frodovdr/vdr-plugin-softhddevice
1
0
Fork 0
mirror of https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git synced 2023-10-10 19:16:51 +02:00
Code Issues Releases Wiki Activity

Buffertime compile time configurable in ms.

Browse Source
This commit is contained in:
Johns 2012-01-21 21:46:47 +01:00
parent fa27a1c73a
commit bd7e6143c7
2 changed files with 26 additions and 6 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
ChangeLog
View File

@ -1,4 +1,8 @@
User johns
Data:
Buffertime compile time configurable in ms.
Date: Sat Jan 21 15:49:16 CET 2012
Release Version 0.4.0

28
audio.c
View File

@ -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)) {