diff options
| -rw-r--r-- | Todo | 7 | ||||
| -rw-r--r-- | audio.c | 378 |
2 files changed, 364 insertions, 21 deletions
@@ -37,6 +37,7 @@ video: radio no need to wait on video buffers starting with radio and own X11 server, shows no video some low-bandwidth tv channels have hiccups. + check start with 24Hz display rate vdpau: software decoder path not working @@ -73,10 +74,8 @@ x11: support embedded mode audio: - Combine alsa+oss ringbuffer code. Make alsa thread/polled and oss thread/polled output module runtime selectable. - software volume support (could be done with asound.conf) Mute should do a real mute and not only set volume to zero. Starting suspended and muted, didn't register the mute. Relaxed audio sync checks at end of packet and already in sync @@ -84,12 +83,10 @@ audio: only wait for video start, if video is running. Not primary device, don't use and block audio/video. multiple open of audio device, reduce them. + Not all channel conversions are written (f.e. 2->3 ... 5->6 ...) audio/alsa: - better downmix of >2 channels on 2 channel hardware remix support of unsupported sample rates - libav supports only resample of mono to 2 channels - ffmpeg didn't support resample of 5 to 2 channels audio/oss: alsa oss emulation mixer "pcm" not working @@ -35,13 +35,12 @@ /// /// /// @todo FIXME: there can be problems with little/big endian. -/// @todo FIXME: can combine OSS and alsa ring buffer /// //#define USE_ALSA ///< enable alsa support //#define USE_OSS ///< enable OSS support #define USE_AUDIO_THREAD ///< use thread for audio playback -#define USE_AUDIORING ///< new audio ring code (incomplete) +#define USE_AUDIORING ///< new audio ring code (testing) #include <stdio.h> #include <stdint.h> @@ -107,12 +106,16 @@ typedef struct _audio_module_ const char *Name; ///< audio output module name int (*const Thread) (void); ///< module thread handler +#ifndef USE_AUDIORING void (*const Enqueue) (const void *, int); ///< enqueue samples for output void (*const VideoReady) (void); ///< video ready, start audio +#endif void (*const FlushBuffers) (void); ///< flush sample buffers +#ifndef USE_AUDIORING void (*const Poller) (void); ///< output poller int (*const FreeBytes) (void); ///< number of bytes free in buffer int (*const UsedBytes) (void); ///< number of bytes used in buffer +#endif int64_t(*const GetDelay) (void); ///< get current audio delay void (*const SetVolume) (int); ///< set output volume int (*const Setup) (int *, int *, int); ///< setup channels, samplerate @@ -142,13 +145,14 @@ static char AudioDoingInit; ///> flag in init, reduce error static volatile char AudioRunning; ///< thread running / stopped static volatile char AudioPaused; ///< audio paused static volatile char AudioVideoIsReady; ///< video ready start early + +#ifndef USE_AUDIORING 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 - -#ifndef USE_AUDIO_THREAD #endif +static const int AudioBytesProSample = 2; ///< number of bytes per sample + static int AudioBufferTime = 336; ///< audio buffer time in ms #ifdef USE_AUDIO_THREAD @@ -170,7 +174,7 @@ static int AudioMaxNormalize; ///< max. normalize factor static int AudioCompressionFactor; ///< current compression factor static int AudioMaxCompression; ///< max. compression factor static int AudioStereoDescent; ///< volume descent for stereo -static int AudioVolume; ///< volume (0 .. 1000) +static int AudioVolume; ///< current volume (0 .. 