diff options
| -rw-r--r-- | audio.h | 1 | ||||
| -rw-r--r-- | audio_alsa.c | 37 | ||||
| -rw-r--r-- | audio_ao.c | 1 | ||||
| -rw-r--r-- | audio_dummy.c | 1 | ||||
| -rw-r--r-- | audio_jack.c | 1 | ||||
| -rw-r--r-- | audio_pa.c | 1 | ||||
| -rw-r--r-- | audio_pipe.c | 1 | ||||
| -rw-r--r-- | audio_sndio.c | 1 | ||||
| -rw-r--r-- | audio_soundio.c | 1 | ||||
| -rw-r--r-- | audio_stdout.c | 1 | ||||
| -rw-r--r-- | common.h | 3 |
11 files changed, 39 insertions, 10 deletions
@@ -22,6 +22,7 @@ typedef struct { void (*start)(int sample_rate, int sample_format); // block of samples + int (*preflight)(void *buf, int samples); // say you are about to send these samples (before interpolation) int (*play)(void *buf, int samples); void (*stop)(void); diff --git a/audio_alsa.c b/audio_alsa.c index b1c7ae7..b7a5de8 100644 --- a/audio_alsa.c +++ b/audio_alsa.c @@ -42,6 +42,7 @@ static int init(int argc, char **argv); static void deinit(void); static void start(int i_sample_rate, int i_sample_format); static int play(void *buf, int samples); +int preflight(void *buf, int samples); static void stop(void); static void flush(void); int delay(long *the_delay); @@ -632,6 +633,8 @@ static int init(int argc, char **argv) { config.audio_backend_buffer_interpolation_threshold_in_seconds = 0.100; // below this, basic interpolation will be used to save time. config.alsa_maximum_stall_time = 0.200; // 200 milliseconds -- if it takes longer, it's a problem + config.audio_backend_silence_threshold = 0.050; //start sending silent frames if the delay goes below this time + config.audio_backend_silence_scan_interval = 0.005; //check silence threshold this often stall_monitor_error_threshold = (uint64_t)1000000 * config.alsa_maximum_stall_time; // stall time max to microseconds; @@ -1081,7 +1084,7 @@ int delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay) { return ret; } -int untimed_delay(long *the_delay) { +int delay(long *the_delay) { // returns 0 if the device is in a valid state -- SND_PCM_STATE_RUNNING or SND_PCM_STATE_PREPARED // or SND_PCM_STATE_DRAINING // and returns the actual delay if running or 0 if prepared in *the_delay @@ -1126,7 +1129,7 @@ int get_rate_information(uint64_t *elapsed_time, uint64_t *frames_played) { return response; } -int untimed_play(void *buf, int samples) { +int play(void *buf, int samples) { // debug(3,"audio_alsa play called."); int oldState; @@ -1258,6 +1261,16 @@ static void flush(void) { pthread_setcancelstate(oldState, NULL); } +int preflight(__attribute__((unused)) void *buf, __attribute__((unused)) int samples) { + uint64_t time_now = + get_absolute_time_in_fp(); // this is to regulate access by the silence filler thread + uint64_t standoff_time = 100; // milliseconds + standoff_time = (standoff_time << 32)/1000; + most_recent_write_time = time_now + standoff_time; + return 0; +} + +/* static int play(void *buf, int samples) { uint64_t time_now = get_absolute_time_in_fp(); // this is to regulate access by the silence filler thread @@ -1273,6 +1286,7 @@ int delay(long *the_delay) { most_recent_write_time = time_now; return untimed_delay(the_delay); } +*/ static void stop(void) { // debug(2,"audio_alsa stop called."); @@ -1451,12 +1465,14 @@ void *alsa_buffer_monitor_thread_code(void *arg) { debug(1, "alsa: dither will %sbe added to inter-session silence.", use_dither ? "" : "not "); - int sleep_time_ms = 30; + int sleep_time_ms = (int)(config.audio_backend_silence_scan_interval * 1000); + long buffer_size_threshold = (long)(config.audio_backend_silence_threshold * desired_sample_rate); + uint64_t sleep_time_in_fp = sleep_time_ms; sleep_time_in_fp = sleep_time_in_fp << 32; sleep_time_in_fp = sleep_time_in_fp / 1000; // debug(1,"alsa: sleep_time: %d ms or 0x%" PRIx64 " in fp form.",sleep_time_ms,sleep_time_in_fp); - int frames_of_silence = (desired_sample_rate * sleep_time_ms * 2) / 1000; + int frames_of_silence = (desired_sample_rate * sleep_time_ms * 4) / 1000; size_t size_of_silence_buffer = frames_of_silence * frame_size; // debug(1,"Silence buffer length: %u bytes.",size_of_silence_buffer); void *silence = malloc(size_of_silence_buffer); @@ -1470,10 +1486,11 @@ void *alsa_buffer_monitor_thread_code(void *arg) { if (config.keep_dac_busy != 0) { uint64_t present_time = get_absolute_time_in_fp(); - if ((most_recent_write_time == 0) || - ((present_time > most_recent_write_time) && - ((present_time - most_recent_write_time) > (sleep_time_in_fp)))) { - reply = untimed_delay(&buffer_size); + if ((most_recent_write_time == 0) || (present_time > most_recent_write_time)) { + +// ((present_time > most_recent_write_time) && +// ((present_time - most_recent_write_time) > (sleep_time_in_fp)))) { + reply = delay(&buffer_size); if (reply != 0) { buffer_size = 0; char errorstring[1024]; @@ -1481,7 +1498,7 @@ void *alsa_buffer_monitor_thread_code(void *arg) { debug(1, "alsa: alsa_buffer_monitor_thread_code delay error %d: \"%s\".", reply, (char *)errorstring); } - if (buffer_size < frames_of_silence) { + if (buffer_size < buffer_size_threshold) { if ((hardware_mixer == 0) && (config.ignore_volume_control == 0) && (config.airplay_volume != 0.0)) use_dither = 1; @@ -1492,7 +1509,7 @@ void *alsa_buffer_monitor_thread_code(void *arg) { dither_random_number_store); // debug(1,"Play %d frames of silence with most_recent_write_time of %" PRIx64 ".", // frames_of_silence,most_recent_write_time); - untimed_play(silence, frames_of_silence); + play(silence, frames_of_silence); } } else { // debug(2,"Skipping sending silence"); @@ -134,6 +134,7 @@ audio_output audio_ao = {.name = "ao", .is_running = NULL, .flush = NULL, .delay = NULL, + .preflight = NULL, .play = &play, .volume = NULL, .parameters = NULL, diff --git a/audio_dummy.c b/audio_dummy.c index d8dea87..69518cd 100644 --- a/audio_dummy.c +++ b/audio_dummy.c @@ -65,6 +65,7 @@ audio_output audio_dummy = {.name = "dummy", .is_running = NULL, .flush = NULL, .delay = NULL, + .preflight = NULL, .play = &play, .volume = NULL, .parameters = NULL, diff --git a/audio_jack.c b/audio_jack.c index 11ae607..90f08ac 100644 --- a/audio_jack.c +++ b/audio_jack.c @@ -67,6 +67,7 @@ audio_output audio_jack = {.name = "jack", .is_running = &jack_is_running, .flush = &jack_flush, .delay = &jack_delay, + .preflight = NULL, .play = &play, .volume = NULL, .parameters = NULL, @@ -312,6 +312,7 @@ audio_output audio_pa = {.name = "pa", .flush = &flush, .delay = &pa_delay, .play = &play, + .preflight = NULL, .volume = NULL, .parameters = NULL, .mute = NULL}; diff --git a/audio_pipe.c b/audio_pipe.c index 89a09a0..0d4a42f 100644 --- a/audio_pipe.c +++ b/audio_pipe.c @@ -139,6 +139,7 @@ audio_output audio_pipe = {.name = "pipe", .flush = NULL, .delay = NULL, .play = &play, + .preflight = NULL, .volume = NULL, .parameters = NULL, .mute = NULL}; diff --git a/audio_sndio.c b/audio_sndio.c index f22ed5a..e106b15 100644 --- a/audio_sndio.c +++ b/audio_sndio.c @@ -49,6 +49,7 @@ audio_output audio_sndio = {.name = "sndio", .flush = &flush, .delay = &delay, .play = &play, + .preflight = NULL, .volume = NULL, .parameters = NULL, .mute = NULL}; diff --git a/audio_soundio.c b/audio_soundio.c index 7d3b48a..0449895 100644 --- a/audio_soundio.c +++ b/audio_soundio.c @@ -221,6 +221,7 @@ audio_output audio_soundio = {.name = "soundio", .flush = &flush, .delay = NULL, .play = &play, + .preflight = NULL, .volume = NULL, .parameters = ¶meters, .mute = NULL}; diff --git a/audio_stdout.c b/audio_stdout.c index 0438b1a..37d6f78 100644 --- a/audio_stdout.c +++ b/audio_stdout.c @@ -86,6 +86,7 @@ audio_output audio_stdout = {.name = "stdout", .flush = NULL, .delay = NULL, .play = &play, + .preflight = NULL, .volume = NULL, .parameters = NULL, .mute = NULL}; @@ -171,6 +171,9 @@ typedef struct { double audio_backend_buffer_interpolation_threshold_in_seconds; // below this, soxr interpolation // will not occur -- it'll be // basic interpolation instead. + double audio_backend_silence_threshold; // below this, silence will be added to the output buffer + double audio_backend_silence_scan_interval; // check the threshold this often + double audio_backend_latency_offset; // this will be the offset in seconds to compensate for any // fixed latency there might be in the audio path double audio_backend_silent_lead_in_time; // the length of the silence that should precede a play. |