in the wrong
- // place... _somebody_'s got to swap it, so this is nominated
- {
- float *AA = &A[n2-8];
- float *d0,*d1, *e0, *e1;
-
- e0 = &v[n4];
- e1 = &v[0];
-
- d0 = &u[n4];
- d1 = &u[0];
-
- while (AA >= A) {
- float v40_20, v41_21;
-
- v41_21 = e0[1] - e1[1];
- v40_20 = e0[0] - e1[0];
- d0[1] = e0[1] + e1[1];
- d0[0] = e0[0] + e1[0];
- d1[1] = v41_21*AA[4] - v40_20*AA[5];
- d1[0] = v40_20*AA[4] + v41_21*AA[5];
-
- v41_21 = e0[3] - e1[3];
- v40_20 = e0[2] - e1[2];
- d0[3] = e0[3] + e1[3];
- d0[2] = e0[2] + e1[2];
- d1[3] = v41_21*AA[0] - v40_20*AA[1];
- d1[2] = v40_20*AA[0] + v41_21*AA[1];
-
- AA -= 8;
-
- d0 += 4;
- d1 += 4;
- e0 += 4;
- e1 += 4;
- }
- }
-
- // step 3
- ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
-
- // optimized step 3:
-
- // the original step3 loop can be nested r inside s or s inside r;
- // it's written originally as s inside r, but this is dumb when r
- // iterates many times, and s few. So I have two copies of it and
- // switch between them halfway.
-
- // this is iteration 0 of step 3
- imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*0, -(n >> 3), A);
- imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*1, -(n >> 3), A);
-
- // this is iteration 1 of step 3
- imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*0, -(n >> 4), A, 16);
- imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*1, -(n >> 4), A, 16);
- imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*2, -(n >> 4), A, 16);
- imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*3, -(n >> 4), A, 16);
-
- l=2;
- for (; l < (ld-3)>>1; ++l) {
- int k0 = n >> (l+2), k0_2 = k0>>1;
- int lim = 1 << (l+1);
- int i;
- for (i=0; i < lim; ++i)
- imdct_step3_inner_r_loop(n >> (l+4), u, n2-1 - k0*i, -k0_2, A, 1 << (l+3));
- }
-
- for (; l < ld-6; ++l) {
- int k0 = n >> (l+2), k1 = 1 << (l+3), k0_2 = k0>>1;
- int rlim = n >> (l+6), r;
- int lim = 1 << (l+1);
- int i_off;
- float *A0 = A;
- i_off = n2-1;
- for (r=rlim; r > 0; --r) {
- imdct_step3_inner_s_loop(lim, u, i_off, -k0_2, A0, k1, k0);
- A0 += k1*4;
- i_off -= 8;
- }
- }
-
- // iterations with count:
- // ld-6,-5,-4 all interleaved together
- // the big win comes from getting rid of needless flops
- // due to the constants on pass 5 & 4 being all 1 and 0;
- // combining them to be simultaneous to improve cache made little difference
- imdct_step3_inner_s_loop_ld654(n >> 5, u, n2-1, A, n);
-
- // output is u
-
- // step 4, 5, and 6
- // cannot be in-place because of step 5
- {
- uint16 *bitrev = f->bit_reverse[blocktype];
- // weirdly, I'd have thought reading sequentially and writing
- // erratically would have been better than vice-versa, but in
- // fact that's not what my testing showed. (That is, with
- // j = bitreverse(i), do you read i and write j, or read j and write i.)
-
- float *d0 = &v[n4-4];
- float *d1 = &v[n2-4];
- while (d0 >= v) {
- int k4;
-
- k4 = bitrev[0];
- d1[3] = u[k4+0];
- d1[2] = u[k4+1];
- d0[3] = u[k4+2];
- d0[2] = u[k4+3];
-
- k4 = bitrev[1];
- d1[1] = u[k4+0];
- d1[0] = u[k4+1];
- d0[1] = u[k4+2];
- d0[0] = u[k4+3];
-
- d0 -= 4;
- d1 -= 4;
- bitrev += 2;
- }
- }
- // (paper output is u, now v)
-
-
- // data must be in buf2
- assert(v == buf2);
-
- // step 7 (paper output is v, now v)
- // this is now in place
- {
- float *C = f->C[blocktype];
- float *d, *e;
-
- d = v;
- e = v + n2 - 4;
-
- while (d < e) {
- float a02,a11,b0,b1,b2,b3;
-
- a02 = d[0] - e[2];
- a11 = d[1] + e[3];
-
- b0 = C[1]*a02 + C[0]*a11;
- b1 = C[1]*a11 - C[0]*a02;
-
- b2 = d[0] + e[ 2];
- b3 = d[1] - e[ 3];
-
- d[0] = b2 + b0;
- d[1] = b3 + b1;
- e[2] = b2 - b0;
- e[3] = b1 - b3;
-
- a02 = d[2] - e[0];
- a11 = d[3] + e[1];
-
- b0 = C[3]*a02 + C[2]*a11;
- b1 = C[3]*a11 - C[2]*a02;
-
- b2 = d[2] + e[ 0];
- b3 = d[3] - e[ 1];
-
- d[2] = b2 + b0;
- d[3] = b3 + b1;
- e[0] = b2 - b0;
- e[1] = b1 - b3;
-
- C += 4;
- d += 4;
- e -= 4;
- }
- }
-
- // data must be in buf2
-
-
- // step 8+decode (paper output is X, now buffer)
- // this generates pairs of data a la 8 and pushes them directly through
- // the decode kernel (pushing rather than pulling) to avoid having
- // to make another pass later
-
- // this cannot POSSIBLY be in place, so we refer to the buffers directly
-
- {
- float *d0,*d1,*d2,*d3;
-
- float *B = f->B[blocktype] + n2 - 8;
- float *e = buf2 + n2 - 8;
- d0 = &buffer[0];
- d1 = &buffer[n2-4];
- d2 = &buffer[n2];
- d3 = &buffer[n-4];
- while (e >= v) {
- float p0,p1,p2,p3;
-
- p3 = e[6]*B[7] - e[7]*B[6];
- p2 = -e[6]*B[6] - e[7]*B[7];
-
- d0[0] = p3;
- d1[3] = - p3;
- d2[0] = p2;
- d3[3] = p2;
-
- p1 = e[4]*B[5] - e[5]*B[4];
- p0 = -e[4]*B[4] - e[5]*B[5];
-
- d0[1] = p1;
- d1[2] = - p1;
- d2[1] = p0;
- d3[2] = p0;
-
- p3 = e[2]*B[3] - e[3]*B[2];
- p2 = -e[2]*B[2] - e[3]*B[3];
-
- d0[2] = p3;
- d1[1] = - p3;
- d2[2] = p2;
- d3[1] = p2;
-
- p1 = e[0]*B[1] - e[1]*B[0];
- p0 = -e[0]*B[0] - e[1]*B[1];
-
- d0[3] = p1;
- d1[0] = - p1;
- d2[3] = p0;
- d3[0] = p0;
-
- B -= 8;
- e -= 8;
- d0 += 4;
- d2 += 4;
- d1 -= 4;
- d3 -= 4;
- }
- }
-
- temp_alloc_restore(f,save_point);
-}
-
-#if 0
-// this is the original version of the above code, if you want to optimize it from scratch
-void inverse_mdct_naive(float *buffer, int n)
-{
- float s;
- float A[1 << 12], B[1 << 12], C[1 << 11];
- int i,k,k2,k4, n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l;
- int n3_4 = n - n4, ld;
- // how can they claim this only uses N words?!
- // oh, because they're only used sparsely, whoops
- float u[1 << 13], X[1 << 13], v[1 << 13], w[1 << 13];
- // set up twiddle factors
-
- for (k=k2=0; k < n4; ++k,k2+=2) {
- A[k2 ] = (float) cos(4*k*M_PI/n);
- A[k2+1] = (float) -sin(4*k*M_PI/n);
- B[k2 ] = (float) cos((k2+1)*M_PI/n/2);
- B[k2+1] = (float) sin((k2+1)*M_PI/n/2);
- }
- for (k=k2=0; k < n8; ++k,k2+=2) {
- C[k2 ] = (float) cos(2*(k2+1)*M_PI/n);
- C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n);
- }
-
- // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio"
- // Note there are bugs in that pseudocode, presumably due to them attempting
- // to rename the arrays nicely rather than representing the way their actual
- // implementation bounces buffers back and forth. As a result, even in the
- // "some formulars corrected" version, a direct implementation fails. These
- // are noted below as "paper bug".
-
- // copy and reflect spectral data
- for (k=0; k < n2; ++k) u[k] = buffer[k];
- for ( ; k < n ; ++k) u[k] = -buffer[n - k - 1];
- // kernel from paper
- // step 1
- for (k=k2=k4=0; k < n4; k+=1, k2+=2, k4+=4) {
- v[n-k4-1] = (u[k4] - u[n-k4-1]) * A[k2] - (u[k4+2] - u[n-k4-3])*A[k2+1];
- v[n-k4-3] = (u[k4] - u[n-k4-1]) * A[k2+1] + (u[k4+2] - u[n-k4-3])*A[k2];
- }
- // step 2
- for (k=k4=0; k < n8; k+=1, k4+=4) {
- w[n2+3+k4] = v[n2+3+k4] + v[k4+3];
- w[n2+1+k4] = v[n2+1+k4] + v[k4+1];
- w[k4+3] = (v[n2+3+k4] - v[k4+3])*A[n2-4-k4] - (v[n2+1+k4]-v[k4+1])*A[n2-3-k4];
- w[k4+1] = (v[n2+1+k4] - v[k4+1])*A[n2-4-k4] + (v[n2+3+k4]-v[k4+3])*A[n2-3-k4];
- }
- // step 3
- ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
- for (l=0; l < ld-3; ++l) {
- int k0 = n >> (l+2), k1 = 1 << (l+3);
- int rlim = n >> (l+4), r4, r;
- int s2lim = 1 << (l+2), s2;
- for (r=r4=0; r < rlim; r4+=4,++r) {
- for (s2=0; s2 < s2lim; s2+=2) {
- u[n-1-k0*s2-r4] = w[n-1-k0*s2-r4] + w[n-1-k0*(s2+1)-r4];
- u[n-3-k0*s2-r4] = w[n-3-k0*s2-r4] + w[n-3-k0*(s2+1)-r4];
- u[n-1-k0*(s2+1)-r4] = (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1]
- - (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1+1];
- u[n-3-k0*(s2+1)-r4] = (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1]
- + (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1+1];
- }
- }
- if (l+1 < ld-3) {
- // paper bug: ping-ponging of u&w here is omitted
- memcpy(w, u, sizeof(u));
- }
- }
-
- // step 4
- for (i=0; i < n8; ++i) {
- int j = bit_reverse(i) >> (32-ld+3);
- assert(j < n8);
- if (i == j) {
- // paper bug: original code probably swapped in place; if copying,
- // need to directly copy in this case
- int i8 = i << 3;
- v[i8+1] = u[i8+1];
- v[i8+3] = u[i8+3];
- v[i8+5] = u[i8+5];
- v[i8+7] = u[i8+7];
- } else if (i < j) {
- int i8 = i << 3, j8 = j << 3;
- v[j8+1] = u[i8+1], v[i8+1] = u[j8 + 1];
- v[j8+3] = u[i8+3], v[i8+3] = u[j8 + 3];
- v[j8+5] = u[i8+5], v[i8+5] = u[j8 + 5];
- v[j8+7] = u[i8+7], v[i8+7] = u[j8 + 7];
- }
- }
- // step 5
- for (k=0; k < n2; ++k) {
- w[k] = v[k*2+1];
- }
- // step 6
- for (k=k2=k4=0; k < n8; ++k, k2 += 2, k4 += 4) {
- u[n-1-k2] = w[k4];
- u[n-2-k2] = w[k4+1];
- u[n3_4 - 1 - k2] = w[k4+2];
- u[n3_4 - 2 - k2] = w[k4+3];
- }
- // step 7
- for (k=k2=0; k < n8; ++k, k2 += 2) {
- v[n2 + k2 ] = ( u[n2 + k2] + u[n-2-k2] + C[k2+1]*(u[n2+k2]-u[n-2-k2]) + C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2;
- v[n-2 - k2] = ( u[n2 + k2] + u[n-2-k2] - C[k2+1]*(u[n2+k2]-u[n-2-k2]) - C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2;
- v[n2+1+ k2] = ( u[n2+1+k2] - u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2;
- v[n-1 - k2] = (-u[n2+1+k2] + u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2;
- }
- // step 8
- for (k=k2=0; k < n4; ++k,k2 += 2) {
- X[k] = v[k2+n2]*B[k2 ] + v[k2+1+n2]*B[k2+1];
- X[n2-1-k] = v[k2+n2]*B[k2+1] - v[k2+1+n2]*B[k2 ];
- }
-
- // decode kernel to output
- // determined the following value experimentally
- // (by first figuring out what made inverse_mdct_slow work); then matching that here
- // (probably vorbis encoder premultiplies by n or n/2, to save it on the decoder?)
