/// /// @file audio.c @brief Audio module /// /// Copyright (c) 2009 - 2012 by Johns. All Rights Reserved. /// /// Contributor(s): /// /// License: AGPLv3 /// /// This program is free software: you can redistribute it and/or modify /// it under the terms of the GNU Affero General Public License as /// published by the Free Software Foundation, either version 3 of the /// License. /// /// This program is distributed in the hope that it will be useful, /// but WITHOUT ANY WARRANTY; without even the implied warranty of /// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the /// GNU Affero General Public License for more details. /// /// $Id$ ////////////////////////////////////////////////////////////////////////////// /// /// @defgroup Audio The audio module. /// /// This module contains all audio output functions. /// /// ALSA PCM/Mixer api is supported. /// @see http://www.alsa-project.org/alsa-doc/alsa-lib /// /// @note alsa async playback is broken, don't use it! /// /// OSS PCM/Mixer api is supported. /// @see http://manuals.opensound.com/developer/ /// /// /// @todo FIXME: there can be problems with little/big endian. /// //#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 (testing) #include #include #include #include #include #include #include #define _(str) gettext(str) ///< gettext shortcut #define _N(str) str ///< gettext_noop shortcut #ifdef USE_ALSA #include #endif #ifdef USE_OSS #include #include #include #include // SNDCTL_DSP_HALT_OUTPUT compatibility #ifndef SNDCTL_DSP_HALT_OUTPUT # if defined(SNDCTL_DSP_RESET_OUTPUT) # define SNDCTL_DSP_HALT_OUTPUT SNDCTL_DSP_RESET_OUTPUT # elif defined(SNDCTL_DSP_RESET) # define SNDCTL_DSP_HALT_OUTPUT SNDCTL_DSP_RESET # else # error "No valid SNDCTL_DSP_HALT_OUTPUT found." # endif #endif #include #include #include #include #endif #ifdef USE_AUDIO_THREAD #ifndef __USE_GNU #define __USE_GNU #endif #include #ifndef HAVE_PTHREAD_NAME /// only available with newer glibc #define pthread_setname_np(thread, name) #endif #endif #include // portable atomic_t #include "ringbuffer.h" #include "misc.h" #include "audio.h" //---------------------------------------------------------------------------- // Declarations //---------------------------------------------------------------------------- /** ** Audio output module structure and typedef. */ 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 void (*const Play) (void); ///< play void (*const Pause) (void); ///< pause void (*const Init) (void); ///< initialize audio output module void (*const Exit) (void); ///< cleanup audio output module } AudioModule; static const AudioModule NoopModule; ///< forward definition of noop module //---------------------------------------------------------------------------- // Variables //---------------------------------------------------------------------------- char AudioAlsaDriverBroken; ///< disable broken driver message static const char *AudioModuleName; ///< which audio module to use /// Selected audio module. static const AudioModule *AudioUsedModule = &NoopModule; static const char *AudioPCMDevice; ///< PCM device name static const char *AudioAC3Device; ///< AC3 device name static const char *AudioMixerDevice; ///< mixer device name static const char *AudioMixerChannel; ///< mixer channel name 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 int64_t AudioPTS; ///< audio pts clock #endif static const int AudioBytesProSample = 2; ///< number of bytes per sample static int AudioBufferTime = 336; ///< audio buffer time in ms #ifdef USE_AUDIO_THREAD static pthread_t AudioThread; ///< audio play thread static pthread_mutex_t AudioMutex; ///< audio condition mutex static pthread_cond_t AudioStartCond; ///< condition variable #else static const int AudioThread; ///< dummy audio thread #endif static char AudioSoftVolume; ///< flag use soft volume static char AudioNormalize; ///< flag use volume normalize static char AudioCompression; ///< flag use compress volume static char AudioMute; ///< flag muted static int AudioAmplifier; ///< software volume factor static int AudioNormalizeFactor; ///< current normalize factor static const int AudioMinNormalize = 100; ///< min. normalize factor 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; ///< current volume (0 .. 