mirror of
https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
synced 2023-10-10 19:16:51 +02:00
New audio ring buffer code, now OSS part.
This commit is contained in:
Johns
parent
53f22a2ed2
commit
3e39ffd5e0
7
Todo
7
Todo
@ -37,6 +37,7 @@ video:
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radio no need to wait on video buffers
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starting with radio and own X11 server, shows no video
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some low-bandwidth tv channels have hiccups.
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check start with 24Hz display rate
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vdpau:
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software decoder path not working
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@ -73,10 +74,8 @@ x11:
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support embedded mode
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audio:
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Combine alsa+oss ringbuffer code.
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Make alsa thread/polled and oss thread/polled output module runtime
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selectable.
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software volume support (could be done with asound.conf)
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Mute should do a real mute and not only set volume to zero.
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Starting suspended and muted, didn't register the mute.
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Relaxed audio sync checks at end of packet and already in sync
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@ -84,12 +83,10 @@ audio:
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only wait for video start, if video is running.
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Not primary device, don't use and block audio/video.
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multiple open of audio device, reduce them.
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Not all channel conversions are written (f.e. 2->3 ... 5->6 ...)
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audio/alsa:
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better downmix of >2 channels on 2 channel hardware
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remix support of unsupported sample rates
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libav supports only resample of mono to 2 channels
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ffmpeg didn't support resample of 5 to 2 channels
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audio/oss:
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alsa oss emulation mixer "pcm" not working
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366
audio.c
366
audio.c
@ -35,13 +35,12 @@
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///
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///
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/// @todo FIXME: there can be problems with little/big endian.
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/// @todo FIXME: can combine OSS and alsa ring buffer
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///
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//#define USE_ALSA ///< enable alsa support
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//#define USE_OSS ///< enable OSS support
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#define USE_AUDIO_THREAD ///< use thread for audio playback
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#define USE_AUDIORING ///< new audio ring code (incomplete)
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#define USE_AUDIORING ///< new audio ring code (testing)
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#include <stdio.h>
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#include <stdint.h>
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@ -107,12 +106,16 @@ typedef struct _audio_module_
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const char *Name; ///< audio output module name
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int (*const Thread) (void); ///< module thread handler
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#ifndef USE_AUDIORING
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void (*const Enqueue) (const void *, int); ///< enqueue samples for output
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void (*const VideoReady) (void); ///< video ready, start audio
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#endif
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void (*const FlushBuffers) (void); ///< flush sample buffers
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#ifndef USE_AUDIORING
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void (*const Poller) (void); ///< output poller
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int (*const FreeBytes) (void); ///< number of bytes free in buffer
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int (*const UsedBytes) (void); ///< number of bytes used in buffer
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#endif
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int64_t(*const GetDelay) (void); ///< get current audio delay
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void (*const SetVolume) (int); ///< set output volume
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int (*const Setup) (int *, int *, int); ///< setup channels, samplerate
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@ -142,13 +145,14 @@ static char AudioDoingInit; ///> flag in init, reduce error
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static volatile char AudioRunning; ///< thread running / stopped
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static volatile char AudioPaused; ///< audio paused
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static volatile char AudioVideoIsReady; ///< video ready start early
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#ifndef USE_AUDIORING
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static unsigned AudioSampleRate; ///< audio sample rate in Hz
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static unsigned AudioChannels; ///< number of audio channels
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static const int AudioBytesProSample = 2; ///< number of bytes per sample
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static int64_t AudioPTS; ///< audio pts clock
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#ifndef USE_AUDIO_THREAD
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#endif
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static const int AudioBytesProSample = 2; ///< number of bytes per sample
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static int AudioBufferTime = 336; ///< audio buffer time in ms
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#ifdef USE_AUDIO_THREAD
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@ -170,7 +174,7 @@ static int AudioMaxNormalize; ///< max. normalize factor
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static int AudioCompressionFactor; ///< current compression factor
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static int AudioMaxCompression; ///< max. compression factor
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static int AudioStereoDescent; ///< volume descent for stereo
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static int AudioVolume; ///< volume (0 .. 1000)
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static int AudioVolume; ///< current volume (0 .. 1000)
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extern int VideoAudioDelay; ///< import audio/video delay
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@ -752,14 +756,15 @@ static int AlsaRatio; ///< internal -> mixer ratio * 1000
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static int AlsaPlayRingbuffer(void)
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{
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int first;
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first = 1;
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for (;;) { // loop for ring buffer wrap
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int avail;
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int n;
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int err;
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int frames;
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const void *p;
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first = 1;
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for (;;) { // loop for ring buffer wrap
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// how many bytes can be written?