- s = 0.5; // theoretically would be n4
-
- // [[[ note! the s value of 0.5 is compensated for by the B[] in the current code,
- // so it needs to use the "old" B values to behave correctly, or else
- // set s to 1.0 ]]]
- for (i=0; i < n4 ; ++i) buffer[i] = s * X[i+n4];
- for ( ; i < n3_4; ++i) buffer[i] = -s * X[n3_4 - i - 1];
- for ( ; i < n ; ++i) buffer[i] = -s * X[i - n3_4];
-}
-#endif
-
-static float *get_window(vorb *f, int len)
-{
- len <<= 1;
- if (len == f->blocksize_0) return f->window[0];
- if (len == f->blocksize_1) return f->window[1];
- assert(0);
- return NULL;
-}
-
-#ifndef STB_VORBIS_NO_DEFER_FLOOR
-typedef int16 YTYPE;
-#else
-typedef int YTYPE;
-#endif
-static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *finalY, uint8 *step2_flag)
-{
- int n2 = n >> 1;
- int s = map->chan[i].mux, floor;
- floor = map->submap_floor[s];
- if (f->floor_types[floor] == 0) {
- return error(f, VORBIS_invalid_stream);
- } else {
- Floor1 *g = &f->floor_config[floor].floor1;
- int j,q;
- int lx = 0, ly = finalY[0] * g->floor1_multiplier;
- for (q=1; q < g->values; ++q) {
- j = g->sorted_order[q];
- #ifndef STB_VORBIS_NO_DEFER_FLOOR
- if (finalY[j] >= 0)
- #else
- if (step2_flag[j])
- #endif
- {
- int hy = finalY[j] * g->floor1_multiplier;
- int hx = g->Xlist[j];
- draw_line(target, lx,ly, hx,hy, n2);
- lx = hx, ly = hy;
- }
- }
- if (lx < n2)
- // optimization of: draw_line(target, lx,ly, n,ly, n2);
- for (j=lx; j < n2; ++j)
- LINE_OP(target[j], inverse_db_table[ly]);
- }
- return TRUE;
-}
-
-static int vorbis_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode)
-{
- Mode *m;
- int i, n, prev, next, window_center;
- f->channel_buffer_start = f->channel_buffer_end = 0;
-
- retry:
- if (f->eof) return FALSE;
- if (!maybe_start_packet(f))
- return FALSE;
- // check packet type
- if (get_bits(f,1) != 0) {
- if (IS_PUSH_MODE(f))
- return error(f,VORBIS_bad_packet_type);
- while (EOP != get8_packet(f));
- goto retry;
- }
-
- if (f->alloc.alloc_buffer)
- assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
-
- i = get_bits(f, ilog(f->mode_count-1));
- if (i == EOP) return FALSE;
- if (i >= f->mode_count) return FALSE;
- *mode = i;
- m = f->mode_config + i;
- if (m->blockflag) {
- n = f->blocksize_1;
- prev = get_bits(f,1);
- next = get_bits(f,1);
- } else {
- prev = next = 0;
- n = f->blocksize_0;
- }
-
-// WINDOWING
-
- window_center = n >> 1;
- if (m->blockflag && !prev) {
- *p_left_start = (n - f->blocksize_0) >> 2;
- *p_left_end = (n + f->blocksize_0) >> 2;
- } else {
- *p_left_start = 0;
- *p_left_end = window_center;
- }
- if (m->blockflag && !next) {
- *p_right_start = (n*3 - f->blocksize_0) >> 2;
- *p_right_end = (n*3 + f->blocksize_0) >> 2;
- } else {
- *p_right_start = window_center;
- *p_right_end = n;
- }
- return TRUE;
-}
-
-static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start, int left_end, int right_start, int right_end, int *p_left)
-{
- Mapping *map;
- int i,j,k,n,n2;
- int zero_channel[256];
- int really_zero_channel[256];
- int window_center;
-
-// WINDOWING
-
- n = f->blocksize[m->blockflag];
- window_center = n >> 1;
-
- map = &f->mapping[m->mapping];
-
-// FLOORS
- n2 = n >> 1;
-
- stb_prof(1);
- for (i=0; i < f->channels; ++i) {
- int s = map->chan[i].mux, floor;
- zero_channel[i] = FALSE;
- floor = map->submap_floor[s];
- if (f->floor_types[floor] == 0) {
- return error(f, VORBIS_invalid_stream);
- } else {
- Floor1 *g = &f->floor_config[floor].floor1;
- if (get_bits(f, 1)) {
- short *finalY;
- uint8 step2_flag[256];
- static int range_list[4] = { 256, 128, 86, 64 };
- int range = range_list[g->floor1_multiplier-1];
- int offset = 2;
- finalY = f->finalY[i];
- finalY[0] = get_bits(f, ilog(range)-1);
- finalY[1] = get_bits(f, ilog(range)-1);
- for (j=0; j < g->partitions; ++j) {
- int pclass = g->partition_class_list[j];
- int cdim = g->class_dimensions[pclass];
- int cbits = g->class_subclasses[pclass];
- int csub = (1 << cbits)-1;
- int cval = 0;
- if (cbits) {
- Codebook *c = f->codebooks + g->class_masterbooks[pclass];
- DECODE(cval,f,c);
- }
- for (k=0; k < cdim; ++k) {
- int book = g->subclass_books[pclass][cval & csub];
- cval = cval >> cbits;
- if (book >= 0) {
- int temp;
- Codebook *c = f->codebooks + book;
- DECODE(temp,f,c);
- finalY[offset++] = temp;
- } else
- finalY[offset++] = 0;
- }
- }
- if (f->valid_bits == INVALID_BITS) goto error; // behavior according to spec
- step2_flag[0] = step2_flag[1] = 1;
- for (j=2; j < g->values; ++j) {
- int low, high, pred, highroom, lowroom, room, val;
- low = g->neighbors[j][0];
- high = g->neighbors[j][1];
- //neighbors(g->Xlist, j, &low, &high);
- pred = predict_point(g->Xlist[j], g->Xlist[low], g->Xlist[high], finalY[low], finalY[high]);
- val = finalY[j];
- highroom = range - pred;
- lowroom = pred;
- if (highroom < lowroom)
- room = highroom * 2;
- else
- room = lowroom * 2;
- if (val) {
- step2_flag[low] = step2_flag[high] = 1;
- step2_flag[j] = 1;
- if (val >= room)
- if (highroom > lowroom)
- finalY[j] = val - lowroom + pred;
- else
- finalY[j] = pred - val + highroom - 1;
- else
- if (val & 1)
- finalY[j] = pred - ((val+1)>>1);
- else
- finalY[j] = pred + (val>>1);
- } else {
- step2_flag[j] = 0;
- finalY[j] = pred;
- }
- }
-
-#ifdef STB_VORBIS_NO_DEFER_FLOOR
- do_floor(f, map, i, n, f->floor_buffers[i], finalY, step2_flag);
-#else
- // defer final floor computation until _after_ residue
- for (j=0; j < g->values; ++j) {
- if (!step2_flag[j])
- finalY[j] = -1;
- }
-#endif
- } else {
- error:
- zero_channel[i] = TRUE;
- }
- // So we just defer everything else to later
-
- // at this point we've decoded the floor into buffer
- }
- }
- stb_prof(0);
- // at this point we've decoded all floors
-
- if (f->alloc.alloc_buffer)
- assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
-
- // re-enable coupled channels if necessary
- memcpy(really_zero_channel, zero_channel, sizeof(really_zero_channel[0]) * f->channels);
- for (i=0; i < map->coupling_steps; ++i)
- if (!zero_channel[map->chan[i].magnitude] || !zero_channel[map->chan[i].angle]) {
- zero_channel[map->chan[i].magnitude] = zero_channel[map->chan[i].angle] = FALSE;
- }
-
-// RESIDUE DECODE
- for (i=0; i < map->submaps; ++i) {
- float *residue_buffers[STB_VORBIS_MAX_CHANNELS];
- int r,t;
- uint8 do_not_decode[256];
- int ch = 0;
- for (j=0; j < f->channels; ++j) {
- if (map->chan[j].mux == i) {
- if (zero_channel[j]) {
- do_not_decode[ch] = TRUE;
- residue_buffers[ch] = NULL;
- } else {
- do_not_decode[ch] = FALSE;
- residue_buffers[ch] = f->channel_buffers[j];
- }
- ++ch;
- }
- }
- r = map->submap_residue[i];
- t = f->residue_types[r];
- decode_residue(f, residue_buffers, ch, n2, r, do_not_decode);
- }
-
- if (f->alloc.alloc_buffer)
- assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
-
-// INVERSE COUPLING
- stb_prof(14);
- for (i = map->coupling_steps-1; i >= 0; --i) {
- int n2 = n >> 1;
- float *m = f->channel_buffers[map->chan[i].magnitude];
- float *a = f->channel_buffers[map->chan[i].angle ];
- for (j=0; j < n2; ++j) {
- float a2,m2;
- if (m[j] > 0)
- if (a[j] > 0)
- m2 = m[j], a2 = m[j] - a[j];
- else
- a2 = m[j], m2 = m[j] + a[j];
- else
- if (a[j] > 0)
- m2 = m[j], a2 = m[j] + a[j];
- else
- a2 = m[j], m2 = m[j] - a[j];
- m[j] = m2;
- a[j] = a2;
- }
- }
-
- // finish decoding the floors
-#ifndef STB_VORBIS_NO_DEFER_FLOOR
- stb_prof(15);
- for (i=0; i < f->channels; ++i) {
- if (really_zero_channel[i]) {
- memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2);
- } else {
- do_floor(f, map, i, n, f->channel_buffers[i], f->finalY[i], NULL);
- }
- }
-#else
- for (i=0; i < f->channels; ++i) {
- if (really_zero_channel[i]) {
- memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2);
- } else {
- for (j=0; j < n2; ++j)
- f->channel_buffers[i][j] *= f->floor_buffers[i][j];
- }
- }
-#endif
-
-// INVERSE MDCT
- stb_prof(16);
- for (i=0; i < f->channels; ++i)
- inverse_mdct(f->channel_buffers[i], n, f, m->blockflag);
- stb_prof(0);
-
- // this shouldn't be necessary, unless we exited on an error
- // and want to flush to get to the next packet
- flush_packet(f);
-
- if (f->first_decode) {
- // assume we start so first non-discarded sample is sample 0
- // this isn't to spec, but spec would require us to read ahead
- // and decode the size of all current frames--could be done,
- // but presumably it's not a commonly used feature
- f->current_loc = -n2; // start of first frame is positioned for discard
- // we might have to discard samples "from" the next frame too,
- // if we're lapping a large block then a small at the start?
- f->discard_samples_deferred = n - right_end;
- f->current_loc_valid = TRUE;
- f->first_decode = FALSE;
- } else if (f->discard_samples_deferred) {
- left_start += f->discard_samples_deferred;
- *p_left = left_start;
- f->discard_samples_deferred = 0;
- } else if (f->previous_length == 0 && f->current_loc_valid) {
- // we're recovering from a seek... that means we're going to discard
- // the samples from this packet even though we know our position from
- // the last page header, so we need to update the position based on
- // the discarded samples here
- // but wait, the code below is going to add this in itself even
- // on a discard, so we don't need to do it here...
- }
-
- // check if we have ogg information about the sample # for this packet
- if (f->last_seg_which == f->end_seg_with_known_loc) {
- // if we have a valid current loc, and this is final:
- if (f->current_loc_valid && (f->page_flag & PAGEFLAG_last_page)) {
- uint32 current_end = f->known_loc_for_packet - (n-right_end);
- // then let's infer the size of the (probably) short final frame
- if (current_end < f->current_loc + right_end) {
- if (current_end < f->current_loc) {
- // negative truncation, that's impossible!
- *len = 0;
- } else {
- *len = current_end - f->current_loc;
- }
- *len += left_start;
- f->current_loc += *len;
- return TRUE;
- }
- }
- // otherwise, just set our sample loc
- // guess that the ogg granule pos refers to the _middle_ of the
- // last frame?
- // set f->current_loc to the position of left_start
- f->current_loc = f->known_loc_for_packet - (n2-left_start);
- f->current_loc_valid = TRUE;
- }
- if (f->current_loc_valid)
- f->current_loc += (right_start - left_start);
-
- if (f->alloc.alloc_buffer)
- assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
- *len = right_end; // ignore samples after the window goes to 0
- return TRUE;
-}
-
-static int vorbis_decode_packet(vorb *f, int *len, int *p_left, int *p_right)
-{
- int mode, left_end, right_end;
- if (!vorbis_decode_initial(f, p_left, &left_end, p_right, &right_end, &mode)) return 0;
- return vorbis_decode_packet_rest(f, len, f->mode_config + mode, *p_left, left_end, *p_right, right_end, p_left);
-}
-
-static int vorbis_finish_frame(stb_vorbis *f, int len, int left, int right)
-{
- int prev,i,j;
- // we use right&left (the start of the right- and left-window sin()-regions)
- // to determine how much to return, rather than inferring from the rules
- // (same result, clearer code); 'left' indicates where our sin() window
- // starts, therefore where the previous window's right edge starts, and
- // therefore where to start mixing from the previous buffer. 'right'
- // indicates where our sin() ending-window starts, therefore that's where
- // we start saving, and where our returned-data ends.
-
- // mixin from previous window
- if (f->previous_length) {
- int i,j, n = f->previous_length;
- float *w = get_window(f, n);
- for (i=0; i < f->channels; ++i) {
- for (j=0; j < n; ++j)
- f->channel_buffers[i][left+j] =
- f->channel_buffers[i][left+j]*w[ j] +
- f->previous_window[i][ j]*w[n-1-j];
- }
- }
-
- prev = f->previous_length;
-
- // last half of this data becomes previous window
- f->previous_length = len - right;
-
- // @OPTIMIZE: could avoid this copy by double-buffering the
- // output (flipping previous_window with channel_buffers), but
- // then previous_window would have to be 2x as large, and
- // channel_buffers couldn't be temp mem (although they're NOT
- // currently temp mem, they could be (unless we want to level
- // performance by spreading out the computation))
- for (i=0; i < f->channels; ++i)
- for (j=0; right+j < len; ++j)
- f->previous_window[i][j] = f->channel_buffers[i][right+j];
-
- if (!prev)
- // there was no previous packet, so this data isn't valid...