1000) extern int VideoAudioDelay; ///< import audio/video delay /// default ring buffer size ~2s 8ch 16bit static const unsigned AudioRingBufferSize = 2 * 48000 * 8 * 2; static int AudioChannelsInHw[9]; ///< table which channels are supported enum _audio_rates { ///< sample rates enumeration // HW: 32000 44100 48000 88200 96000 176400 192000 //Audio32000, ///< 32.0Khz Audio44100, ///< 44.1Khz Audio48000, ///< 48.0Khz //Audio88200, ///< 88.2Khz //Audio96000, ///< 96.0Khz //Audio176400, ///< 176.4Khz //Audio192000, ///< 192.0Khz AudioRatesMax ///< max index }; /// table which rates are supported static int AudioRatesInHw[AudioRatesMax]; /// input to hardware channel matrix static int AudioChannelMatrix[AudioRatesMax][9]; /// rates tables static const unsigned AudioRatesTable[AudioRatesMax] = { 44100, 48000, }; #ifdef USE_AUDIORING //---------------------------------------------------------------------------- // filter //---------------------------------------------------------------------------- static const int AudioNormSamples = 4096; ///< number of samples #define AudioNormMaxIndex 128 ///< number of average values /// average of n last sample blocks static uint32_t AudioNormAverage[AudioNormMaxIndex]; static int AudioNormIndex; ///< index into average table static int AudioNormReady; ///< index counter static int AudioNormCounter; ///< sample counter /** ** Audio normalizer. ** ** @param samples sample buffer ** @param count number of bytes in sample buffer */ static void AudioNormalizer(int16_t * samples, int count) { int i; int l; int n; uint32_t avg; int factor; int16_t *data; // average samples l = count / AudioBytesProSample; data = samples; do { n = l; if (AudioNormCounter + n > AudioNormSamples) { n = AudioNormSamples - AudioNormCounter; } avg = AudioNormAverage[AudioNormIndex]; for (i = 0; i < n; ++i) { int t; t = data[i]; avg += (t * t) / AudioNormSamples; } AudioNormAverage[AudioNormIndex] = avg; AudioNormCounter += n; if (AudioNormCounter >= AudioNormSamples) { if (AudioNormReady < AudioNormMaxIndex) { AudioNormReady++; } else { avg = 0; for (i = 0; i < AudioNormMaxIndex; ++i) { avg += AudioNormAverage[i] / AudioNormMaxIndex; } // calculate normalize factor if (avg > 0) { factor = ((INT16_MAX / 8) * 1000U) / (uint32_t) sqrt(avg); // smooth normalize AudioNormalizeFactor = (AudioNormalizeFactor * 500 + factor * 500) / 1000; if (AudioNormalizeFactor < AudioMinNormalize) { AudioNormalizeFactor = AudioMinNormalize; } if (AudioNormalizeFactor > AudioMaxNormalize) { AudioNormalizeFactor = AudioMaxNormalize; } } else { factor = 1000; } Debug(4, "audio/noramlize: avg %8d, fac=%6.3f, norm=%6.3f\n", avg, factor / 1000.0, AudioNormalizeFactor / 1000.0); } AudioNormIndex = (AudioNormIndex + 1) % AudioNormMaxIndex; AudioNormCounter = 0; AudioNormAverage[AudioNormIndex] = 0U; } data += n; l -= n; } while (l > 0); // apply normalize factor for (i = 0; i < count / AudioBytesProSample; ++i) { int t; t = (samples[i] * AudioNormalizeFactor) / 1000; if (t < INT16_MIN) { t = INT16_MIN; } else if (t > INT16_MAX) { t = INT16_MAX; } samples[i] = t; } } /** ** Reset normalizer. */ static void AudioResetNormalizer(void) { int i; AudioNormCounter = 0; AudioNormReady = 0; for (i = 0; i < AudioNormMaxIndex; ++i) { AudioNormAverage[i] = 0U; } AudioNormalizeFactor = 1000; } /** ** Audio compression. ** ** @param samples sample buffer ** @param count number of bytes in sample buffer */ static void AudioCompressor(int16_t * samples, int count) { int max_sample; int i; int factor; // find loudest sample max_sample = 0; for (i = 0; i < count / AudioBytesProSample; ++i) { int t; t = abs(samples[i]); if (t > max_sample) { max_sample = t; } } // calculate compression factor if (max_sample > 0) { factor = (INT16_MAX * 1000) / max_sample; // smooth compression (FIXME: make configurable?) AudioCompressionFactor = (AudioCompressionFactor * 950 + factor * 50) / 1000; if (AudioCompressionFactor > factor) { AudioCompressionFactor = factor; // no clipping } if (AudioCompressionFactor > AudioMaxCompression) { AudioCompressionFactor = AudioMaxCompression; } } else { return; // silent nothing todo } Debug(4, "audio/compress: max %5d, fac=%6.3f, com=%6.3f\n", max_sample, factor / 1000.0, AudioCompressionFactor / 1000.0); // apply compression factor for (i = 0; i < count / AudioBytesProSample; ++i) { int t; t = (samples[i] * AudioCompressionFactor) / 1000; if (t < INT16_MIN) { t = INT16_MIN; } else if (t > INT16_MAX) { t = INT16_MAX; } samples[i] = t; } } /** ** Reset compressor. */ static void AudioResetCompressor(void) { AudioCompressionFactor = 2000; if (AudioCompressionFactor > AudioMaxCompression) { AudioCompressionFactor = AudioMaxCompression; } } /** ** Audio software amplifier. ** ** @param samples sample buffer ** @param count number of bytes in sample buffer ** ** @todo FIXME: this does hard clipping */ static void AudioSoftAmplifier(int16_t * samples, int count) { int i; // silence if (AudioMute || !AudioAmplifier) { memset(samples, 0, count); return; } for (i = 0; i < count / AudioBytesProSample; ++i) { int t; t = (samples[i] * AudioAmplifier) / 1000; if (t < INT16_MIN) { t = INT16_MIN; } else if (t > INT16_MAX) { t = INT16_MAX; } samples[i] = t; } } /** ** Upmix mono to stereo. ** ** @param in input sample buffer ** @param frames number of frames in sample buffer ** @param out output sample buffer */ static void AudioMono2Stereo(const int16_t * in, int frames, int16_t * out) { int i; for (i = 0; i < frames; ++i) { int t; t = in[i]; out[i * 2 + 0] = t; out[i * 2 + 1] = t; } } /** ** Downmix stereo to mono. ** ** @param in input sample buffer ** @param frames number of frames in sample buffer ** @param out output sample buffer */ static void AudioStereo2Mono(const int16_t * in, int frames, int16_t * out) { int i; for (i = 0; i < frames; i += 2) { out[i / 2] = (in[i + 0] + in[i + 1]) / 2; } } /** ** Downmix surround to stereo. ** ** ffmpeg L R C Ls Rs -> alsa L R Ls Rs C ** ffmpeg L R C LFE Ls Rs -> alsa L R Ls Rs C LFE ** ffmpeg L R C LFE Ls Rs Rl Rr -> alsa L R Ls Rs C LFE Rl Rr ** ** @param in input sample buffer ** @param in_chan nr. of input channels ** @param frames number of frames in sample buffer ** @param out output sample buffer */ static void AudioSurround2Stereo(const int16_t * in, int in_chan, int frames, int16_t * out) { while (frames--) { int l; int r; switch (in_chan) { case 3: // stereo or surround? =>stereo l = in[0] * 600; // L r = in[1] * 600; // R l += in[2] * 400; // C r += in[2] * 400; break; case 4: // quad or surround? =>quad l = in[0] * 600; // L r = in[1] * 600; // R l += in[2] * 400; // Ls r += in[3] * 400; // Rs break; case 5: // 5.0 l = in[0] * 500; // L r = in[1] * 500; // R l += in[2] * 200; // Ls r += in[3] * 200; // Rs l += in[4] * 300; // C r += in[4] * 300; break; case 6: // 5.1 l = in[0] * 400; // L r = in[1] * 400; // R l += in[2] * 200; // Ls r += in[3] * 200; // Rs l += in[4] * 300; // C r += in[4] * 300; l += in[5] * 300; // LFE r += in[5] * 100; break; case 7: // 7.0 l = in[0] * 400; // L r = in[1] * 400; // R l += in[2] * 200; // Ls r += in[3] * 200; // Rs l += in[4] * 300; // C r += in[4] * 300; l += in[5] * 100; // RL r += in[6] * 100; // RR break; case 8: // 7.1 l = in[0] * 400; // L r = in[1] * 400; // R l += in[2] * 150; // Ls r += in[3] * 150; // Rs l += in[4] * 250; // C r += in[4] * 250; l += in[5] * 100; // LFE r += in[5] * 100; l += in[6] * 100; // RL r += in[7] * 100; // RR break; default: abort(); } in += in_chan; out[0] = l / 1000; out[1] = r / 1000; out += 2; } } /** ** Resample ffmpeg sample format to hardware format. ** ** @param in input sample buffer ** @param in_chan nr. of input channels ** @param frames number of frames in sample buffer ** @param out output sample buffer ** @param out_chan nr. of output channels */ static void AudioResample(const int16_t * in, int