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authorAlfred E. Heggestad <alfred.heggestad@gmail.com>2017-12-25 16:56:44 +0100
committerAlfred E. Heggestad <alfred.heggestad@gmail.com>2017-12-25 16:56:44 +0100
commit99e9bf9ca7d65994312ad615132650176d430d6f (patch)
tree5ccccd07554c6bb9b4716bdf5c7058dd3e8fb1b1 /src/audio.c
parent917bc4ee372da83381efb381f8f535ca64839173 (diff)
audio: update doxygen comments
Diffstat (limited to 'src/audio.c')
-rw-r--r--src/audio.c18
1 files changed, 9 insertions, 9 deletions
diff --git a/src/audio.c b/src/audio.c
index 17eb5f6..f0d6720 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -81,7 +81,7 @@ struct autx {
struct auenc_state *enc; /**< Audio encoder state (optional) */
struct aubuf *aubuf; /**< Packetize outgoing stream */
size_t aubuf_maxsz; /**< Maximum aubuf size in [bytes] */
- volatile bool aubuf_started;
+ volatile bool aubuf_started; /**< Aubuf was started flag */
struct auresamp resamp; /**< Optional resampler for DSP */
struct list filtl; /**< Audio filters in encoding order */
struct mbuf *mb; /**< Buffer for outgoing RTP packets */
@@ -90,14 +90,14 @@ struct autx {
int16_t *sampv_rs; /**< Sample buffer for resampler */
uint32_t ptime; /**< Packet time for sending */
uint64_t ts_ext; /**< Ext. Timestamp for outgoing RTP */
- uint32_t ts_base;
+ uint32_t ts_base; /**< First timestamp sent */
uint32_t ts_tel; /**< Timestamp for Telephony Events */
size_t psize; /**< Packet size for sending */
bool marker; /**< Marker bit for outgoing RTP */
bool muted; /**< Audio source is muted */
int cur_key; /**< Currently transmitted event */
- enum aufmt src_fmt;
- bool need_conv;
+ enum aufmt src_fmt; /**< Sample format for audio source */
+ bool need_conv; /**< Sample format conversion needed */
struct {
uint64_t aubuf_overrun;
@@ -137,7 +137,7 @@ struct aurx {
struct audec_state *dec; /**< Audio decoder state (optional) */
struct aubuf *aubuf; /**< Incoming audio buffer */
size_t aubuf_maxsz; /**< Maximum aubuf size in [bytes] */
- volatile bool aubuf_started;
+ volatile bool aubuf_started; /**< Aubuf was started flag */
struct auresamp resamp; /**< Optional resampler for DSP */
struct list filtl; /**< Audio filters in decoding order */
char device[64]; /**< Audio player device name */
@@ -147,9 +147,9 @@ struct aurx {
int pt; /**< Payload type for incoming RTP */
double level_last;
bool level_set;
- enum aufmt play_fmt;
- bool need_conv;
- struct timestamp_recv ts_recv;
+ enum aufmt play_fmt; /**< Sample format for audio playback*/
+ bool need_conv; /**< Sample format conversion needed */
+ struct timestamp_recv ts_recv;/**< Receive timestamp state */
uint64_t n_discard;
struct {
@@ -309,7 +309,7 @@ static inline double calc_ptime(size_t nsamp, uint32_t srate, uint8_t channels)
/**
- * Get the DSP samplerate for an audio-codec (exception for G.722 and MPA)
+ * Get the DSP samplerate for an audio-codec
*/
static inline uint32_t get_srate(const struct aucodec *ac)
{