- // this isn't entirely true, only the would-have-overlapped data
- // isn't valid, but this seems to be what the spec requires
- return 0;
-
- // truncate a short frame
- if (len < right) right = len;
-
- f->samples_output += right-left;
-
- return right - left;
-}
-
-static void vorbis_pump_first_frame(stb_vorbis *f)
-{
- int len, right, left;
- if (vorbis_decode_packet(f, &len, &left, &right))
- vorbis_finish_frame(f, len, left, right);
-}
-
-#ifndef STB_VORBIS_NO_PUSHDATA_API
-static int is_whole_packet_present(stb_vorbis *f, int end_page)
-{
- // make sure that we have the packet available before continuing...
- // this requires a full ogg parse, but we know we can fetch from f->stream
-
- // instead of coding this out explicitly, we could save the current read state,
- // read the next packet with get8() until end-of-packet, check f->eof, then
- // reset the state? but that would be slower, esp. since we'd have over 256 bytes
- // of state to restore (primarily the page segment table)
-
- int s = f->next_seg, first = TRUE;
- uint8 *p = f->stream;
-
- if (s != -1) { // if we're not starting the packet with a 'continue on next page' flag
- for (; s < f->segment_count; ++s) {
- p += f->segments[s];
- if (f->segments[s] < 255) // stop at first short segment
- break;
- }
- // either this continues, or it ends it...
- if (end_page)
- if (s < f->segment_count-1) return error(f, VORBIS_invalid_stream);
- if (s == f->segment_count)
- s = -1; // set 'crosses page' flag
- if (p > f->stream_end) return error(f, VORBIS_need_more_data);
- first = FALSE;
- }
- for (; s == -1;) {
- uint8 *q;
- int n;
-
- // check that we have the page header ready
- if (p + 26 >= f->stream_end) return error(f, VORBIS_need_more_data);
- // validate the page
- if (memcmp(p, ogg_page_header, 4)) return error(f, VORBIS_invalid_stream);
- if (p[4] != 0) return error(f, VORBIS_invalid_stream);
- if (first) { // the first segment must NOT have 'continued_packet', later ones MUST
- if (f->previous_length)
- if ((p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream);
- // if no previous length, we're resynching, so we can come in on a continued-packet,
- // which we'll just drop
- } else {
- if (!(p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream);
- }
- n = p[26]; // segment counts
- q = p+27; // q points to segment table
- p = q + n; // advance past header
- // make sure we've read the segment table
- if (p > f->stream_end) return error(f, VORBIS_need_more_data);
- for (s=0; s < n; ++s) {
- p += q[s];
- if (q[s] < 255)
- break;
- }
- if (end_page)
- if (s < n-1) return error(f, VORBIS_invalid_stream);
- if (s == f->segment_count)
- s = -1; // set 'crosses page' flag
- if (p > f->stream_end) return error(f, VORBIS_need_more_data);
- first = FALSE;
- }
- return TRUE;
-}
-#endif // !STB_VORBIS_NO_PUSHDATA_API
-
-static int start_decoder(vorb *f)
-{
- uint8 header[6], x,y;
- int len,i,j,k, max_submaps = 0;
- int longest_floorlist=0;
-
- // first page, first packet
-
- if (!start_page(f)) return FALSE;
- // validate page flag
- if (!(f->page_flag & PAGEFLAG_first_page)) return error(f, VORBIS_invalid_first_page);
- if (f->page_flag & PAGEFLAG_last_page) return error(f, VORBIS_invalid_first_page);
- if (f->page_flag & PAGEFLAG_continued_packet) return error(f, VORBIS_invalid_first_page);
- // check for expected packet length
- if (f->segment_count != 1) return error(f, VORBIS_invalid_first_page);
- if (f->segments[0] != 30) return error(f, VORBIS_invalid_first_page);
- // read packet
- // check packet header
- if (get8(f) != VORBIS_packet_id) return error(f, VORBIS_invalid_first_page);
- if (!getn(f, header, 6)) return error(f, VORBIS_unexpected_eof);
- if (!vorbis_validate(header)) return error(f, VORBIS_invalid_first_page);
- // vorbis_version
- if (get32(f) != 0) return error(f, VORBIS_invalid_first_page);
- f->channels = get8(f); if (!f->channels) return error(f, VORBIS_invalid_first_page);
- if (f->channels > STB_VORBIS_MAX_CHANNELS) return error(f, VORBIS_too_many_channels);
- f->sample_rate = get32(f); if (!f->sample_rate) return error(f, VORBIS_invalid_first_page);
- get32(f); // bitrate_maximum
- get32(f); // bitrate_nominal
- get32(f); // bitrate_minimum
- x = get8(f);
- { int log0,log1;
- log0 = x & 15;
- log1 = x >> 4;
- f->blocksize_0 = 1 << log0;
- f->blocksize_1 = 1 << log1;
- if (log0 < 6 || log0 > 13) return error(f, VORBIS_invalid_setup);
- if (log1 < 6 || log1 > 13) return error(f, VORBIS_invalid_setup);
- if (log0 > log1) return error(f, VORBIS_invalid_setup);
- }
-
- // framing_flag
- x = get8(f);
- if (!(x & 1)) return error(f, VORBIS_invalid_first_page);
-
- // second packet!
- if (!start_page(f)) return FALSE;
-
- if (!start_packet(f)) return FALSE;
- do {
- len = next_segment(f);
- skip(f, len);
- f->bytes_in_seg = 0;
- } while (len);
-
- // third packet!
- if (!start_packet(f)) return FALSE;
-
- #ifndef STB_VORBIS_NO_PUSHDATA_API
- if (IS_PUSH_MODE(f)) {
- if (!is_whole_packet_present(f, TRUE)) {
- // convert error in ogg header to write type
- if (f->error == VORBIS_invalid_stream)
- f->error = VORBIS_invalid_setup;
- return FALSE;
- }
- }
- #endif
-
- crc32_init(); // always init it, to avoid multithread race conditions
-
- if (get8_packet(f) != VORBIS_packet_setup) return error(f, VORBIS_invalid_setup);
- for (i=0; i < 6; ++i) header[i] = get8_packet(f);
- if (!vorbis_validate(header)) return error(f, VORBIS_invalid_setup);
-
- // codebooks
-
- f->codebook_count = get_bits(f,8) + 1;
- f->codebooks = (Codebook *) setup_malloc(f, sizeof(*f->codebooks) * f->codebook_count);
- if (f->codebooks == NULL) return error(f, VORBIS_outofmem);
- memset(f->codebooks, 0, sizeof(*f->codebooks) * f->codebook_count);
- for (i=0; i < f->codebook_count; ++i) {
- uint32 *values;
- int ordered, sorted_count;
- int total=0;
- uint8 *lengths;
- Codebook *c = f->codebooks+i;
- x = get_bits(f, 8); if (x != 0x42) return error(f, VORBIS_invalid_setup);
- x = get_bits(f, 8); if (x != 0x43) return error(f, VORBIS_invalid_setup);
- x = get_bits(f, 8); if (x != 0x56) return error(f, VORBIS_invalid_setup);
- x = get_bits(f, 8);
- c->dimensions = (get_bits(f, 8)<<8) + x;
- x = get_bits(f, 8);
- y = get_bits(f, 8);
- c->entries = (get_bits(f, 8)<<16) + (y<<8) + x;
- ordered = get_bits(f,1);
- c->sparse = ordered ? 0 : get_bits(f,1);
-
- if (c->sparse)
- lengths = (uint8 *) setup_temp_malloc(f, c->entries);
- else
- lengths = c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries);
-
- if (!lengths) return error(f, VORBIS_outofmem);
-
- if (ordered) {
- int current_entry = 0;
- int current_length = get_bits(f,5) + 1;
- while (current_entry < c->entries) {
- int limit = c->entries - current_entry;
- int n = get_bits(f, ilog(limit));
- if (current_entry + n > (int) c->entries) { return error(f, VORBIS_invalid_setup); }
- memset(lengths + current_entry, current_length, n);
- current_entry += n;
- ++current_length;
- }
- } else {
- for (j=0; j < c->entries; ++j) {
- int present = c->sparse ? get_bits(f,1) : 1;
- if (present) {
- lengths[j] = get_bits(f, 5) + 1;
- ++total;
- } else {
- lengths[j] = NO_CODE;
- }
- }
- }
-
- if (c->sparse && total >= c->entries >> 2) {
- // convert sparse items to non-sparse!
- if (c->entries > (int) f->setup_temp_memory_required)
- f->setup_temp_memory_required = c->entries;
-
- c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries);
- memcpy(c->codeword_lengths, lengths, c->entries);
- setup_temp_free(f, lengths, c->entries); // note this is only safe if there have been no intervening temp mallocs!
- lengths = c->codeword_lengths;
- c->sparse = 0;
- }
-
- // compute the size of the sorted tables
- if (c->sparse) {
- sorted_count = total;
- //assert(total != 0);
- } else {
- sorted_count = 0;
- #ifndef STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
- for (j=0; j < c->entries; ++j)
- if (lengths[j] > STB_VORBIS_FAST_HUFFMAN_LENGTH && lengths[j] != NO_CODE)
- ++sorted_count;
- #endif
- }
-
- c->sorted_entries = sorted_count;
- values = NULL;
-
- if (!c->sparse) {
- c->codewords = (uint32 *) setup_malloc(f, sizeof(c->codewords[0]) * c->entries);
- if (!c->codewords) return error(f, VORBIS_outofmem);
- } else {
- unsigned int size;
- if (c->sorted_entries) {
- c->codeword_lengths = (uint8 *) setup_malloc(f, c->sorted_entries);
- if (!c->codeword_lengths) return error(f, VORBIS_outofmem);
- c->codewords = (uint32 *) setup_temp_malloc(f, sizeof(*c->codewords) * c->sorted_entries);
- if (!c->codewords) return error(f, VORBIS_outofmem);
- values = (uint32 *) setup_temp_malloc(f, sizeof(*values) * c->sorted_entries);
- if (!values) return error(f, VORBIS_outofmem);
- }
- size = c->entries + (sizeof(*c->codewords) + sizeof(*values)) * c->sorted_entries;
- if (size > f->setup_temp_memory_required)
- f->setup_temp_memory_required = size;
- }
-
- if (!compute_codewords(c, lengths, c->entries, values)) {
- if (c->sparse) setup_temp_free(f, values, 0);
- return error(f, VORBIS_invalid_setup);
- }
-
- if (c->sorted_entries) {
- // allocate an extra slot for sentinels
- c->sorted_codewords = (uint32 *) setup_malloc(f, sizeof(*c->sorted_codewords) * (c->sorted_entries+1));
- // allocate an extra slot at the front so that c->sorted_values[-1] is defined
- // so that we can catch that case without an extra if
- c->sorted_values = ( int *) setup_malloc(f, sizeof(*c->sorted_values ) * (c->sorted_entries+1));
- if (c->sorted_values) { ++c->sorted_values; c->sorted_values[-1] = -1; }
- compute_sorted_huffman(c, lengths, values);
- }
-
- if (c->sparse) {
- setup_temp_free(f, values, sizeof(*values)*c->sorted_entries);
- setup_temp_free(f, c->codewords, sizeof(*c->codewords)*c->sorted_entries);
- setup_temp_free(f, lengths, c->entries);
- c->codewords = NULL;
- }
-
- compute_accelerated_huffman(c);
-
- c->lookup_type = get_bits(f, 4);
- if (c->lookup_type > 2) return error(f, VORBIS_invalid_setup);
- if (c->lookup_type > 0) {