in_chan, int frames, int16_t * out, int out_chan) { switch (in_chan * 8 + out_chan) { case 1 * 8 + 1: case 2 * 8 + 2: case 3 * 8 + 3: case 4 * 8 + 4: case 5 * 8 + 5: case 6 * 8 + 6: case 7 * 8 + 7: case 8 * 8 + 8: // input = output channels memcpy(out, in, frames * in_chan * AudioBytesProSample); break; case 2 * 8 + 1: AudioStereo2Mono(in, frames, out); break; case 1 * 8 + 2: AudioMono2Stereo(in, frames, out); break; case 3 * 8 + 2: case 4 * 8 + 2: case 5 * 8 + 2: case 6 * 8 + 2: case 7 * 8 + 2: case 8 * 8 + 2: AudioSurround2Stereo(in, in_chan, frames, out); break; default: Error("audio: unsupported %d -> %d channels resample\n", in_chan, out_chan); // play silence memset(out, 0, frames * out_chan * AudioBytesProSample); break; } } //---------------------------------------------------------------------------- // ring buffer //---------------------------------------------------------------------------- #define AUDIO_RING_MAX 8 ///< number of audio ring buffers /** ** Audio ring buffer. */ typedef struct _audio_ring_ring_ { char FlushBuffers; ///< flag: flush buffers char UseAc3; ///< flag: use ac3 pass-through unsigned HwSampleRate; ///< hardware sample rate in Hz unsigned HwChannels; ///< hardware number of channels unsigned InSampleRate; ///< input sample rate in Hz unsigned InChannels; ///< input number of channels int64_t PTS; ///< pts clock RingBuffer *RingBuffer; ///< sample ring buffer } AudioRingRing; /// default ring buffer size ~2s 8ch 16bit //static const unsigned AudioRingBufferSize = 2 * 48000 * 8 * 2; /// ring of audio ring buffers static AudioRingRing AudioRing[AUDIO_RING_MAX]; static int AudioRingWrite; ///< audio ring write pointer static int AudioRingRead; ///< audio ring read pointer static atomic_t AudioRingFilled; ///< how many of the ring is used static unsigned AudioStartThreshold; ///< start play, if filled /** ** Add sample-rate, number of channel change to ring. ** ** @param sample_rate sample-rate frequency ** @param channels number of channels ** @param use_ac3 use ac3/pass-through device ** ** @retval -1 error ** @retval 0 okay */ static int AudioRingAdd(unsigned sample_rate, int channels, int use_ac3) { unsigned u; // search supported sample-rates for (u = 0; u < AudioRatesMax; ++u) { if (AudioRatesTable[u] == sample_rate) { break; } } if (u == AudioRatesMax) { // unsupported sample-rate Error(_("audio: %dHz sample-rate unsupported\n"), sample_rate); return -1; } if (!AudioChannelMatrix[u][channels]) { Error(_("audio: %d channels unsupported\n"), channels); return -1; // unsupported nr. of channels } if (atomic_read(&AudioRingFilled) == AUDIO_RING_MAX) { // no free slot // FIXME: can wait for ring buffer empty Error(_("audio: out of ring buffers\n")); return -1; } AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX; // FIXME: don't flush buffers here AudioRing[AudioRingWrite].FlushBuffers = 1; AudioRing[AudioRingWrite].UseAc3 = use_ac3; AudioRing[AudioRingWrite].InSampleRate = sample_rate; AudioRing[AudioRingWrite].InChannels = channels; AudioRing[AudioRingWrite].HwSampleRate = sample_rate; AudioRing[AudioRingWrite].HwChannels = AudioChannelMatrix[u][channels]; AudioRing[AudioRingWrite].PTS = INT64_C(0x8000000000000000); // reset ring-buffer RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer)); atomic_inc(&AudioRingFilled); #ifdef USE_AUDIO_THREAD if (AudioThread) { // tell thread, that there is something todo AudioRunning = 1; pthread_cond_signal(&AudioStartCond); } #endif return 0; } /** ** Setup audio ring. */ static void AudioRingInit(void) { int i; for (i = 0; i < AUDIO_RING_MAX; ++i) { // ~2s 8ch 16bit AudioRing[i].RingBuffer = RingBufferNew(AudioRingBufferSize); } atomic_set(&AudioRingFilled, 0); } /** ** Cleanup audio ring. */ static void AudioRingExit(void) { int i; for (i = 0; i < AUDIO_RING_MAX; ++i) { if (AudioRing[i].RingBuffer) { RingBufferDel(AudioRing[i].RingBuffer); AudioRing[i].RingBuffer = NULL; } AudioRing[i].HwSampleRate = 0; // checked for valid setup AudioRing[i].InSampleRate = 0; } AudioRingRead = 0; AudioRingWrite = 0; } #endif #ifdef USE_ALSA //============================================================================ // A L S A //============================================================================ //---------------------------------------------------------------------------- // Alsa variables //---------------------------------------------------------------------------- static snd_pcm_t *AlsaPCMHandle; ///< alsa pcm handle static char AlsaCanPause; ///< hw supports pause static int AlsaUseMmap; ///< use mmap #ifndef USE_AUDIORING static RingBuffer *AlsaRingBuffer; ///< audio ring buffer static unsigned AlsaStartThreshold; ///< start play, if filled #ifdef USE_AUDIO_THREAD static volatile char AlsaFlushBuffer; ///< flag empty buffer #endif #endif static snd_mixer_t *AlsaMixer; ///< alsa mixer handle static snd_mixer_elem_t *AlsaMixerElem; ///< alsa pcm mixer element static int AlsaRatio; ///< internal -> mixer ratio * 1000 #ifdef USE_AUDIORING //---------------------------------------------------------------------------- // alsa pcm //---------------------------------------------------------------------------- /** ** Play samples from ringbuffer. ** ** Fill the kernel buffer, as much as possible. ** ** @retval 0 ok ** @retval 1 ring buffer empty ** @retval -1 underrun error */ static int AlsaPlayRingbuffer(void) { int first; 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) { if (n == -EAGAIN) { continue; } Warning(_("audio/alsa: avail underrun error? '%s'\n"), snd_strerror(n)); err = snd_pcm_recover(AlsaPCMHandle, n, 0); if (err >= 0) { continue; } Error(_("audio/alsa: snd_pcm_avail_update(): %s\n"), snd_strerror(n)); return -1; } avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, n); if (avail < 256) { // too much overhead if (first) { // happens with broken alsa drivers if (AudioThread) { if (!AudioAlsaDriverBroken) { Error(_("audio/alsa: broken driver %d state '%s'\n"), avail, snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle))); } // try to recover if (snd_pcm_state(AlsaPCMHandle) == SND_PCM_STATE_PREPARED) { if ((err = snd_pcm_start(AlsaPCMHandle)) < 0) { Error(_("audio/alsa: snd_pcm_start(): %s\n"), snd_strerror(err)); } } usleep(5 * 1000); } } Debug(4, "audio/alsa: break state '%s'\n", snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle))); break; } n = RingBufferGetReadPointer(AudioRing[AudioRingRead].RingBuffer, &p); if (!n) { // ring buffer empty if (first) { // only error on first loop Debug(4, "audio/alsa: empty buffers %d\n", avail); // ring buffer empty // AlsaLowWaterMark = 1; return 1; } return 0; } if (n < avail) { // not enough bytes in ring buffer avail = n; } if (!avail) { // full or buffer empty break; } 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); for (;;) { if (AlsaUseMmap) { err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames); } else { err = snd_pcm_writei(AlsaPCMHandle, p, frames); } //Debug(3, "audio/alsa: wrote %d/%d frames\n", err, frames); if (err != frames) { if (err < 0) { if (err == -EAGAIN) { continue; } /* if (err == -EBADFD) { goto again; } */ Warning(_("audio/alsa: writei underrun error? '%s'\n"), snd_strerror(err)); err = snd_pcm_recover(AlsaPCMHandle, err, 0); if (err >= 0) { continue; } Error(_("audio/alsa: snd_pcm_writei failed: %s\n"), snd_strerror(err)); return -1; } // this could happen, if underrun happened Warning(_("audio/alsa: not all frames written\n")); avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err); } break; } RingBufferReadAdvance(AudioRing[AudioRingRead].RingBuffer, avail); first = 0; } return 0; } /** ** Flush alsa buffers. */ static void AlsaFlushBuffers(void) { if (AlsaPCMHandle) { int err; snd_pcm_state_t state; state = snd_pcm_state(AlsaPCMHandle); Debug(3, "audio/alsa: flush state %s\n", snd_pcm_state_name(state)); if (state != SND_PCM_STATE_OPEN) { if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) { Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err)); } // ****ing alsa crash, when in open state here