1000) extern int VideoAudioDelay; ///< import audio/video delay @@ -752,14 +756,15 @@ static int AlsaRatio; ///< internal -> mixer ratio * 1000 static int AlsaPlayRingbuffer(void) { int first; - int avail; - int n; - int err; - int frames; - const void *p; first = 1; for (;;) { // loop for ring buffer wrap + int avail; + int n; + int err; + int frames; + const void *p; + // how many bytes can be written? n = snd_pcm_avail_update(AlsaPCMHandle); if (n < 0) { @@ -821,6 +826,7 @@ static int AlsaPlayRingbuffer(void) if (AudioSoftVolume && !AudioRing[AudioRingRead].UseAc3) { // FIXME: quick&dirty cast AudioSoftAmplifier((int16_t *) p, avail); + // FIXME: if not all are written, we double amplify them } frames = snd_pcm_bytes_to_frames(AlsaPCMHandle, avail); @@ -1658,9 +1664,9 @@ static int64_t AlsaGetDelay(void) /** ** Setup alsa audio for requested format. ** -** @param freq sample frequency -** @param channels number of channels -** @param use_ac3 use ac3/pass-through device +** @param freq sample frequency +** @param channels number of channels +** @param use_ac3 use ac3/pass-through device ** ** @retval 0 everything ok ** @retval 1 didn't support frequency/channels combination @@ -2209,14 +2215,104 @@ static const AudioModule AlsaModule = { static int OssPcmFildes = -1; ///< pcm file descriptor static int OssMixerFildes = -1; ///< mixer file descriptor static int OssMixerChannel; ///< mixer channel index -static RingBuffer *OssRingBuffer; ///< audio ring buffer static int OssFragmentTime; ///< fragment time in ms + +#ifndef USE_AUDIORING +static RingBuffer *OssRingBuffer; ///< audio ring buffer static unsigned OssStartThreshold; ///< start play, if filled +#endif #ifdef USE_AUDIO_THREAD static volatile char OssFlushBuffer; ///< flag empty buffer #endif +#ifdef USE_AUDIORING + +//---------------------------------------------------------------------------- +// OSS pcm +//---------------------------------------------------------------------------- + +/** +** Play samples from ringbuffer. +** +** @retval 0 ok +** @retval 1 ring buffer empty +** @retval -1 underrun error +*/ +static int OssPlayRingbuffer(void) +{ + int first; + + first = 1; + for (;;) { + audio_buf_info bi; + const void *p; + int n; + + if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) { + Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"), + strerror(errno)); + return -1; + } + Debug(4, "audio/oss: %d bytes free\n", bi.bytes); + + n = RingBufferGetReadPointer(AudioRing[AudioRingRead].RingBuffer, &p); + if (!n) { // ring buffer empty + if (first) { // only error on first loop + return 1; + } + return 0; + } + if (n < bi.bytes) { // not enough bytes in ring buffer + bi.bytes = n; + } + if (bi.bytes <= 0) { // full or buffer empty + break; // bi.bytes could become negative! + } + + if (AudioSoftVolume && !AudioRing[AudioRingRead].UseAc3) { + // FIXME: quick&dirty cast + AudioSoftAmplifier((int16_t *) p, bi.bytes); + // FIXME: if not all are written, we double amplify them + } + for (;;) { + n = write(OssPcmFildes, p, bi.bytes); + if (n != bi.bytes) { + if (n < 0) { + if (n == EAGAIN) { + continue; + } + Error(_("audio/oss: write error: %s\n"), strerror(errno)); + return 1; + } + Warning(_("audio/oss: error not all bytes written\n")); + } + break; + } + // advance how many could written + RingBufferReadAdvance(AudioRing[AudioRingRead].RingBuffer, n); + first = 0; + } + + return 0; +} + +/** +** Flush OSS