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n = snd_pcm_avail_update(AlsaPCMHandle);
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if (n < 0) {
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@ -821,6 +826,7 @@ static int AlsaPlayRingbuffer(void)
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if (AudioSoftVolume && !AudioRing[AudioRingRead].UseAc3) {
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// FIXME: quick&dirty cast
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AudioSoftAmplifier((int16_t *) p, avail);
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// FIXME: if not all are written, we double amplify them
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}
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frames = snd_pcm_bytes_to_frames(AlsaPCMHandle, avail);
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@ -2209,14 +2215,104 @@ static const AudioModule AlsaModule = {
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static int OssPcmFildes = -1; ///< pcm file descriptor
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static int OssMixerFildes = -1; ///< mixer file descriptor
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static int OssMixerChannel; ///< mixer channel index
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static RingBuffer *OssRingBuffer; ///< audio ring buffer
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static int OssFragmentTime; ///< fragment time in ms
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#ifndef USE_AUDIORING
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static RingBuffer *OssRingBuffer; ///< audio ring buffer
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static unsigned OssStartThreshold; ///< start play, if filled
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#endif
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#ifdef USE_AUDIO_THREAD
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static volatile char OssFlushBuffer; ///< flag empty buffer
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#endif
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#ifdef USE_AUDIORING
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//----------------------------------------------------------------------------
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// OSS pcm
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//----------------------------------------------------------------------------
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/**
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** Play samples from ringbuffer.
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**
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** @retval 0 ok
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** @retval 1 ring buffer empty
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** @retval -1 underrun error
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*/
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static int OssPlayRingbuffer(void)
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{
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int first;
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first = 1;
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for (;;) {
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audio_buf_info bi;
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const void *p;
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int n;
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if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) {
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Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"),
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strerror(errno));
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return -1;
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}
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Debug(4, "audio/oss: %d bytes free\n", bi.bytes);
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n = RingBufferGetReadPointer(AudioRing[AudioRingRead].RingBuffer, &p);
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if (!n) { // ring buffer empty
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if (first) { // only error on first loop
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return 1;
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}
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return 0;
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}
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if (n < bi.bytes) { // not enough bytes in ring buffer
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bi.bytes = n;
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}
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if (bi.bytes <= 0) { // full or buffer empty
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break; // bi.bytes could become negative!
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}
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if (AudioSoftVolume && !AudioRing[AudioRingRead].UseAc3) {
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// FIXME: quick&dirty cast
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AudioSoftAmplifier((int16_t *) p, bi.bytes);
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// FIXME: if not all are written, we double amplify them
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}
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for (;;) {
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n = write(OssPcmFildes, p, bi.bytes);
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if (n != bi.bytes) {
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if (n < 0) {
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if (n == EAGAIN) {
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continue;
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}
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Error(_("audio/oss: write error: %s\n"), strerror(errno));
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return 1;
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}
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Warning(_("audio/oss: error not all bytes written\n"));
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}
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break;
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}
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// advance how many could written
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RingBufferReadAdvance(AudioRing[AudioRingRead].RingBuffer, n);
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first = 0;
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}
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return 0;
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}
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/**
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** Flush OSS buffers.