- uint16 *mults;
- c->minimum_value = float32_unpack(get_bits(f, 32));
- c->delta_value = float32_unpack(get_bits(f, 32));
- c->value_bits = get_bits(f, 4)+1;
- c->sequence_p = get_bits(f,1);
- if (c->lookup_type == 1) {
- c->lookup_values = lookup1_values(c->entries, c->dimensions);
- } else {
- c->lookup_values = c->entries * c->dimensions;
- }
- mults = (uint16 *) setup_temp_malloc(f, sizeof(mults[0]) * c->lookup_values);
- if (mults == NULL) return error(f, VORBIS_outofmem);
- for (j=0; j < (int) c->lookup_values; ++j) {
- int q = get_bits(f, c->value_bits);
- if (q == EOP) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_invalid_setup); }
- mults[j] = q;
- }
-
-#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
- if (c->lookup_type == 1) {
- int len, sparse = c->sparse;
- // pre-expand the lookup1-style multiplicands, to avoid a divide in the inner loop
- if (sparse) {
- if (c->sorted_entries == 0) goto skip;
- c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->sorted_entries * c->dimensions);
- } else
- c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->entries * c->dimensions);
- if (c->multiplicands == NULL) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); }
- len = sparse ? c->sorted_entries : c->entries;
- for (j=0; j < len; ++j) {
- int z = sparse ? c->sorted_values[j] : j, div=1;
- for (k=0; k < c->dimensions; ++k) {
- int off = (z / div) % c->lookup_values;
- c->multiplicands[j*c->dimensions + k] =
- #ifndef STB_VORBIS_CODEBOOK_FLOATS
- mults[off];
- #else
- mults[off]*c->delta_value + c->minimum_value;
- // in this case (and this case only) we could pre-expand c->sequence_p,
- // and throw away the decode logic for it; have to ALSO do
- // it in the case below, but it can only be done if
- // STB_VORBIS_CODEBOOK_FLOATS
- // !STB_VORBIS_DIVIDES_IN_CODEBOOK
- #endif
- div *= c->lookup_values;
- }
- }
- setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values);
- c->lookup_type = 2;
- }
- else
-#endif
- {
- c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values);
- #ifndef STB_VORBIS_CODEBOOK_FLOATS
- memcpy(c->multiplicands, mults, sizeof(c->multiplicands[0]) * c->lookup_values);
- #else
- for (j=0; j < (int) c->lookup_values; ++j)
- c->multiplicands[j] = mults[j] * c->delta_value + c->minimum_value;
- #endif
- setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values);
- }
- skip:;
-
- #ifdef STB_VORBIS_CODEBOOK_FLOATS
- if (c->lookup_type == 2 && c->sequence_p) {
- for (j=1; j < (int) c->lookup_values; ++j)
- c->multiplicands[j] = c->multiplicands[j-1];
- c->sequence_p = 0;
- }
- #endif
- }
- }
-
- // time domain transfers (notused)
-
- x = get_bits(f, 6) + 1;
- for (i=0; i < x; ++i) {
- uint32 z = get_bits(f, 16);
- if (z != 0) return error(f, VORBIS_invalid_setup);
- }
-
- // Floors
- f->floor_count = get_bits(f, 6)+1;
- f->floor_config = (Floor *) setup_malloc(f, f->floor_count * sizeof(*f->floor_config));
- for (i=0; i < f->floor_count; ++i) {
- f->floor_types[i] = get_bits(f, 16);
- if (f->floor_types[i] > 1) return error(f, VORBIS_invalid_setup);
- if (f->floor_types[i] == 0) {
- Floor0 *g = &f->floor_config[i].floor0;
- g->order = get_bits(f,8);
- g->rate = get_bits(f,16);
- g->bark_map_size = get_bits(f,16);
- g->amplitude_bits = get_bits(f,6);
- g->amplitude_offset = get_bits(f,8);
- g->number_of_books = get_bits(f,4) + 1;
- for (j=0; j < g->number_of_books; ++j)
- g->book_list[j] = get_bits(f,8);
- return error(f, VORBIS_feature_not_supported);
- } else {
- Point p[31*8+2];
- Floor1 *g = &f->floor_config[i].floor1;
- int max_class = -1;
- g->partitions = get_bits(f, 5);
- for (j=0; j < g->partitions; ++j) {
- g->partition_class_list[j] = get_bits(f, 4);
- if (g->partition_class_list[j] > max_class)
- max_class = g->partition_class_list[j];
- }
- for (j=0; j <= max_class; ++j) {
- g->class_dimensions[j] = get_bits(f, 3)+1;
- g->class_subclasses[j] = get_bits(f, 2);
- if (g->class_subclasses[j]) {
- g->class_masterbooks[j] = get_bits(f, 8);
- if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
- }
- for (k=0; k < 1 << g->class_subclasses[j]; ++k) {
- g->subclass_books[j][k] = get_bits(f,8)-1;
- if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
- }
- }
- g->floor1_multiplier = get_bits(f,2)+1;
- g->rangebits = get_bits(f,4);
- g->Xlist[0] = 0;
- g->Xlist[1] = 1 << g->rangebits;
- g->values = 2;
- for (j=0; j < g->partitions; ++j) {
- int c = g->partition_class_list[j];
- for (k=0; k < g->class_dimensions[c]; ++k) {
- g->Xlist[g->values] = get_bits(f, g->rangebits);
- ++g->values;
- }
- }
- // precompute the sorting
- for (j=0; j < g->values; ++j) {
- p[j].x = g->Xlist[j];
- p[j].y = j;
- }
- qsort(p, g->values, sizeof(p[0]), point_compare);
- for (j=0; j < g->values; ++j)
- g->sorted_order[j] = (uint8) p[j].y;
- // precompute the neighbors
- for (j=2; j < g->values; ++j) {
- int low,hi;
- neighbors(g->Xlist, j, &low,&hi);
- g->neighbors[j][0] = low;
- g->neighbors[j][1] = hi;
- }
-
- if (g->values > longest_floorlist)
- longest_floorlist = g->values;
- }
- }
-
- // Residue
- f->residue_count = get_bits(f, 6)+1;
- f->residue_config = (Residue *) setup_malloc(f, f->residue_count * sizeof(*f->residue_config));
- for (i=0; i < f->residue_count; ++i) {
- uint8 residue_cascade[64];
- Residue *r = f->residue_config+i;
- f->residue_types[i] = get_bits(f, 16);
- if (f->residue_types[i] > 2) return error(f, VORBIS_invalid_setup);
- r->begin = get_bits(f, 24);
- r->end = get_bits(f, 24);
- r->part_size = get_bits(f,24)+1;
- r->classifications = get_bits(f,6)+1;
- r->classbook = get_bits(f,8);
- for (j=0; j < r->classifications; ++j) {
- uint8 high_bits=0;
- uint8 low_bits=get_bits(f,3);
- if (get_bits(f,1))
- high_bits = get_bits(f,5);
- residue_cascade[j] = high_bits*8 + low_bits;
- }
- r->residue_books = (short (*)[8]) setup_malloc(f, sizeof(r->residue_books[0]) * r->classifications);
- for (j=0; j < r->classifications; ++j) {
- for (k=0; k < 8; ++k) {
- if (residue_cascade[j] & (1 << k)) {
- r->residue_books[j][k] = get_bits(f, 8);
- if (r->residue_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
- } else {
- r->residue_books[j][k] = -1;
- }
- }
- }
- // precompute the classifications[] array to avoid inner-loop mod/divide
- // call it 'classdata' since we already have r->classifications
- r->classdata = (uint8 **) setup_malloc(f, sizeof(*r->classdata) * f->codebooks[r->classbook].entries);
- if (!r->classdata) return error(f, VORBIS_outofmem);
- memset(r->classdata, 0, sizeof(*r->classdata) * f->codebooks[r->classbook].entries);
- for (j=0; j < f->codebooks[r->classbook].entries; ++j) {
- int classwords = f->codebooks[r->classbook].dimensions;
- int temp = j;
- r->classdata[j] = (uint8 *) setup_malloc(f, sizeof(r->classdata[j][0]) * classwords);
- for (k=classwords-1; k >= 0; --k) {
- r->classdata[j][k] = temp % r->classifications;
- temp /= r->classifications;
- }
- }
- }
-
- f->mapping_count = get_bits(f,6)+1;
- f->mapping = (Mapping *) setup_malloc(f, f->mapping_count * sizeof(*f->mapping));
- for (i=0; i < f->mapping_count; ++i) {
- Mapping *m = f->mapping + i;
- int mapping_type = get_bits(f,16);
- if (mapping_type != 0) return error(f, VORBIS_invalid_setup);
- m->chan = (MappingChannel *) setup_malloc(f, f->channels * sizeof(*m->chan));
- if (get_bits(f,1))
- m->submaps = get_bits(f,4);
- else
- m->submaps = 1;
- if (m->submaps > max_submaps)
- max_submaps = m->submaps;
- if (get_bits(f,1)) {
- m->coupling_steps = get_bits(f,8)+1;
- for (k=0; k < m->coupling_steps; ++k) {
- m->chan[k].magnitude = get_bits(f, ilog(f->channels)-1);
- m->chan[k].angle = get_bits(f, ilog(f->channels)-1);
- if (m->chan[k].magnitude >= f->channels) return error(f, VORBIS_invalid_setup);
- if (m->chan[k].angle >= f->channels) return error(f, VORBIS_invalid_setup);
- if (m->chan[k].magnitude == m->chan[k].angle) return error(f, VORBIS_invalid_setup);
- }
- } else
- m->coupling_steps = 0;
-
- // reserved field
- if (get_bits(f,2)) return error(f, VORBIS_invalid_setup);
- if (m->submaps > 1) {
- for (j=0; j < f->channels; ++j) {
- m->chan[j].mux = get_bits(f, 4);
- if (m->chan[j].mux >= m->submaps) return error(f, VORBIS_invalid_setup);
- }
- } else
- // @SPECIFICATION: this case is missing from the spec
- for (j=0; j < f->channels; ++j)
- m->chan[j].mux = 0;
-
- for (j=0; j < m->submaps; ++j) {
- get_bits(f,8); // discard
- m->submap_floor[j] = get_bits(f,8);
- m->submap_residue[j] = get_bits(f,8);
- if (m->submap_floor[j] >= f->floor_count) return error(f, VORBIS_invalid_setup);
- if (m->submap_residue[j] >= f->residue_count) return error(f, VORBIS_invalid_setup);
- }
- }
-
- // Modes
- f->mode_count = get_bits(f, 6)+1;
- for (i=0; i < f->mode_count; ++i) {
- Mode *m = f->mode_config+i;
- m->blockflag = get_bits(f,1);
- m->windowtype = get_bits(f,16);
- m->transformtype = get_bits(f,16);
- m->mapping = get_bits(f,8);
- if (m->windowtype != 0) return error(f, VORBIS_invalid_setup);
- if (m->transformtype != 0) return error(f, VORBIS_invalid_setup);
- if (m->mapping >= f->mapping_count) return error(f, VORBIS_invalid_setup);
- }
-
- flush_packet(f);
-
- f->previous_length = 0;
-
- for (i=0; i < f->channels; ++i) {
- f->channel_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1);
- f->previous_window[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2);
- f->finalY[i] = (int16 *) setup_malloc(f, sizeof(int16) * longest_floorlist);
- #ifdef STB_VORBIS_NO_DEFER_FLOOR
- f->floor_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2);
- #endif
- }
-
- if (!init_blocksize(f, 0, f->blocksize_0)) return FALSE;
- if (!init_blocksize(f, 1, f->blocksize_1)) return FALSE;
- f->blocksize[0] = f->blocksize_0;
- f->blocksize[1] = f->blocksize_1;
-
-#ifdef STB_VORBIS_DIVIDE_TABLE
- if (integer_divide_table[1][1]==0)
- for (i=0; i < DIVTAB_NUMER; ++i)
- for (j=1; j < DIVTAB_DENOM; ++j)
- integer_divide_table[i][j] = i / j;
-#endif
-
- // compute how much temporary memory is needed
-
- // 1.