buffers. +*/ +static void OssFlushBuffers(void) +{ + if (OssPcmFildes != -1) { + // flush kernel buffers + if (ioctl(OssPcmFildes, SNDCTL_DSP_HALT_OUTPUT, NULL) < 0) { + Error(_("audio/oss: ioctl(SNDCTL_DSP_HALT_OUTPUT): %s\n"), + strerror(errno)); + } + } +} + +#else + //---------------------------------------------------------------------------- // OSS pcm //---------------------------------------------------------------------------- @@ -2392,12 +2488,71 @@ static int OssUsedBytes(void) return OssRingBuffer ? RingBufferUsedBytes(OssRingBuffer) : 0; } +#endif + #ifdef USE_AUDIO_THREAD //---------------------------------------------------------------------------- // thread playback //---------------------------------------------------------------------------- +#ifdef USE_AUDIORING + +/** +** OSS thread +** +** @retval -1 error +** @retval 0 underrun +** @retval 1 running +*/ +static int OssThread(void) +{ + int err; + + if (!OssPcmFildes) { + usleep(OssFragmentTime * 1000); + return -1; + } + for (;;) { + struct pollfd fds[1]; + + pthread_testcancel(); + if (AudioPaused) { + return 1; + } + // wait for space in kernel buffers + fds[0].fd = OssPcmFildes; + fds[0].events = POLLOUT | POLLERR; + // wait for space in kernel buffers + err = poll(fds, 1, OssFragmentTime); + if (err < 0) { + if (err == EAGAIN) { + continue; + } + Error(_("audio/oss: error poll %s\n"), strerror(errno)); + usleep(OssFragmentTime * 1000); + return -1; + } + break; + } + if (!err || AudioPaused) { // timeout or some commands + return 1; + } + + if ((err = OssPlayRingbuffer())) { // empty / error + if (err < 0) { // underrun error + return -1; + } + pthread_yield(); + usleep(OssFragmentTime * 1000); // let fill/empty the buffers + return 0; + } + + return 1; +} + +#else + /** ** OSS thread */ @@ -2500,6 +2655,7 @@ static void OssThreadFlushBuffers(void) } #endif +#endif //---------------------------------------------------------------------------- @@ -2519,7 +2675,10 @@ static int OssOpenPCM(int use_ac3) && !(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) { device = "/dev/dsp"; } - Info(_("audio/oss: using %sdevice '%s'\n"), use_ac3 ? "ac3" : "", device); + if (!AudioDoingInit) { + Info(_("audio/oss: using %sdevice '%s'\n"), use_ac3 ? "ac3 " : "", + device); + } if ((fildes = open(device, O_WRONLY)) < 0) { Error(_("audio/oss: can't open dsp device '%s': %s\n"), device, @@ -2626,6 +2785,171 @@ static void OssInitMixer(void) // OSS API //---------------------------------------------------------------------------- +#ifdef USE_AUDIORING + +/** +** Get OSS audio delay in time stamps. +** +** @returns audio delay in time stamps. +*/ +static int64_t OssGetDelay(void) +{ + int delay; + int64_t pts; + + // setup failure + if (OssPcmFildes == -1 || !AudioRing[AudioRingRead].HwSampleRate) { + return 0L; + } + if (!AudioRunning) { // audio not running + Error(_("audio/oss: should not happen\n")); + return 0L; + } + // delay in bytes in kernel buffers + delay = -1; + if (ioctl(OssPcmFildes, SNDCTL_DSP_GETODELAY, &delay) == -1) { + Error(_("audio/oss: ioctl(SNDCTL_DSP_GETODELAY): %s\n"), + strerror(errno)); + return 0L; + } + if (delay < 0) { + delay = 0; + } + + pts = ((int64_t) delay * 90 * 1000) + / (AudioRing[AudioRingRead].HwSampleRate * + AudioRing[AudioRingRead].HwChannels * AudioBytesProSample); + + return pts; +} + +/** +** Setup OSS audio for requested format. +** +** @param sample_rate sample