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| author | Johns <johns98@gmx.net> | 2012-01-21 21:46:47 +0100 |
|---|---|---|
| committer | Johns <johns98@gmx.net> | 2012-01-21 21:46:47 +0100 |
| commit | bd7e6143c734a40eadd5937a7f4a8d03070e8b11 (patch) | |
| tree | 57b6319f0afc7ef8ea9906a5ab98803be76f9a91 /audio.c | |
| parent | fa27a1c73a1070b7af127695362ad0e5c424b079 (diff) | |
| download | vdr-plugin-softhddevice-bd7e6143c734a40eadd5937a7f4a8d03070e8b11.tar.gz vdr-plugin-softhddevice-bd7e6143c734a40eadd5937a7f4a8d03070e8b11.tar.bz2 | |
Buffertime compile time configurable in ms.
Diffstat (limited to 'audio.c')
| -rw-r--r-- | audio.c | 28 |
1 files changed, 22 insertions, 6 deletions
@@ -136,6 +136,7 @@ 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 = 350; ///< audio buffer time in ms #ifdef USE_AUDIO_THREAD static pthread_t AudioThread; ///< audio play thread @@ -574,6 +575,8 @@ static void AlsaEnqueue(const void *samples, int count) // direct play produces underuns on some hardware +#ifndef USE_AUDIO_THREAD + /** ** Place samples in audio output queue. ** @@ -587,6 +590,8 @@ static void AlsaEnqueue(const void *samples, int count) } } +#endif + #ifdef USE_AUDIO_THREAD //---------------------------------------------------------------------------- @@ -1070,9 +1075,12 @@ static int AlsaSetup(int *freq, int *channels) snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle))); AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size); - // min 333ms - if (AlsaStartThreshold < (*freq * *channels * AudioBytesProSample) / 3U) { - AlsaStartThreshold = (*freq * *channels * AudioBytesProSample) / 3U; + // buffer time/delay in ms + if (AlsaStartThreshold < + (*freq * *channels * AudioBytesProSample * AudioBufferTime) / 1000U) { + AlsaStartThreshold = + (*freq * *channels * AudioBytesProSample * AudioBufferTime) / + 1000U; } // no bigger, than the buffer if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) { @@ -1286,6 +1294,8 @@ static void OssFlushBuffers(void) // OSS pcm polled //---------------------------------------------------------------------------- +#ifndef USE_AUDIO_THREAD + /** ** Place samples in audio output queue. ** @@ -1312,6 +1322,8 @@ static void OssEnqueue(const void *samples, int count) } } +#endif + /** ** Play all samples possible, without blocking. */ @@ -1656,9 +1668,13 @@ static int OssSetup(int *freq, int *channels) } // start when enough bytes for initial write OssStartThreshold = bi.bytes + tmp; - // min 333ms - if (OssStartThreshold < (*freq * *channels * AudioBytesProSample) / 3U) { - OssStartThreshold = (*freq * *channels * AudioBytesProSample) / 3U; + // buffer time/delay in ms + if (OssStartThreshold < + (*freq * *channels * AudioBytesProSample * AudioBufferTime) / + 1000U) { + OssStartThreshold = + (*freq * *channels * AudioBytesProSample * AudioBufferTime) / + 1000U; } // no bigger, than the buffer if (OssStartThreshold > RingBufferFreeBytes(OssRingBuffer)) { |