- {
- uint32 imdct_mem = (f->blocksize_1 * sizeof(float) >> 1);
- uint32 classify_mem;
- int i,max_part_read=0;
- for (i=0; i < f->residue_count; ++i) {
- Residue *r = f->residue_config + i;
- int n_read = r->end - r->begin;
- int part_read = n_read / r->part_size;
- if (part_read > max_part_read)
- max_part_read = part_read;
- }
- #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
- classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(uint8 *));
- #else
- classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(int *));
- #endif
-
- f->temp_memory_required = classify_mem;
- if (imdct_mem > f->temp_memory_required)
- f->temp_memory_required = imdct_mem;
- }
-
- f->first_decode = TRUE;
-
- if (f->alloc.alloc_buffer) {
- assert(f->temp_offset == f->alloc.alloc_buffer_length_in_bytes);
- // check if there's enough temp memory so we don't error later
- if (f->setup_offset + sizeof(*f) + f->temp_memory_required > (unsigned) f->temp_offset)
- return error(f, VORBIS_outofmem);
- }
-
- f->first_audio_page_offset = stb_vorbis_get_file_offset(f);
-
- return TRUE;
-}
-
-static void vorbis_deinit(stb_vorbis *p)
-{
- int i,j;
- for (i=0; i < p->residue_count; ++i) {
- Residue *r = p->residue_config+i;
- if (r->classdata) {
- for (j=0; j < p->codebooks[r->classbook].entries; ++j)
- setup_free(p, r->classdata[j]);
- setup_free(p, r->classdata);
- }
- setup_free(p, r->residue_books);
- }
-
- if (p->codebooks) {
- for (i=0; i < p->codebook_count; ++i) {
- Codebook *c = p->codebooks + i;
- setup_free(p, c->codeword_lengths);
- setup_free(p, c->multiplicands);
- setup_free(p, c->codewords);
- setup_free(p, c->sorted_codewords);
- // c->sorted_values[-1] is the first entry in the array
- setup_free(p, c->sorted_values ? c->sorted_values-1 : NULL);
- }
- setup_free(p, p->codebooks);
- }
- setup_free(p, p->floor_config);
- setup_free(p, p->residue_config);
- for (i=0; i < p->mapping_count; ++i)
- setup_free(p, p->mapping[i].chan);
- setup_free(p, p->mapping);
- for (i=0; i < p->channels; ++i) {
- setup_free(p, p->channel_buffers[i]);
- setup_free(p, p->previous_window[i]);
- #ifdef STB_VORBIS_NO_DEFER_FLOOR
- setup_free(p, p->floor_buffers[i]);
- #endif
- setup_free(p, p->finalY[i]);
- }
- for (i=0; i < 2; ++i) {
- setup_free(p, p->A[i]);
- setup_free(p, p->B[i]);
- setup_free(p, p->C[i]);
- setup_free(p, p->window[i]);
- setup_free(p, p->bit_reverse[i]);
- }
- #ifndef STB_VORBIS_NO_STDIO
- if (p->close_on_free) fclose(p->f);
- #endif
-}
-
-void stb_vorbis_close(stb_vorbis *p)
-{
- if (p == NULL) return;
- vorbis_deinit(p);
- setup_free(p,p);
-}
-
-static void vorbis_init(stb_vorbis *p, stb_vorbis_alloc *z)
-{
- memset(p, 0, sizeof(*p)); // NULL out all malloc'd pointers to start
- if (z) {
- p->alloc = *z;
- p->alloc.alloc_buffer_length_in_bytes = (p->alloc.alloc_buffer_length_in_bytes+3) & ~3;
- p->temp_offset = p->alloc.alloc_buffer_length_in_bytes;
- }
- p->eof = 0;
- p->error = VORBIS__no_error;
- p->stream = NULL;
- p->codebooks = NULL;
- p->page_crc_tests = -1;
- #ifndef STB_VORBIS_NO_STDIO
- p->close_on_free = FALSE;
- p->f = NULL;
- #endif
-}
-
-int stb_vorbis_get_sample_offset(stb_vorbis *f)
-{
- if (f->current_loc_valid)
- return f->current_loc;
- else
- return -1;
-}
-
-stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f)
-{
- stb_vorbis_info d;
- d.channels = f->channels;
- d.sample_rate = f->sample_rate;
- d.setup_memory_required = f->setup_memory_required;
- d.setup_temp_memory_required = f->setup_temp_memory_required;
- d.temp_memory_required = f->temp_memory_required;
- d.max_frame_size = f->blocksize_1 >> 1;
- return d;
-}
-
-int stb_vorbis_get_error(stb_vorbis *f)
-{
- int e = f->error;
- f->error = VORBIS__no_error;
- return e;
-}
-
-static stb_vorbis * vorbis_alloc(stb_vorbis *f)
-{
- stb_vorbis *p = (stb_vorbis *) setup_malloc(f, sizeof(*p));
- return p;
-}
-
-#ifndef STB_VORBIS_NO_PUSHDATA_API
-
-void stb_vorbis_flush_pushdata(stb_vorbis *f)
-{
- f->previous_length = 0;
- f->page_crc_tests = 0;
- f->discard_samples_deferred = 0;
- f->current_loc_valid = FALSE;
- f->first_decode = FALSE;
- f->samples_output = 0;
- f->channel_buffer_start = 0;
- f->channel_buffer_end = 0;
-}
-
-static int vorbis_search_for_page_pushdata(vorb *f, uint8 *data, int data_len)
-{
- int i,n;
- for (i=0; i < f->page_crc_tests; ++i)
- f->scan[i].bytes_done = 0;
-
- // if we have room for more scans, search for them first, because
- // they may cause us to stop early if their header is incomplete
- if (f->page_crc_tests < STB_VORBIS_PUSHDATA_CRC_COUNT) {
- if (data_len < 4) return 0;
- data_len -= 3; // need to look for 4-byte sequence, so don't miss
- // one that straddles a boundary
- for (i=0; i < data_len; ++i) {
- if (data[i] == 0x4f) {
- if (0==memcmp(data+i, ogg_page_header, 4)) {
- int j,len;
- uint32 crc;
- // make sure we have the whole page header
- if (i+26 >= data_len || i+27+data[i+26] >= data_len) {
- // only read up to this page start, so hopefully we'll
- // have the whole page header start next time
- data_len = i;
- break;
- }
- // ok, we have it all; compute the length of the page
- len = 27 + data[i+26];
- for (j=0; j < data[i+26]; ++j)
- len += data[i+27+j];
- // scan everything up to the embedded crc (which we must 0)
- crc = 0;
- for (j=0; j < 22; ++j)
- crc = crc32_update(crc, data[i+j]);
- // now process 4 0-bytes
- for ( ; j < 26; ++j)
- crc = crc32_update(crc, 0);
- // len is the total number of bytes we need to scan
- n = f->page_crc_tests++;
- f->scan[n].bytes_left = len-j;
- f->scan[n].crc_so_far = crc;
- f->scan[n].goal_crc = data[i+22] + (data[i+23] << 8) + (data[i+24]<<16) + (data[i+25]<<24);
- // if the last frame on a page is continued to the next, then
- // we can't recover the sample_loc immediately
- if (data[i+27+data[i+26]-1] == 255)
- f->scan[n].sample_loc = ~0;
- else
- f->scan[n].sample_loc = data[i+6] + (data[i+7] << 8) + (data[i+ 8]<<16) + (data[i+ 9]<<24);
- f->scan[n].bytes_done = i+j;
- if (f->page_crc_tests == STB_VORBIS_PUSHDATA_CRC_COUNT)
- break;
- // keep going if we still have room for more
- }
- }
- }
- }
-
- for (i=0; i < f->page_crc_tests;) {
- uint32 crc;
- int j;
- int n = f->scan[i].bytes_done;
- int m = f->scan[i].bytes_left;
- if (m > data_len - n) m = data_len - n;
- // m is the bytes to scan in the current chunk
- crc = f->scan[i].crc_so_far;
- for (j=0; j < m; ++j)
- crc = crc32_update(crc, data[n+j]);
- f->scan[i].bytes_left -= m;
- f->scan[i].crc_so_far = crc;
- if (f->scan[i].bytes_left == 0) {
- // does it match?
- if (f->scan[i].crc_so_far == f->scan[i].goal_crc) {
- // Houston, we have page
- data_len = n+m; // consumption amount is wherever that scan ended
- f->page_crc_tests = -1; // drop out of page scan mode
- f->previous_length = 0; // decode-but-don't-output one frame
- f->next_seg = -1; // start a new page
- f->current_loc = f->scan[i].sample_loc; // set the current sample location
- // to the amount we'd have decoded had we decoded this page
- f->current_loc_valid = f->current_loc != ~0;
- return data_len;
- }
- // delete entry
- f->scan[i] = f->scan[--f->page_crc_tests];
- } else {
- ++i;
- }
- }
-
- return data_len;
-}
-
-// return value: number of bytes we used
-int stb_vorbis_decode_frame_pushdata(
- stb_vorbis *f, // the file we're decoding
- uint8 *data, int data_len, // the memory available for decoding
- int *channels, // place to write number of float * buffers
- float ***output, // place to write float ** array of float * buffers
- int *samples // place to write number of output samples
- )
-{
- int i;
- int len,right,left;
-
- if (!IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
-
- if (f->page_crc_tests >= 0) {
- *samples = 0;
- return vorbis_search_for_page_pushdata(f, data, data_len);
- }
-
- f->stream = data;
- f->stream_end = data + data_len;
- f->error = VORBIS__no_error;
-
- // check that we have the entire packet in memory
- if (!is_whole_packet_present(f, FALSE)) {
- *samples = 0;
- return 0;
- }
-
- if (!vorbis_decode_packet(f, &len, &left, &right)) {
- // save the actual error we encountered
- enum STBVorbisError error = f->error;
- if (error == VORBIS_bad_packet_type) {
- // flush and resynch
- f->error = VORBIS__no_error;
- while (get8_packet(f) != EOP)
- if (f->eof) break;
- *samples = 0;
- return f->stream - data;
- }
- if (error == VORBIS_continued_packet_flag_invalid) {
- if (f->previous_length == 0) {
- // we may be resynching, in which case it's ok to hit one
- // of these; just discard the packet
- f->error = VORBIS__no_error;
- while (get8_packet(f) != EOP)
- if (f->eof) break;
- *samples = 0;
- return f->stream - data;
- }
- }
- // if we get an error while parsing, what to do?
- // well, it DEFINITELY won't work to continue from where we are!
- stb_vorbis_flush_pushdata(f);
- // restore the error that actually made us bail
- f->error = error;
- *samples = 0;
- return 1;
- }
-
- // success!
- len = vorbis_finish_frame(f, len, left, right);
- for (i=0; i < f->channels; ++i)
- f->outputs[i] = f->channel_buffers[i] + left;
-
- if (channels) *channels = f->channels;
- *samples = len;
- *output = f->outputs;
- return f->stream - data;
-}
-
-stb_vorbis *stb_vorbis_open_pushdata(
- unsigned char *data, int data_len, // the memory available for decoding
- int *data_used, // only defined if result is not NULL
- int *error, stb_vorbis_alloc *alloc)
-{
- stb_vorbis *f, p;
- vorbis_init(&p, alloc);
- p.stream = data;
- p.stream_end = data + data_len;
- p.push_mode = TRUE;
- if (!start_decoder(&p)) {
- if (p.eof)
- *error = VORBIS_need_more_data;
- else
- *error = p.error;
- return NULL;
- }
- f = vorbis_alloc(&p);
- if (f) {
- *f = p;
- *data_used = f->stream - data;
- *error = 0;
- return f;
- } else {
- vorbis_deinit(&p);
- return NULL;
- }
-}
-#endif // STB_VORBIS_NO_PUSHDATA_API
-
-unsigned int stb_vorbis_get_file_offset(stb_vorbis *f)
-{
- #ifndef STB_VORBIS_NO_PUSHDATA_API
- if (f->push_mode) return 0;
- #endif
- if (USE_MEMORY(f)) return f->stream - f->stream_start;
- #ifndef STB_VORBIS_NO_STDIO
- return ftell(f->f) - f->f_start;
- #endif
-}
-
-#ifndef STB_VORBIS_NO_PULLDATA_API
-//
-// DATA-PULLING API
-//
-
-static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last)
-{
- for(;;) {
- int n;
- if (f->eof) return 0;
- n = get8(f);
- if (n == 0x4f) { // page header
- unsigned int retry_loc = stb_vorbis_get_file_offset(f);
- int i;
- // check if we're off the end of a file_section stream
- if (retry_loc - 25 > f->stream_len)
- return 0;
- // check the rest of the header
- for (i=1; i < 4; ++i)
- if (get8(f) != ogg_page_header[i])
- break;
- if (f->eof) return 0;
- if (i == 4) {
- uint8 header[27];
- uint32 i, crc, goal, len;
- for (i=0; i < 4; ++i)
- header[i] = ogg_page_header[i];
- for (; i < 27; ++i)
- header[i] = get8(f);
- if (f->eof) return 0;
- if (header[4] != 0) goto invalid;
- goal = header[22] + (header[23] << 8) + (header[24]<<16) + (header[25]<<24);
- for (i=22; i < 26; ++i)
- header[i] = 0;
- crc = 0;
- for (i=0; i < 27; ++i)
- crc = crc32_update(crc, header[i]);
- len = 0;
- for (i=0; i < header[26]; ++i) {
- int s = get8(f);
- crc = crc32_update(crc, s);
- len += s;
- }
- if (len && f->eof) return 0;
- for (i=0; i < len; ++i)
- crc = crc32_update(crc, get8(f));
- // finished parsing probable page
- if (crc == goal) {
- // we could now check that it's either got the last
- // page flag set, OR it's followed by the capture
- // pattern, but I guess TECHNICALLY you could have
- // a file with garbage between each ogg page and recover
- // from it automatically? So even though that paranoia
- // might decrease the chance of an invalid decode by
- // another 2^32, not worth it since it would hose those
- // invalid-but-useful files?
- if (end)
- *end = stb_vorbis_get_file_offset(f);
- if (last)
- if (header[5] & 0x04)
- *last = 1;
- else
- *last = 0;
- set_file_offset(f, retry_loc-1);
- return 1;
- }
- }
- invalid:
- // not a valid page, so rewind and look for next one
- set_file_offset(f, retry_loc);
- }
- }
-}
-
-// seek is implemented with 'interpolation search'--this is like
-// binary search, but we use the data values to estimate the likely
-// location of the data item (plus a bit of a bias so when the
-// estimation is wrong we don't waste overly much time)
-
-#define SAMPLE_unknown 0xffffffff
-
-
-// ogg vorbis, in its insane infinite wisdom, only provides
-// information about the sample at the END of the page.
-// therefore we COULD have the data we need in the current
-// page, and not know it. we could just use the end location
-// as our only knowledge for bounds, seek back, and eventually
-// the binary search finds it. or we can try to be smart and
-// not waste time trying to locate more pages. we try to be
-// smart, since this data is already in memory anyway, so
-// doing needless I/O would be crazy!
-static int vorbis_analyze_page(stb_vorbis *f, ProbedPage *z)
-{
- uint8 header[27], lacing[255];
- uint8 packet_type[255];
- int num_packet, packet_start, previous =0;
- int i,len;
- uint32 samples;
-
- // record where the page starts
- z->page_start = stb_vorbis_get_file_offset(f);
-
- // parse the header
- getn(f, header, 27);
- assert(header[0] == 'O' && header[1] == 'g' && header[2] == 'g' && header[3] == 'S');
- getn(f, lacing, header[26]);
-
- // determine the length of the payload
- len = 0;
- for (i=0; i < header[26]; ++i)
- len += lacing[i];
-
- // this implies where the page ends
- z->page_end = z->page_start + 27 + header[26] + len;
-
- // read the last-decoded sample out of the data
- z->last_decoded_sample = header[6] + (header[7] << 8) + (header[8] << 16) + (header[9] << 16);
-
- if (header[5] & 4) {
- // if this is the last page, it's not possible to work
- // backwards to figure out the first sample! whoops! fuck.
- z->first_decoded_sample = SAMPLE_unknown;
- set_file_offset(f, z->page_start);
- return 1;
- }
-
- // scan through the frames to determine the sample-count of each one...
- // our goal is the sample # of the first fully-decoded sample on the
- // page, which is the first decoded sample of the 2nd page
-
- num_packet=0;
-
- packet_start = ((header[5] & 1) == 0);
-
- for (i=0; i < header[26]; ++i) {
- if (packet_start) {
- uint8 n,b,m;
- if (lacing[i] == 0) goto bail; // trying to read from zero-length packet
- n = get8(f);
- // if bottom bit is non-zero, we've got corruption
- if (n & 1) goto bail;
- n >>= 1;
- b = ilog(f->mode_count-1);
- m = n >> b;
- n &= (1 << b)-1;
- if (n >= f->mode_count) goto bail;
- if (num_packet == 0 && f->mode_config[n].blockflag)
- previous = (m & 1);
- packet_type[num_packet++] = f->mode_config[n].blockflag;
- skip(f, lacing[i]-1);
- } else
- skip(f, lacing[i]);
- packet_start = (lacing[i] < 255);
- }
-
- // now that we know the sizes of all the pages, we can start determining
- // how much sample data there is.
