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Code Issues Releases Wiki Activity

Increase audio buffer, if bigger audio delay used.

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This commit is contained in:
Johns 2012-02-02 16:01:08 +01:00
parent e258c35537
commit 3585f1df19
1 changed files with 17 additions and 8 deletions
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25
audio.c
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@ -137,7 +137,7 @@ static unsigned AudioSampleRate; ///< audio sample rate in hz
static unsigned AudioChannels; ///< number of audio channels static unsigned AudioChannels; ///< number of audio channels
static const int AudioBytesProSample = 2; ///< number of bytes per sample static const int AudioBytesProSample = 2; ///< number of bytes per sample
static int64_t AudioPTS; ///< audio pts clock static int64_t AudioPTS; ///< audio pts clock
static const int AudioBufferTime = 450; ///< audio buffer time in ms static const int AudioBufferTime = 350; ///< audio buffer time in ms
#ifdef USE_AUDIO_THREAD #ifdef USE_AUDIO_THREAD
static pthread_t AudioThread; ///< audio play thread static pthread_t AudioThread; ///< audio play thread
@ -147,6 +147,8 @@ static pthread_cond_t AudioStartCond; ///< condition variable
static const int AudioThread; ///< dummy audio thread static const int AudioThread; ///< dummy audio thread
#endif #endif
extern int VideoAudioDelay; /// import audio/video delay
#ifdef USE_AUDIORING #ifdef USE_AUDIORING
//---------------------------------------------------------------------------- //----------------------------------------------------------------------------
@ -900,6 +902,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
snd_pcm_uframes_t period_size; snd_pcm_uframes_t period_size;
int err; int err;
int ret; int ret;
int delay;
snd_pcm_t *handle; snd_pcm_t *handle;
if (!AlsaPCMHandle) { // alsa not running yet if (!AlsaPCMHandle) { // alsa not running yet
@ -1085,11 +1088,14 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size); AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size);
// buffer time/delay in ms // buffer time/delay in ms
delay = AudioBufferTime;
if (VideoAudioDelay > -100) {
delay += 100 + VideoAudioDelay / 90;
}
if (AlsaStartThreshold < if (AlsaStartThreshold <
(*freq * *channels * AudioBytesProSample * AudioBufferTime) / 1000U) { (*freq * *channels * AudioBytesProSample * delay) / 1000U) {
AlsaStartThreshold = AlsaStartThreshold =
(*freq * *channels * AudioBytesProSample * AudioBufferTime) / (*freq * *channels * AudioBytesProSample * delay) / 1000U;
1000U;
} }
// no bigger, than the buffer // no bigger, than the buffer
if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) { if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) {
@ -1622,6 +1628,7 @@ static int OssSetup(int *freq, int *channels, int use_ac3)
{ {
int ret; int ret;
int tmp; int tmp;
int delay;
if (OssPcmFildes == -1) { // OSS not ready if (OssPcmFildes == -1) { // OSS not ready
return -1; return -1;
@ -1708,12 +1715,14 @@ static int OssSetup(int *freq, int *channels, int use_ac3)
// start when enough bytes for initial write // start when enough bytes for initial write
OssStartThreshold = bi.bytes + tmp; OssStartThreshold = bi.bytes + tmp;
// buffer time/delay in ms // buffer time/delay in ms
delay = AudioBufferTime;
if (VideoAudioDelay > -100) {
delay += 100 + VideoAudioDelay / 90;
}
if (OssStartThreshold < if (OssStartThreshold <
(*freq * *channels * AudioBytesProSample * AudioBufferTime) / (*freq * *channels * AudioBytesProSample * delay) / 1000U) {
1000U) {
OssStartThreshold = OssStartThreshold =
(*freq * *channels * AudioBytesProSample * AudioBufferTime) / (*freq * *channels * AudioBytesProSample * delay) / 1000U;
1000U;
} }
// no bigger, than the buffer // no bigger, than the buffer
if (OssStartThreshold > RingBufferFreeBytes(OssRingBuffer)) { if (OssStartThreshold > RingBufferFreeBytes(OssRingBuffer)) {