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*/
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static void OssFlushBuffers(void)
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{
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if (OssPcmFildes != -1) {
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// flush kernel buffers
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if (ioctl(OssPcmFildes, SNDCTL_DSP_HALT_OUTPUT, NULL) < 0) {
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Error(_("audio/oss: ioctl(SNDCTL_DSP_HALT_OUTPUT): %s\n"),
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strerror(errno));
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}
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}
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}
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#else
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//----------------------------------------------------------------------------
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// OSS pcm
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//----------------------------------------------------------------------------
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@ -2392,12 +2488,71 @@ static int OssUsedBytes(void)
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return OssRingBuffer ? RingBufferUsedBytes(OssRingBuffer) : 0;
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}
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#endif
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#ifdef USE_AUDIO_THREAD
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//----------------------------------------------------------------------------
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// thread playback
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//----------------------------------------------------------------------------
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#ifdef USE_AUDIORING
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/**
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** OSS thread
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**
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** @retval -1 error
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** @retval 0 underrun
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** @retval 1 running
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*/
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static int OssThread(void)
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{
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int err;
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if (!OssPcmFildes) {
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usleep(OssFragmentTime * 1000);
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return -1;
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}
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for (;;) {
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struct pollfd fds[1];
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pthread_testcancel();
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if (AudioPaused) {
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return 1;
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}
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// wait for space in kernel buffers
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fds[0].fd = OssPcmFildes;
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fds[0].events = POLLOUT | POLLERR;
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// wait for space in kernel buffers
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err = poll(fds, 1, OssFragmentTime);
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if (err < 0) {
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if (err == EAGAIN) {
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continue;
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}
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Error(_("audio/oss: error poll %s\n"), strerror(errno));
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usleep(OssFragmentTime * 1000);
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return -1;
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}
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break;
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}
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if (!err || AudioPaused) { // timeout or some commands
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return 1;
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}
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if ((err = OssPlayRingbuffer())) { // empty / error
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if (err < 0) { // underrun error
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return -1;
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}
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pthread_yield();
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usleep(OssFragmentTime * 1000); // let fill/empty the buffers
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return 0;
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}
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return 1;
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}
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#else
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/**
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** OSS thread
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*/
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@ -2499,6 +2654,7 @@ static void OssThreadFlushBuffers(void)
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}
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}
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#endif
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#endif
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//----------------------------------------------------------------------------
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@ -2519,7 +2675,10 @@ static int OssOpenPCM(int use_ac3)
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&& !(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) {
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device = "/dev/dsp";
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}
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Info(_("audio/oss: using %sdevice '%s'\n"), use_ac3 ? "ac3" : "", device);
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if (!AudioDoingInit) {
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Info(_("audio/oss: using %sdevice '%s'\n"), use_ac3 ? "ac3 " : "",
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device);
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}
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if ((fildes = open(device, O_WRONLY)) < 0) {
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Error(_("audio/oss: can't open dsp device '%s': %s\n"), device,
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@ -2626,6 +2785,171 @@ static void OssInitMixer(void)
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// OSS API
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//----------------------------------------------------------------------------
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#ifdef USE_AUDIORING
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/**
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** Get OSS audio delay in time stamps.
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**
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** @returns audio delay in time stamps.
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*/
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static int64_t OssGetDelay(void)
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{
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int delay;
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int64_t pts;
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// setup failure
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if (OssPcmFildes == -1 || !AudioRing[AudioRingRead].HwSampleRate) {
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return 0L;
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}
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if (!AudioRunning) { // audio not running
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Error(_("audio/oss: should not happen\n"));
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return 0L;
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}
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// delay in bytes in kernel buffers
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delay = -1;
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if (ioctl(OssPcmFildes, SNDCTL_DSP_GETODELAY, &delay) == -1) {
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Error(_("audio/oss: ioctl(SNDCTL_DSP_GETODELAY): %s\n"),
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strerror(errno));
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return 0L;
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}
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if (delay < 0) {
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delay = 0;
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}
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pts = ((int64_t) delay * 90 * 1000)
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/ (AudioRing[AudioRingRead].HwSampleRate *
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AudioRing[AudioRingRead].HwChannels * AudioBytesProSample);
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return pts;
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}
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/**
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** Setup OSS audio for requested format.
