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| author | Johns <johns98@gmx.net> | 2012-02-02 16:01:08 +0100 |
|---|---|---|
| committer | Johns <johns98@gmx.net> | 2012-02-02 16:01:08 +0100 |
| commit | 3585f1df193c771083c01a124249472dfc4111bd (patch) | |
| tree | 1b6abdc81511126e1be715968943a51b3d222298 | |
| parent | e258c355371ae45f3ee6cf00a1badfdbe1591551 (diff) | |
| download | vdr-plugin-softhddevice-3585f1df193c771083c01a124249472dfc4111bd.tar.gz vdr-plugin-softhddevice-3585f1df193c771083c01a124249472dfc4111bd.tar.bz2 | |
Increase audio buffer, if bigger audio delay used.
| -rw-r--r-- | audio.c | 25 |
1 files changed, 17 insertions, 8 deletions
@@ -137,7 +137,7 @@ static unsigned AudioSampleRate; ///< audio sample rate in hz static unsigned AudioChannels; ///< number of audio channels static const int AudioBytesProSample = 2; ///< number of bytes per sample 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 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 #endif +extern int VideoAudioDelay; /// import audio/video delay + #ifdef USE_AUDIORING //---------------------------------------------------------------------------- @@ -900,6 +902,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3) snd_pcm_uframes_t period_size; int err; int ret; + int delay; snd_pcm_t *handle; 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); // buffer time/delay in ms + delay = AudioBufferTime; + if (VideoAudioDelay > -100) { + delay += 100 + VideoAudioDelay / 90; + } if (AlsaStartThreshold < - (*freq * *channels * AudioBytesProSample * AudioBufferTime) / 1000U) { + (*freq * *channels * AudioBytesProSample * delay) / 1000U) { AlsaStartThreshold = - (*freq * *channels * AudioBytesProSample * AudioBufferTime) / - 1000U; + (*freq * *channels * AudioBytesProSample * delay) / 1000U; } // no bigger, than the buffer if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) { @@ -1622,6 +1628,7 @@ static int OssSetup(int *freq, int *channels, int use_ac3) { int ret; int tmp; + int delay; if (OssPcmFildes == -1) { // OSS not ready return -1; @@ -1708,12 +1715,14 @@ static int OssSetup(int *freq, int *channels, int use_ac3) // start when enough bytes for initial write OssStartThreshold = bi.bytes + tmp; // buffer time/delay in ms + delay = AudioBufferTime; + if (VideoAudioDelay > -100) { + delay += 100 + VideoAudioDelay / 90; + } if (OssStartThreshold < - (*freq * *channels * AudioBytesProSample * AudioBufferTime) / - 1000U) { + (*freq * *channels * AudioBytesProSample * delay) / 1000U) { OssStartThreshold = - (*freq * *channels * AudioBytesProSample * AudioBufferTime) / - 1000U; + (*freq * *channels * AudioBytesProSample * delay) / 1000U; } // no bigger, than the buffer if (OssStartThreshold > RingBufferFreeBytes(OssRingBuffer)) { |