-
- samples = 0;
-
- // for the last packet, we step by its whole length, because the definition
- // is that we encoded the end sample loc of the 'last packet completed',
- // where 'completed' refers to packets being split, and we are left to guess
- // what 'end sample loc' means. we assume it means ignoring the fact that
- // the last half of the data is useless without windowing against the next
- // packet... (so it's not REALLY complete in that sense)
- if (num_packet > 1)
- samples += f->blocksize[packet_type[num_packet-1]];
-
- for (i=num_packet-2; i >= 1; --i) {
- // now, for this packet, how many samples do we have that
- // do not overlap the following packet?
- if (packet_type[i] == 1)
- if (packet_type[i+1] == 1)
- samples += f->blocksize_1 >> 1;
- else
- samples += ((f->blocksize_1 - f->blocksize_0) >> 2) + (f->blocksize_0 >> 1);
- else
- samples += f->blocksize_0 >> 1;
- }
- // now, at this point, we've rewound to the very beginning of the
- // _second_ packet. if we entirely discard the first packet after
- // a seek, this will be exactly the right sample number. HOWEVER!
- // we can't as easily compute this number for the LAST page. The
- // only way to get the sample offset of the LAST page is to use
- // the end loc from the previous page. But what that returns us
- // is _exactly_ the place where we get our first non-overlapped
- // sample. (I think. Stupid spec for being ambiguous.) So for
- // consistency it's better to do that here, too. However, that
- // will then require us to NOT discard all of the first frame we
- // decode, in some cases, which means an even weirder frame size
- // and extra code. what a fucking pain.
-
- // we're going to discard the first packet if we
- // start the seek here, so we don't care about it. (we could actually
- // do better; if the first packet is long, and the previous packet
- // is short, there's actually data in the first half of the first
- // packet that doesn't need discarding... but not worth paying the
- // effort of tracking that of that here and in the seeking logic)
- // except crap, if we infer it from the _previous_ packet's end
- // location, we DO need to use that definition... and we HAVE to
- // infer the start loc of the LAST packet from the previous packet's
- // end location. fuck you, ogg vorbis.
-
- z->first_decoded_sample = z->last_decoded_sample - samples;
-
- // restore file state to where we were
- set_file_offset(f, z->page_start);
- return 1;
-
- // restore file state to where we were
- bail:
- set_file_offset(f, z->page_start);
- return 0;
-}
-
-static int vorbis_seek_frame_from_page(stb_vorbis *f, uint32 page_start, uint32 first_sample, uint32 target_sample, int fine)
-{
- int left_start, left_end, right_start, right_end, mode,i;
- int frame=0;
- uint32 frame_start;
- int frames_to_skip, data_to_skip;
-
- // first_sample is the sample # of the first sample that doesn't
- // overlap the previous page... note that this requires us to
- // _partially_ discard the first packet! bleh.
- set_file_offset(f, page_start);
-
- f->next_seg = -1; // force page resync
-
- frame_start = first_sample;
- // frame start is where the previous packet's last decoded sample
- // was, which corresponds to left_end... EXCEPT if the previous
- // packet was long and this packet is short? Probably a bug here.
-
-
- // now, we can start decoding frames... we'll only FAKE decode them,
- // until we find the frame that contains our sample; then we'll rewind,
- // and try again
- for (;;) {
- int start;
-
- if (!vorbis_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode))
- return error(f, VORBIS_seek_failed);
-
- if (frame == 0)
- start = left_end;
- else
- start = left_start;
-
- // the window starts at left_start; the last valid sample we generate
- // before the next frame's window start is right_start-1
- if (target_sample < frame_start + right_start-start)
- break;
-
- flush_packet(f);
- if (f->eof)
- return error(f, VORBIS_seek_failed);
-
- frame_start += right_start - start;
-
- ++frame;
- }
-
- // ok, at this point, the sample we want is contained in frame #'frame'
-
- // to decode frame #'frame' normally, we have to decode the
- // previous frame first... but if it's the FIRST frame of the page
- // we can't. if it's the first frame, it means it falls in the part
- // of the first frame that doesn't overlap either of the other frames.
- // so, if we have to handle that case for the first frame, we might
- // as well handle it for all of them, so:
- if (target_sample > frame_start + (left_end - left_start)) {
- // so what we want to do is go ahead and just immediately decode
- // this frame, but then make it so the next get_frame_float() uses
- // this already-decoded data? or do we want to go ahead and rewind,
- // and leave a flag saying to skip the first N data? let's do that
- frames_to_skip = frame; // if this is frame #1, skip 1 frame (#0)
- data_to_skip = left_end - left_start;
- } else {
- // otherwise, we want to skip frames 0, 1, 2, ... frame-2
- // (which means frame-2+1 total frames) then decode frame-1,
- // then leave frame pending
- frames_to_skip = frame - 1;
- assert(frames_to_skip >= 0);
- data_to_skip = -1;
- }
-
- set_file_offset(f, page_start);
- f->next_seg = - 1; // force page resync
-
- for (i=0; i < frames_to_skip; ++i) {
- maybe_start_packet(f);
- flush_packet(f);
- }
-
- if (data_to_skip >= 0) {
- int i,j,n = f->blocksize_0 >> 1;
- f->discard_samples_deferred = data_to_skip;
- for (i=0; i < f->channels; ++i)
- for (j=0; j < n; ++j)
- f->previous_window[i][j] = 0;
- f->previous_length = n;
- frame_start += data_to_skip;
- } else {
- f->previous_length = 0;
- vorbis_pump_first_frame(f);
- }
-
- // at this point, the NEXT decoded frame will generate the desired sample
- if (fine) {
- // so if we're doing sample accurate streaming, we want to go ahead and decode it!
- if (target_sample != frame_start) {
- int n;
- stb_vorbis_get_frame_float(f, &n, NULL);
- assert(target_sample > frame_start);
- assert(f->channel_buffer_start + (int) (target_sample-frame_start) < f->channel_buffer_end);
- f->channel_buffer_start += (target_sample - frame_start);
- }
- }
-
- return 0;
-}
-
-static int vorbis_seek_base(stb_vorbis *f, unsigned int sample_number, int fine)
-{
- ProbedPage p[2],q;
- if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
-
- // do we know the location of the last page?
- if (f->p_last.page_start == 0) {
- uint32 z = stb_vorbis_stream_length_in_samples(f);
- if (z == 0) return error(f, VORBIS_cant_find_last_page);
- }
-
- p[0] = f->p_first;
- p[1] = f->p_last;
-
- if (sample_number >= f->p_last.last_decoded_sample)
- sample_number = f->p_last.last_decoded_sample-1;
-
- if (sample_number < f->p_first.last_decoded_sample) {
- vorbis_seek_frame_from_page(f, p[0].page_start, 0, sample_number, fine);
- return 0;
- } else {
- int attempts=0;
- while (p[0].page_end < p[1].page_start) {
- uint32 probe;
- uint32 start_offset, end_offset;
- uint32 start_sample, end_sample;
-
- // copy these into local variables so we can tweak them
- // if any are unknown
- start_offset = p[0].page_end;
- end_offset = p[1].after_previous_page_start; // an address known to seek to page p[1]
- start_sample = p[0].last_decoded_sample;
- end_sample = p[1].last_decoded_sample;
-
- // currently there is no such tweaking logic needed/possible?
- if (start_sample == SAMPLE_unknown || end_sample == SAMPLE_unknown)
- return error(f, VORBIS_seek_failed);
-
- // now we want to lerp between these for the target samples...
-
- // step 1: we need to bias towards the page start...
- if (start_offset + 4000 < end_offset)
- end_offset -= 4000;
-
- // now compute an interpolated search loc
- probe = start_offset + (int) floor((float) (end_offset - start_offset) / (end_sample - start_sample) * (sample_number - start_sample));
-
- // next we need to bias towards binary search...
- // code is a little wonky to allow for full 32-bit unsigned values
- if (attempts >= 4) {
- uint32 probe2 = start_offset + ((end_offset - start_offset) >> 1);
- if (attempts >= 8)
- probe = probe2;
- else if (probe < probe2)
- probe = probe + ((probe2 - probe) >> 1);
- else
- probe = probe2 + ((probe - probe2) >> 1);
- }
- ++attempts;
-
- set_file_offset(f, probe);
- if (!vorbis_find_page(f, NULL, NULL)) return error(f, VORBIS_seek_failed);
- if (!vorbis_analyze_page(f, &q)) return error(f, VORBIS_seek_failed);
- q.after_previous_page_start = probe;
-
- // it's possible we've just found the last page again
- if (q.page_start == p[1].page_start) {
- p[1] = q;
- continue;
- }
-
- if (sample_number < q.last_decoded_sample)
- p[1] = q;
- else
- p[0] = q;
- }
-
- if (p[0].last_decoded_sample <= sample_number && sample_number < p[1].last_decoded_sample) {
- vorbis_seek_frame_from_page(f, p[1].page_start, p[0].last_decoded_sample, sample_number, fine);
- return 0;
- }
- return error(f, VORBIS_seek_failed);
- }
-}
-
-int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number)
-{
- return vorbis_seek_base(f, sample_number, FALSE);
-}
-
-int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number)
-{
- return vorbis_seek_base(f, sample_number, TRUE);
-}
-
-void stb_vorbis_seek_start(stb_vorbis *f)
-{
- if (IS_PUSH_MODE(f)) { error(f, VORBIS_invalid_api_mixing); return; }
- set_file_offset(f, f->first_audio_page_offset);
- f->previous_length = 0;
- f->first_decode = TRUE;
- f->next_seg = -1;
- vorbis_pump_first_frame(f);
-}
-
-unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f)
-{
- unsigned int restore_offset, previous_safe;
- unsigned int end, last_page_loc;
-
- if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
- if (!f->total_samples) {
- int last;
- uint32 lo,hi;
- char header[6];
-
- // first, store the current decode position so we can restore it
- restore_offset = stb_vorbis_get_file_offset(f);
-
- // now we want to seek back 64K from the end (the last page must
- // be at most a little less than 64K, but let's allow a little slop)
- if (f->stream_len >= 65536 && f->stream_len-65536 >= f->first_audio_page_offset)
- previous_safe = f->stream_len - 65536;
- else
- previous_safe = f->first_audio_page_offset;
-
- set_file_offset(f, previous_safe);
- // previous_safe is now our candidate 'earliest known place that seeking
- // to will lead to the final page'
-
- if (!vorbis_find_page(f, &end, (int unsigned *)&last)) {
- // if we can't find a page, we're hosed!
- f->error = VORBIS_cant_find_last_page;
- f->total_samples = 0xffffffff;
- goto done;
- }
-
- // check if there are more pages
- last_page_loc = stb_vorbis_get_file_offset(f);
-
- // stop when the last_page flag is set, not when we reach eof;
- // this allows us to stop short of a 'file_section' end without
- // explicitly checking the length of the section
- while (!last) {
- set_file_offset(f, end);
- if (!vorbis_find_page(f, &end, (int unsigned *)&last)) {
- // the last page we found didn't have the 'last page' flag
- // set. whoops!