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**
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** @param sample_rate sample rate/frequency
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** @param channels number of channels
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** @param use_ac3 use ac3/pass-through device
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**
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** @retval 0 everything ok
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** @retval 1 didn't support frequency/channels combination
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** @retval -1 something gone wrong
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*/
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static int OssSetup(int *sample_rate, int *channels, int use_ac3)
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{
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int ret;
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int tmp;
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int delay;
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audio_buf_info bi;
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if (OssPcmFildes == -1) { // OSS not ready
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// FIXME: if open fails for ac3, we never recover
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return -1;
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}
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if (1) { // close+open for pcm / ac3
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int fildes;
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fildes = OssPcmFildes;
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OssPcmFildes = -1;
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close(fildes);
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if (!(fildes = OssOpenPCM(use_ac3))) {
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return -1;
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}
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OssPcmFildes = fildes;
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}
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ret = 0;
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tmp = AFMT_S16_NE; // native 16 bits
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if (ioctl(OssPcmFildes, SNDCTL_DSP_SETFMT, &tmp) == -1) {
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Error(_("audio/oss: ioctl(SNDCTL_DSP_SETFMT): %s\n"), strerror(errno));
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// FIXME: stop player, set setup failed flag
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return -1;
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}
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if (tmp != AFMT_S16_NE) {
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Error(_("audio/oss: device doesn't support 16 bit sample format.\n"));
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// FIXME: stop player, set setup failed flag
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return -1;
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}
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tmp = *channels;
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if (ioctl(OssPcmFildes, SNDCTL_DSP_CHANNELS, &tmp) == -1) {
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Error(_("audio/oss: ioctl(SNDCTL_DSP_CHANNELS): %s\n"),
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strerror(errno));
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return -1;
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}
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if (tmp != *channels) {
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Warning(_("audio/oss: device doesn't support %d channels.\n"),
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*channels);
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*channels = tmp;
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ret = 1;
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}
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tmp = *sample_rate;
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if (ioctl(OssPcmFildes, SNDCTL_DSP_SPEED, &tmp) == -1) {
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Error(_("audio/oss: ioctl(SNDCTL_DSP_SPEED): %s\n"), strerror(errno));
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return -1;
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}
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if (tmp != *sample_rate) {
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Warning(_("audio/oss: device doesn't support %dHz sample rate.\n"),
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*sample_rate);
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*sample_rate = tmp;
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ret = 1;
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}
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#ifdef SNDCTL_DSP_POLICY
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tmp = 3;
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if (ioctl(OssPcmFildes, SNDCTL_DSP_POLICY, &tmp) == -1) {
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Error(_("audio/oss: ioctl(SNDCTL_DSP_POLICY): %s\n"), strerror(errno));
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} else {
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Info("audio/oss: set policy to %d\n", tmp);
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}
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#endif
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if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) {
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Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"),
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strerror(errno));
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bi.fragsize = 4096;
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bi.fragstotal = 16;
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} else {
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Debug(3, "audio/oss: %d bytes buffered\n", bi.bytes);
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}
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OssFragmentTime = (bi.fragsize * 1000)
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/ (*sample_rate * *channels * AudioBytesProSample);
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Debug(3, "audio/oss: buffer size %d %dms, fragment size %d %dms\n",
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bi.fragsize * bi.fragstotal, (bi.fragsize * bi.fragstotal * 1000)
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/ (*sample_rate * *channels * AudioBytesProSample), bi.fragsize,
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OssFragmentTime);
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// start when enough bytes for initial write
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AudioStartThreshold = (bi.fragsize - 1) * bi.fragstotal;
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// buffer time/delay in ms
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delay = AudioBufferTime + 300;
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if (VideoAudioDelay > 0) {
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delay += VideoAudioDelay / 90;
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}
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if (AudioStartThreshold <
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(*sample_rate * *channels * AudioBytesProSample * delay) / 1000U) {
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AudioStartThreshold =
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(*sample_rate * *channels * AudioBytesProSample * delay) / 1000U;
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}
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// no bigger, than 1/3 the buffer
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if (AudioStartThreshold > AudioRingBufferSize / 3) {
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AudioStartThreshold = AudioRingBufferSize / 3;
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}
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if (!AudioDoingInit) {
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Info(_("audio/oss: delay %ums\n"), (AudioStartThreshold * 1000)
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/ (*sample_rate * *channels * AudioBytesProSample));
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}
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return ret;
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}
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#else
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/**
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** Get OSS audio delay in time stamps.