rate/frequency +** @param channels number of channels +** @param use_ac3 use ac3/pass-through device +** +** @retval 0 everything ok +** @retval 1 didn't support frequency/channels combination +** @retval -1 something gone wrong +*/ +static int OssSetup(int *sample_rate, int *channels, int use_ac3) +{ + int ret; + int tmp; + int delay; + audio_buf_info bi; + + if (OssPcmFildes == -1) { // OSS not ready + // FIXME: if open fails for ac3, we never recover + return -1; + } + + if (1) { // close+open for pcm / ac3 + int fildes; + + fildes = OssPcmFildes; + OssPcmFildes = -1; + close(fildes); + if (!(fildes = OssOpenPCM(use_ac3))) { + return -1; + } + OssPcmFildes = fildes; + } + + ret = 0; + + tmp = AFMT_S16_NE; // native 16 bits + if (ioctl(OssPcmFildes, SNDCTL_DSP_SETFMT, &tmp) == -1) { + Error(_("audio/oss: ioctl(SNDCTL_DSP_SETFMT): %s\n"), strerror(errno)); + // FIXME: stop player, set setup failed flag + return -1; + } + if (tmp != AFMT_S16_NE) { + Error(_("audio/oss: device doesn't support 16 bit sample format.\n")); + // FIXME: stop player, set setup failed flag + return -1; + } + + tmp = *channels; + if (ioctl(OssPcmFildes, SNDCTL_DSP_CHANNELS, &tmp) == -1) { + Error(_("audio/oss: ioctl(SNDCTL_DSP_CHANNELS): %s\n"), + strerror(errno)); + return -1; + } + if (tmp != *channels) { + Warning(_("audio/oss: device doesn't support %d channels.\n"), + *channels); + *channels = tmp; + ret = 1; + } + + tmp = *sample_rate; + if (ioctl(OssPcmFildes, SNDCTL_DSP_SPEED, &tmp) == -1) { + Error(_("audio/oss: ioctl(SNDCTL_DSP_SPEED): %s\n"), strerror(errno)); + return -1; + } + if (tmp != *sample_rate) { + Warning(_("audio/oss: device doesn't support %dHz sample rate.\n"), + *sample_rate); + *sample_rate = tmp; + ret = 1; + } +#ifdef SNDCTL_DSP_POLICY + tmp = 3; + if (ioctl(OssPcmFildes, SNDCTL_DSP_POLICY, &tmp) == -1) { + Error(_("audio/oss: ioctl(SNDCTL_DSP_POLICY): %s\n"), strerror(errno)); + } else { + Info("audio/oss: set policy to %d\n", tmp); + } +#endif + + if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) { + Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"), + strerror(errno)); + bi.fragsize = 4096; + bi.fragstotal = 16; + } else { + Debug(3, "audio/oss: %d bytes buffered\n", bi.bytes); + } + + OssFragmentTime = (bi.fragsize * 1000) + / (*sample_rate * *channels * AudioBytesProSample); + + Debug(3, "audio/oss: buffer size %d %dms, fragment size %d %dms\n", + bi.fragsize * bi.fragstotal, (bi.fragsize * bi.fragstotal * 1000) + / (*sample_rate * *channels * AudioBytesProSample), bi.fragsize, + OssFragmentTime); + + // start when enough bytes for initial write + AudioStartThreshold = (bi.fragsize - 1) * bi.fragstotal; + + // buffer time/delay in ms + delay = AudioBufferTime + 300; + if (VideoAudioDelay > 0) { + delay += VideoAudioDelay / 90; + } + if (AudioStartThreshold < + (*sample_rate * *channels * AudioBytesProSample * delay) / 1000U) { + AudioStartThreshold = + (*sample_rate * *channels * AudioBytesProSample * delay) / 1000U; + } + // no bigger, than 1/3 the buffer + if (AudioStartThreshold > AudioRingBufferSize / 3) { + AudioStartThreshold = AudioRingBufferSize / 3; + } + + if (!AudioDoingInit) { + Info(_("audio/oss: delay %ums\n"), (AudioStartThreshold * 1000) + / (*sample_rate * *channels * AudioBytesProSample)); + } + + return ret; +} + +#else + /** ** Get OSS audio delay in time stamps. ** @@ -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 } /** |