- break;
- }
- previous_safe = last_page_loc+1;
- last_page_loc = stb_vorbis_get_file_offset(f);
- }
-
- set_file_offset(f, last_page_loc);
-
- // parse the header
- getn(f, (unsigned char *)header, 6);
- // extract the absolute granule position
- lo = get32(f);
- hi = get32(f);
- if (lo == 0xffffffff && hi == 0xffffffff) {
- f->error = VORBIS_cant_find_last_page;
- f->total_samples = SAMPLE_unknown;
- goto done;
- }
- if (hi)
- lo = 0xfffffffe; // saturate
- f->total_samples = lo;
-
- f->p_last.page_start = last_page_loc;
- f->p_last.page_end = end;
- f->p_last.last_decoded_sample = lo;
- f->p_last.first_decoded_sample = SAMPLE_unknown;
- f->p_last.after_previous_page_start = previous_safe;
-
- done:
- set_file_offset(f, restore_offset);
- }
- return f->total_samples == SAMPLE_unknown ? 0 : f->total_samples;
-}
-
-float stb_vorbis_stream_length_in_seconds(stb_vorbis *f)
-{
- return stb_vorbis_stream_length_in_samples(f) / (float) f->sample_rate;
-}
-
-
-
-int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output)
-{
- int len, right,left,i;
- if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
-
- if (!vorbis_decode_packet(f, &len, &left, &right)) {
- f->channel_buffer_start = f->channel_buffer_end = 0;
- return 0;
- }
-
- len = vorbis_finish_frame(f, len, left, right);
- for (i=0; i < f->channels; ++i)
- f->outputs[i] = f->channel_buffers[i] + left;
-
- f->channel_buffer_start = left;
- f->channel_buffer_end = left+len;
-
- if (channels) *channels = f->channels;
- if (output) *output = f->outputs;
- return len;
-}
-
-#ifndef STB_VORBIS_NO_STDIO
-
-stb_vorbis * stb_vorbis_open_file_section(FILE *file, int close_on_free, int *error, stb_vorbis_alloc *alloc, unsigned int length)
-{
- stb_vorbis *f, p;
- vorbis_init(&p, alloc);
- p.f = file;
- p.f_start = ftell(file);
- p.stream_len = length;
- p.close_on_free = close_on_free;
- if (start_decoder(&p)) {
- f = vorbis_alloc(&p);
- if (f) {
- *f = p;
- vorbis_pump_first_frame(f);
- return f;
- }
- }
- if (error) *error = p.error;
- vorbis_deinit(&p);
- return NULL;
-}
-
-stb_vorbis * stb_vorbis_open_file(FILE *file, int close_on_free, int *error, stb_vorbis_alloc *alloc)
-{
- unsigned int len, start;
- start = ftell(file);
- fseek(file, 0, SEEK_END);
- len = ftell(file) - start;
- fseek(file, start, SEEK_SET);
- return stb_vorbis_open_file_section(file, close_on_free, error, alloc, len);
-}
-
-stb_vorbis * stb_vorbis_open_filename(char *filename, int *error, stb_vorbis_alloc *alloc)
-{
- FILE *f = fopen(filename, "rb");
- if (f)
- return stb_vorbis_open_file(f, TRUE, error, alloc);
- if (error) *error = VORBIS_file_open_failure;
- return NULL;
-}
-#endif // STB_VORBIS_NO_STDIO
-
-stb_vorbis * stb_vorbis_open_memory(unsigned char *data, int len, int *error, stb_vorbis_alloc *alloc)
-{
- stb_vorbis *f, p;
- if (data == NULL) return NULL;
- vorbis_init(&p, alloc);
- p.stream = data;
- p.stream_end = data + len;
- p.stream_start = p.stream;
- p.stream_len = len;
- p.push_mode = FALSE;
- if (start_decoder(&p)) {
- f = vorbis_alloc(&p);
- if (f) {
- *f = p;
- vorbis_pump_first_frame(f);
- return f;
- }
- }
- if (error) *error = p.error;
- vorbis_deinit(&p);
- return NULL;
-}
-
-#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
-#define PLAYBACK_MONO 1
-#define PLAYBACK_LEFT 2
-#define PLAYBACK_RIGHT 4
-
-#define L (PLAYBACK_LEFT | PLAYBACK_MONO)
-#define C (PLAYBACK_LEFT | PLAYBACK_RIGHT | PLAYBACK_MONO)
-#define R (PLAYBACK_RIGHT | PLAYBACK_MONO)
-
-static int8 channel_position[7][6] =
-{
- { 0 },
- { C },
- { L, R },
- { L, C, R },
- { L, R, L, R },
- { L, C, R, L, R },
- { L, C, R, L, R, C },
-};
-
-
-#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
- // add (1<<23) to convert to int, then divide by 2^SHIFT, then add 0.5/2^SHIFT to round
- #define MAGIC(SHIFT) (1.5f * (1 << (23-SHIFT)) + 0.5f/(1 << SHIFT))
- #define ADDEND(SHIFT) (((150-SHIFT) << 23) + (1 << 22))
- #define FAST_SCALED_FLOAT_TO_INT(x,s) ((temp = (x) + MAGIC(s)), (*(int *)&temp) - ADDEND(s))
- #define check_endianness()
- typedef char stb_vorbis_float_size_test[sizeof(float)==4 && sizeof(int) == 4];
- #define FASTDEF(x) x
-#else
- #define FAST_SCALED_FLOAT_TO_INT(x,s) ((int) ((x) * (1 << (s))))
- #define check_endianness()
- #define FASTDEF(x)
-#endif
-
-static void copy_samples(short *dest, float *src, int len)
-{
- int i;
- FASTDEF(float temp);
- check_endianness();
- for (i=0; i < len; ++i) {
- int v = FAST_SCALED_FLOAT_TO_INT(src[i],15);
- if ((unsigned int) (v + 32768) > 65535)
- v = v < 0 ? -32768 : 32767;
- dest[i] = v;
- }
-}
-
-static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len)
-{
- #define BUFFER_SIZE 32
- float buffer[BUFFER_SIZE];
- int i,j,o,n = BUFFER_SIZE;
- FASTDEF(float temp);
- check_endianness();
- for (o = 0; o < len; o += BUFFER_SIZE) {
- memset(buffer, 0, sizeof(buffer));
- if (o + n > len) n = len - o;
- for (j=0; j < num_c; ++j) {
- if (channel_position[num_c][j] & mask) {
- for (i=0; i < n; ++i)
- buffer[i] += data[j][d_offset+o+i];
- }
- }
- for (i=0; i < n; ++i) {
- int v = FAST_SCALED_FLOAT_TO_INT(buffer[i],15);
- if ((unsigned int) (v + 32768) > 65535)
- v = v < 0 ? -32768 : 32767;
- output[o+i] = v;
- }
- }
-}
-
-static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} };
-static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len)
-{
- #define BUFFER_SIZE 32
- float buffer[BUFFER_SIZE];
- int i,j,o,n = BUFFER_SIZE >> 1;
- FASTDEF(float temp);
- // o is the offset in the source data
- check_endianness();
- for (o = 0; o < len; o += BUFFER_SIZE >> 1) {
- // o2 is the offset in the output data
- int o2 = o << 1;
- memset(buffer, 0, sizeof(buffer));
- if (o + n > len) n = len - o;
- for (j=0; j < num_c; ++j) {
- int m = channel_position[num_c][j] & (PLAYBACK_LEFT | PLAYBACK_RIGHT);
- if (m == (PLAYBACK_LEFT | PLAYBACK_RIGHT)) {
- for (i=0; i < n; ++i) {
- buffer[i*2+0] += data[j][d_offset+o+i];
- buffer[i*2+1] += data[j][d_offset+o+i];
- }
- } else if (m == PLAYBACK_LEFT) {
- for (i=0; i < n; ++i) {
- buffer[i*2+0] += data[j][d_offset+o+i];
- }
- } else if (m == PLAYBACK_RIGHT) {
- for (i=0; i < n; ++i) {
- buffer[i*2+1] += data[j][d_offset+o+i];
- }
- }
- }
- for (i=0; i < (n<<1); ++i) {
- int v = FAST_SCALED_FLOAT_TO_INT(buffer[i],15);
- if ((unsigned int) (v + 32768) > 65535)
- v = v < 0 ? -32768 : 32767;
- output[o2+i] = v;
- }
- }
-}
-
-static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples)
-{
- int i;
- if (buf_c != data_c && buf_c <= 2 && data_c <= 6) {
- static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} };
- for (i=0; i < buf_c; ++i)
- compute_samples(channel_selector[buf_c][i], buffer[i]+b_offset, data_c, data, d_offset, samples);
- } else {
- int limit = buf_c < data_c ? buf_c : data_c;
- for (i=0; i < limit; ++i)
- copy_samples(buffer[i]+b_offset, data[i], samples);
- for ( ; i < buf_c; ++i)
- memset(buffer[i]+b_offset, 0, sizeof(short) * samples);
- }
-}
-
-int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples)
-{
- float **output;
- int len = stb_vorbis_get_frame_float(f, NULL, &output);
- if (len > num_samples) len = num_samples;
- if (len)
- convert_samples_short(num_c, buffer, 0, f->channels, output, 0, len);
- return len;
-}
-
-static void convert_channels_short_interleaved(int buf_c, short *buffer, int data_c, float **data, int d_offset, int len)
-{
- int i;
- check_endianness();
- if (buf_c != data_c && buf_c <= 2 && data_c <= 6) {
- assert(buf_c == 2);
- for (i=0; i < buf_c; ++i)
- compute_stereo_samples(buffer, data_c, data, d_offset, len);
- } else {
- int limit = buf_c < data_c ? buf_c : data_c;
- int j;
- FASTDEF(float temp);
- for (j=0; j < len; ++j) {
- for (i=0; i < limit; ++i) {
- int v = FAST_SCALED_FLOAT_TO_INT(data[i][d_offset+j],15);
- if ((unsigned int) (v + 32768) > 65535)
- v = v < 0 ? -32768 : 32767;
- *buffer++ = v;
- }
- for ( ; i < buf_c; ++i)
- *buffer++ = 0;
- }
- }
-}
-
-int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts)
-{
- float **output;
- int len;
- if (num_c == 1) return stb_vorbis_get_frame_short(f,num_c,&buffer, num_shorts);
- len = stb_vorbis_get_frame_float(f, NULL, &output);
- if (len) {
- if (len*num_c > num_shorts) len = num_shorts / num_c;
- convert_channels_short_interleaved(num_c, buffer, f->channels, output, 0, len);
- }
- return len;
-}
-
-int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts)
-{
- float **outputs;
- int len = num_shorts / channels;
- int n=0;
- int z = f->channels;
- if (z > channels) z = channels;
- while (n < len) {
- int k = f->channel_buffer_end - f->channel_buffer_start;
- if (n+k >= len) k = len - n;
- if (k)
- convert_channels_short_interleaved(channels, buffer, f->channels, f->channel_buffers, f->channel_buffer_start, k);
- buffer += k*channels;
- n += k;
- f->channel_buffer_start += k;
- if (n == len) break;
- if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
- }
- return n;
-}
-
-int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int len)
-{
- float **outputs;
- int n=0;
- int z = f->channels;
- if (z > channels) z = channels;
- while (n < len) {
- int k = f->channel_buffer_end - f->channel_buffer_start;
- if (n+k >= len) k = len - n;
- if (k)
- convert_samples_short(channels, buffer, n, f->channels, f->channel_buffers, f->channel_buffer_start, k);
- n += k;
- f->channel_buffer_start += k;
- if (n == len) break;
- if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
- }
- return n;
-}
-
-#ifndef STB_VORBIS_NO_STDIO
-int stb_vorbis_decode_filename(char *filename, int *channels, short **output)
-{
- int data_len, offset, total, limit, error;
- short *data;
- stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL);
- if (v == NULL) return -1;
- limit = v->channels * 4096;
- *channels = v->channels;
- offset = data_len = 0;
- total = limit;
- data = (short *) malloc(total * sizeof(*data));
- if (data == NULL) {
- stb_vorbis_close(v);
- return -2;
- }
- for (;;) {
- int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset);
- if (n == 0) break;
- data_len += n;
- offset += n * v->channels;
- if (offset + limit > total) {
- short *data2;
- total *= 2;
- data2 = (short *) realloc(data, total * sizeof(*data));
- if (data2 == NULL) {
- free(data);
- stb_vorbis_close(v);
- return -2;
- }
- data = data2;
- }
- }
- *output = data;
- return data_len;
-}
-#endif // NO_STDIO
-
-int stb_vorbis_decode_memory(uint8 *mem, int len, int *channels, short **output)
-{
- int data_len, offset, total, limit, error;
- short *data;
- stb_vorbis *v = stb_vorbis_open_memory(mem, len, &error, NULL);
- if (v == NULL) return -1;
- limit = v->channels * 4096;
- *channels = v->channels;
- offset = data_len = 0;
- total = limit;
- data = (short *) malloc(total * sizeof(*data));
- if (data == NULL) {
- stb_vorbis_close(v);
- return -2;
- }
- for (;;) {
- int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset);
- if (n == 0) break;
- data_len += n;
- offset += n * v->channels;
- if (offset + limit > total) {
- short *data2;
- total *= 2;
- data2 = (short *) realloc(data, total * sizeof(*data));
- if (data2 == NULL) {
- free(data);
- stb_vorbis_close(v);
- return -2;
- }
- data = data2;
- }
- }
- *output = data;
- return data_len;
-}
-#endif
-
-int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats)
-{
- float **outputs;
- int len = num_floats / channels;
- int n=0;
- int z = f->channels;
- if (z > channels) z = channels;
- while (n < len) {
- int i,j;
- int k = f->channel_buffer_end - f->channel_buffer_start;
- if (n+k >= len) k = len - n;
- for (j=0; j < k; ++j) {
- for (i=0; i < z; ++i)
- *buffer++ = f->channel_buffers[i][f->channel_buffer_start+j];
- for ( ; i < channels; ++i)
- *buffer++ = 0;
- }
- n += k;
- f->channel_buffer_start += k;
- if (n == len) break;
- if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
- }
- return n;
-}
-
-int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples)
-{
- float **outputs;
- int n=0;
- int z = f->channels;
- if (z > channels) z = channels;
- while (n < num_samples) {
- int i;
- int k = f->channel_buffer_end - f->channel_buffer_start;
- if (n+k >= num_samples) k = num_samples - n;
- if (k) {
- for (i=0; i < z; ++i)
- memcpy(buffer[i]+n, f->channel_buffers+f->channel_buffer_start, sizeof(float)*k);
- for ( ; i < channels; ++i)
- memset(buffer[i]+n, 0, sizeof(float) * k);
- }
- n += k;
- f->channel_buffer_start += k;
- if (n == num_samples) break;
- if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
- }
- return n;
-}
-#endif // STB_VORBIS_NO_PULLDATA_API
-
-#endif // STB_VORBIS_HEADER_ONLY
diff --git a/src/SFML/Audio/stb_vorbis/stb_vorbis.h b/src/SFML/Audio/stb_vorbis/stb_vorbis.h deleted file mode 100755 index e2355e6..0000000 --- a/src/SFML/Audio/stb_vorbis/stb_vorbis.h +++ /dev/null @@ -1,357 +0,0 @@ -// Ogg Vorbis I audio decoder -- version 0.99994
-//
-// Written in April 2007 by Sean Barrett, sponsored by RAD Game Tools.
-//
-// Placed in the public domain April 2007 by the author: no copyright is
-// claimed, and you may use it for any purpose you like.
-//
-// No warranty for any purpose is expressed or implied by the author (nor
-// by RAD Game Tools). Report bugs and send enhancements to the author.
-//
-// Get the latest version and other information at:
-// http://nothings.org/stb_vorbis/
-
-
-// Todo:
-//
-// - seeking (note you can seek yourself using the pushdata API)
-//
-// Limitations:
-//
-// - floor 0 not supported (used in old ogg vorbis files)
-// - lossless sample-truncation at beginning ignored
-// - cannot concatenate multiple vorbis streams
-// - sample positions are 32-bit, limiting seekable 192Khz
-// files to around 6 hours (Ogg supports 64-bit)
-//
-// All of these limitations may be removed in future versions.