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**
|
||||
@ -2795,6 +3119,8 @@ static int OssSetup(int *freq, int *channels, int use_ac3)
|
||||
return ret;
|
||||
}
|
||||
|
||||
#endif
|
||||
|
||||
/**
|
||||
** Play audio.
|
||||
*/
|
||||
@ -2814,7 +3140,9 @@ void OssPause(void)
|
||||
*/
|
||||
static void OssInit(void)
|
||||
{
|
||||
#ifndef USE_AUDIORING
|
||||
OssRingBuffer = RingBufferNew(AudioRingBufferSize);
|
||||
#endif
|
||||
|
||||
OssInitPCM();
|
||||
OssInitMixer();
|
||||
@ -2843,17 +3171,25 @@ static const AudioModule OssModule = {
|
||||
.Name = "oss",
|
||||
#ifdef USE_AUDIO_THREAD
|
||||
.Thread = OssThread,
|
||||
#ifdef USE_AUDIORING
|
||||
//.Enqueue = OssThreadEnqueue,
|
||||
//.VideoReady = OssVideoReady,
|
||||
.FlushBuffers = OssFlushBuffers,
|
||||
#else
|
||||
.Enqueue = OssThreadEnqueue,
|
||||
.VideoReady = OssVideoReady,
|
||||
.FlushBuffers = OssThreadFlushBuffers,
|
||||
#endif
|
||||
#else
|
||||
.Enqueue = OssEnqueue,
|
||||
.VideoReady = OssVideoReady,
|
||||
.FlushBuffers = OssFlushBuffers,
|
||||
#endif
|
||||
#ifndef USE_AUDIORING
|
||||
.Poller = OssPoller,
|
||||
.FreeBytes = OssFreeBytes,
|
||||
.UsedBytes = OssUsedBytes,
|
||||
#endif
|
||||
.GetDelay = OssGetDelay,
|
||||
.SetVolume = OssSetVolume,
|
||||
.Setup = OssSetup,
|
||||
@ -2869,6 +3205,8 @@ static const AudioModule OssModule = {
|
||||
// Noop
|
||||
//============================================================================
|
||||
|
||||
#ifndef USE_AUDIORING
|
||||
|
||||
/**
|
||||
** Noop enqueue samples.
|
||||
**
|
||||
@ -2897,6 +3235,8 @@ static int NoopUsedBytes(void)
|
||||
return 0; // no driver, nothing used
|
||||
}
|
||||
|
||||
#endif
|
||||
|
||||
/**
|
||||
** Get audio delay in time stamps.
|
||||
**
|
||||
@ -2943,12 +3283,16 @@ static void NoopVoid(void)
|
||||
*/
|
||||
static const AudioModule NoopModule = {
|
||||
.Name = "noop",
|
||||
#ifndef USE_AUDIORING
|
||||
.Enqueue = NoopEnqueue,
|
||||
.VideoReady = NoopVoid,
|
||||
#endif
|
||||
.FlushBuffers = NoopVoid,
|
||||
#ifndef USE_AUDIORING
|
||||
.Poller = NoopVoid,
|
||||
.FreeBytes = NoopFreeBytes,
|
||||
.UsedBytes = NoopUsedBytes,
|
||||
#endif
|
||||
.GetDelay = NoopGetDelay,
|
||||
.SetVolume = NoopSetVolume,
|
||||
.Setup = NoopSetup,
|
||||
@ -3388,7 +3732,9 @@ void AudioFlushBuffers(void)
|
||||
*/
|
||||
void AudioPoller(void)
|
||||
{
|
||||
#ifndef USE_AUDIORING
|
||||
AudioUsedModule->Poller();
|
||||
#endif
|
||||
}
|
||||
|
||||
/**
|
||||
|
||||
Loading…
Reference in New Issue
Block a user