-
-
-//////////////////////////////////////////////////////////////////////////////
-//
-// HEADER BEGINS HERE
-//
-
-#ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H
-#define STB_VORBIS_INCLUDE_STB_VORBIS_H
-
-#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
-#define STB_VORBIS_NO_STDIO 1
-#endif
-
-#ifndef STB_VORBIS_NO_STDIO
-#include <stdio.h>
-#endif
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/////////// THREAD SAFETY
-
-// Individual stb_vorbis* handles are not thread-safe; you cannot decode from
-// them from multiple threads at the same time. However, you can have multiple
-// stb_vorbis* handles and decode from them independently in multiple thrads.
-
-
-/////////// MEMORY ALLOCATION
-
-// normally stb_vorbis uses malloc() to allocate memory at startup,
-// and alloca() to allocate temporary memory during a frame on the
-// stack. (Memory consumption will depend on the amount of setup
-// data in the file and how you set the compile flags for speed
-// vs. size. In my test files the maximal-size usage is ~150KB.)
-//
-// You can modify the wrapper functions in the source (setup_malloc,
-// setup_temp_malloc, temp_malloc) to change this behavior, or you
-// can use a simpler allocation model: you pass in a buffer from
-// which stb_vorbis will allocate _all_ its memory (including the
-// temp memory). "open" may fail with a VORBIS_outofmem if you
-// do not pass in enough data; there is no way to determine how
-// much you do need except to succeed (at which point you can
-// query get_info to find the exact amount required. yes I know
-// this is lame).
-//
-// If you pass in a non-NULL buffer of the type below, allocation
-// will occur from it as described above. Otherwise just pass NULL
-// to use malloc()/alloca()
-
-typedef struct
-{
- char *alloc_buffer;
- int alloc_buffer_length_in_bytes;
-} stb_vorbis_alloc;
-
-
-/////////// FUNCTIONS USEABLE WITH ALL INPUT MODES
-
-typedef struct stb_vorbis stb_vorbis;
-
-typedef struct
-{
- unsigned int sample_rate;
- int channels;
-
- unsigned int setup_memory_required;
- unsigned int setup_temp_memory_required;
- unsigned int temp_memory_required;
-
- int max_frame_size;
-} stb_vorbis_info;
-
-// get general information about the file
-extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f);
-
-// get the last error detected (clears it, too)
-extern int stb_vorbis_get_error(stb_vorbis *f);
-
-// close an ogg vorbis file and free all memory in use
-extern void stb_vorbis_close(stb_vorbis *f);
-
-// this function returns the offset (in samples) from the beginning of the
-// file that will be returned by the next decode, if it is known, or -1
-// otherwise. after a flush_pushdata() call, this may take a while before
-// it becomes valid again.
-// NOT WORKING YET after a seek with PULLDATA API
-extern int stb_vorbis_get_sample_offset(stb_vorbis *f);
-
-// returns the current seek point within the file, or offset from the beginning
-// of the memory buffer. In pushdata mode it returns 0.
-extern unsigned int stb_vorbis_get_file_offset(stb_vorbis *f);
-
-/////////// PUSHDATA API
-
-#ifndef STB_VORBIS_NO_PUSHDATA_API
-
-// this API allows you to get blocks of data from any source and hand
-// them to stb_vorbis. you have to buffer them; stb_vorbis will tell
-// you how much it used, and you have to give it the rest next time;
-// and stb_vorbis may not have enough data to work with and you will
-// need to give it the same data again PLUS more. Note that the Vorbis
-// specification does not bound the size of an individual frame.
-
-extern stb_vorbis *stb_vorbis_open_pushdata(
- unsigned char *datablock, int datablock_length_in_bytes,
- int *datablock_memory_consumed_in_bytes,
- int *error,
- stb_vorbis_alloc *alloc_buffer);
-// create a vorbis decoder by passing in the initial data block containing
-// the ogg&vorbis headers (you don't need to do parse them, just provide
-// the first N bytes of the file--you're told if it's not enough, see below)
-// on success, returns an stb_vorbis *, does not set error, returns the amount of
-// data parsed/consumed on this call in *datablock_memory_consumed_in_bytes;
-// on failure, returns NULL on error and sets *error, does not change *datablock_memory_consumed
-// if returns NULL and *error is VORBIS_need_more_data, then the input block was
-// incomplete and you need to pass in a larger block from the start of the file
-
-extern int stb_vorbis_decode_frame_pushdata(
- stb_vorbis *f, unsigned char *datablock, int datablock_length_in_bytes,
- int *channels, // place to write number of float * buffers
- float ***output, // place to write float ** array of float * buffers
- int *samples // place to write number of output samples
- );
-// decode a frame of audio sample data if possible from the passed-in data block
-//
-// return value: number of bytes we used from datablock
-// possible cases:
-// 0 bytes used, 0 samples output (need more data)
-// N bytes used, 0 samples output (resynching the stream, keep going)
-// N bytes used, M samples output (one frame of data)
-// note that after opening a file, you will ALWAYS get one N-bytes,0-sample
-// frame, because Vorbis always "discards" the first frame.
-//
-// Note that on resynch, stb_vorbis will rarely consume all of the buffer,
-// instead only datablock_length_in_bytes-3 or less. This is because it wants
-// to avoid missing parts of a page header if they cross a datablock boundary,
-// without writing state-machiney code to record a partial detection.
-//
-// The number of channels returned are stored in *channels (which can be
-// NULL--it is always the same as the number of channels reported by
-// get_info). *output will contain an array of float* buffers, one per
-// channel. In other words, (*output)[0][0] contains the first sample from
-// the first channel, and (*output)[1][0] contains the first sample from
-// the second channel.
-
-extern void stb_vorbis_flush_pushdata(stb_vorbis *f);
-// inform stb_vorbis that your next datablock will not be contiguous with
-// previous ones (e.g. you've seeked in the data); future attempts to decode
-// frames will cause stb_vorbis to resynchronize (as noted above), and
-// once it sees a valid Ogg page (typically 4-8KB, as large as 64KB), it
-// will begin decoding the _next_ frame.
-//
-// if you want to seek using pushdata, you need to seek in your file, then
-// call stb_vorbis_flush_pushdata(), then start calling decoding, then once
-// decoding is returning you data, call stb_vorbis_get_sample_offset, and
-// if you don't like the result, seek your file again and repeat.
-#endif
-
-
-////////// PULLING INPUT API
-
-#ifndef STB_VORBIS_NO_PULLDATA_API
-// This API assumes stb_vorbis is allowed to pull data from a source--
-// either a block of memory containing the _entire_ vorbis stream, or a
-// FILE * that you or it create, or possibly some other reading mechanism
-// if you go modify the source to replace the FILE * case with some kind
-// of callback to your code. (But if you don't support seeking, you may
-// just want to go ahead and use pushdata.)
-
-#if !defined(STB_VORBIS_NO_STDIO) && !defined(STB_VORBIS_NO_INTEGER_CONVERSION)
-extern int stb_vorbis_decode_filename(char *filename, int *channels, short **output);
-#endif
-extern int stb_vorbis_decode_memory(unsigned char *mem, int len, int *channels, short **output);
-// decode an entire file and output the data interleaved into a malloc()ed
-// buffer stored in *output. The return value is the number of samples
-// decoded, or -1 if the file could not be opened or was not an ogg vorbis file.
-// When you're done with it, just free() the pointer returned in *output.
-
-extern stb_vorbis * stb_vorbis_open_memory(unsigned char *data, int len,
- int *error, stb_vorbis_alloc *alloc_buffer);
-// create an ogg vorbis decoder from an ogg vorbis stream in memory (note
-// this must be the entire stream!). on failure, returns NULL and sets *error
-
-#ifndef STB_VORBIS_NO_STDIO
-extern stb_vorbis * stb_vorbis_open_filename(char *filename,
- int *error, stb_vorbis_alloc *alloc_buffer);
-// create an ogg vorbis decoder from a filename via fopen(). on failure,
-// returns NULL and sets *error (possibly to VORBIS_file_open_failure).
-
-extern stb_vorbis * stb_vorbis_open_file(FILE *f, int close_handle_on_close,
- int *error, stb_vorbis_alloc *alloc_buffer);
-// create an ogg vorbis decoder from an open FILE *, looking for a stream at
-// the _current_ seek point (ftell). on failure, returns NULL and sets *error.
-// note that stb_vorbis must "own" this stream; if you seek it in between
-// calls to stb_vorbis, it will become confused. Morever, if you attempt to
-// perform stb_vorbis_seek_*() operations on this file, it will assume it
-// owns the _entire_ rest of the file after the start point. Use the next
-// function, stb_vorbis_open_file_section(), to limit it.
-
-extern stb_vorbis * stb_vorbis_open_file_section(FILE *f, int close_handle_on_close,
- int *error, stb_vorbis_alloc *alloc_buffer, unsigned int len);
-// create an ogg vorbis decoder from an open FILE *, looking for a stream at
-// the _current_ seek point (ftell); the stream will be of length 'len' bytes.
-// on failure, returns NULL and sets *error. note that stb_vorbis must "own"
-// this stream; if you seek it in between calls to stb_vorbis, it will become
-// confused.
-#endif
-
-extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number);
-extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number);
-// NOT WORKING YET
-// these functions seek in the Vorbis file to (approximately) 'sample_number'.
-// after calling seek_frame(), the next call to get_frame_*() will include
-// the specified sample. after calling stb_vorbis_seek(), the next call to
-// stb_vorbis_get_samples_* will start with the specified sample. If you
-// do not need to seek to EXACTLY the target sample when using get_samples_*,
-// you can also use seek_frame().
-
-extern void stb_vorbis_seek_start(stb_vorbis *f);
-// this function is equivalent to stb_vorbis_seek(f,0), but it
-// actually works
-
-extern unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f);
-extern float stb_vorbis_stream_length_in_seconds(stb_vorbis *f);
-// these functions return the total length of the vorbis stream
-
-extern int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output);
-// decode the next frame and return the number of samples. the number of
-// channels returned are stored in *channels (which can be NULL--it is always
-// the same as the number of channels reported by get_info). *output will
-// contain an array of float* buffers, one per channel. These outputs will
-// be overwritten on the next call to stb_vorbis_get_frame_*.
-//
-// You generally should not intermix calls to stb_vorbis_get_frame_*()
-// and stb_vorbis_get_samples_*(), since the latter calls the former.
-
-#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
-extern int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts);
-extern int stb_vorbis_get_frame_short (stb_vorbis *f, int num_c, short **buffer, int num_samples);
-#endif
-// decode the next frame and return the number of samples per channel. the
-// data is coerced to the number of channels you request according to the
-// channel coercion rules (see below). You must pass in the size of your
-// buffer(s) so that stb_vorbis will not overwrite the end of the buffer.
-// The maximum buffer size needed can be gotten from get_info(); however,
-// the Vorbis I specification implies an absolute maximum of 4096 samples
-// per channel. Note that for interleaved data, you pass in the number of
-// shorts (the size of your array), but the return value is the number of
-// samples per channel, not the total number of samples.
-
-// Channel coercion rules:
-// Let M be the number of channels requested, and N the number of channels present,
-// and Cn be the nth channel; let stereo L be the sum of all L and center channels,
-// and stereo R be the sum of all R and center channels (channel assignment from the
-// vorbis spec).
-// M N output
-// 1 k sum(Ck) for all k
-// 2 * stereo L, stereo R
-// k l k > l, the first l channels, then 0s
-// k l k <= l, the first k channels
-// Note that this is not _good_ surround etc. mixing at all! It's just so
-// you get something useful.
-
-extern int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats);
-extern int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples);
-// gets num_samples samples, not necessarily on a frame boundary--this requires
-// buffering so you have to supply the buffers. DOES NOT APPLY THE COERCION RULES.
-// Returns the number of samples stored per channel; it may be less than requested
-// at the end of the file. If there are no more samples in the file, returns 0.
-
-#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
-extern int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts);
-extern int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int num_samples);
-#endif
-// gets num_samples samples, not necessarily on a frame boundary--this requires
-// buffering so you have to supply the buffers. Applies the coercion rules above
-// to produce 'channels' channels. Returns the number of samples stored per channel;
-// it may be less than requested at the end of the file. If there are no more
-// samples in the file, returns 0.
-
-#endif
-
-//////// ERROR CODES
-
-enum STBVorbisError
-{
- VORBIS__no_error,
-
- VORBIS_need_more_data=1, // not a real error
-
- VORBIS_invalid_api_mixing, // can't mix API modes
- VORBIS_outofmem, // not enough memory
- VORBIS_feature_not_supported, // uses floor 0
- VORBIS_too_many_channels, // STB_VORBIS_MAX_CHANNELS is too small
- VORBIS_file_open_failure, // fopen() failed
- VORBIS_seek_without_length, // can't seek in unknown-length file
-
- VORBIS_unexpected_eof=10, // file is truncated?
- VORBIS_seek_invalid, // seek past EOF
-
- // decoding errors (corrupt/invalid stream) -- you probably
- // don't care about the exact details of these
-
- // vorbis errors:
- VORBIS_invalid_setup=20,
- VORBIS_invalid_stream,
-
- // ogg errors:
- VORBIS_missing_capture_pattern=30,
- VORBIS_invalid_stream_structure_version,
- VORBIS_continued_packet_flag_invalid,
- VORBIS_incorrect_stream_serial_number,
- VORBIS_invalid_first_page,
- VORBIS_bad_packet_type,
- VORBIS_cant_find_last_page,
- VORBIS_seek_failed
-};
-
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // STB_VORBIS_INCLUDE_STB_VORBIS_H
-//
-// HEADER ENDS HERE
-//
-//////////////////////////////////////////////////////////////////////////////
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