if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) { Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err)); } } } } #else //---------------------------------------------------------------------------- // alsa pcm //---------------------------------------------------------------------------- /** ** Place samples in ringbuffer. ** ** @param samples sample buffer ** @param count number of bytes in sample buffer ** ** @returns true if play should be started. */ static int AlsaAddToRingbuffer(const void *samples, int count) { int n; n = RingBufferWrite(AlsaRingBuffer, samples, count); if (n != count) { Error(_("audio/alsa: can't place %d samples in ring buffer\n"), count); // too many bytes are lost // FIXME: should skip more, longer skip, but less often? } if (!AudioRunning) { Debug(4, "audio/alsa: start %4zdms\n", (RingBufferUsedBytes(AlsaRingBuffer) * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample)); // forced start if (AlsaStartThreshold * 2 < RingBufferUsedBytes(AlsaRingBuffer)) { return 1; } // enough video + audio buffered if (AudioVideoIsReady && AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) { // restart play-back return 1; } } return 0; } /** ** Play samples from ringbuffer. */ static int AlsaPlayRingbuffer(void) { int first; int avail; int n; int err; int frames; const void *p; first = 1; for (;;) { // how many bytes can be written? n = snd_pcm_avail_update(AlsaPCMHandle); if (n < 0) { if (n == -EAGAIN) { continue; } Error(_("audio/alsa: avail underrun error? '%s'\n"), snd_strerror(n)); err = snd_pcm_recover(AlsaPCMHandle, n, 0); if (err >= 0) { continue; } Error(_("audio/alsa: snd_pcm_avail_update(): %s\n"), snd_strerror(n)); return -1; } avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, n); if (avail < 256) { // too much overhead if (first) { // happens with broken alsa drivers if (AudioThread) { if (!AudioAlsaDriverBroken) { Error(_("audio/alsa: broken driver %d\n"), avail); Error("audio/alsa: state %s\n", snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle))); } if (snd_pcm_state(AlsaPCMHandle) == SND_PCM_STATE_PREPARED) { if ((err = snd_pcm_start(AlsaPCMHandle)) < 0) { Error(_("audio/alsa: snd_pcm_start(): %s\n"), snd_strerror(err)); } } usleep(5 * 1000); } } Debug(4, "audio/alsa: break state %s\n", snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle))); break; } n = RingBufferGetReadPointer(AlsaRingBuffer, &p); if (!n) { // ring buffer empty if (first) { // only error on first loop Debug(4, "audio/alsa: empty buffers %d\n", avail); // ring buffer empty // AlsaLowWaterMark = 1; return 1; } return 0; } if (n < avail) { // not enough bytes in ring buffer avail = n; } if (!avail) { // full or buffer empty break; } frames = snd_pcm_bytes_to_frames(AlsaPCMHandle, avail); again: if (AlsaUseMmap) { err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames); } else { err = snd_pcm_writei(AlsaPCMHandle, p, frames); } //Debug(3, "audio/alsa: wrote %d/%d frames\n", err, frames); if (err != frames) { if (err < 0) { if (err == -EAGAIN) { goto again; } /* if (err == -EBADFD) { goto again; } */ Error(_("audio/alsa: writei underrun error? '%s'\n"), snd_strerror(err)); err = snd_pcm_recover(AlsaPCMHandle, err, 0); if (err >= 0) { goto again; } Error(_("audio/alsa: snd_pcm_writei failed: %s\n"), snd_strerror(err)); return -1; } // this could happen, if underrun happened Error(_("audio/alsa: error not all frames written\n")); avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err); } RingBufferReadAdvance(AlsaRingBuffer, avail); first = 0; } return 0; } /** ** Flush alsa buffers. */ static void AlsaFlushBuffers(void) { int err; snd_pcm_state_t state; if (AlsaRingBuffer && AlsaPCMHandle) { #ifdef DEBUG const void *r; void *w; #endif RingBufferReadAdvance(AlsaRingBuffer, RingBufferUsedBytes(AlsaRingBuffer)); #ifdef DEBUG RingBufferGetWritePointer(AlsaRingBuffer, &w); RingBufferGetReadPointer(AlsaRingBuffer, &r); if (r != w) { Fatal(_("audio/alsa: ringbuffer out of sync %zd-%zd\n"), RingBufferGetWritePointer(AlsaRingBuffer, &w), RingBufferGetReadPointer(AlsaRingBuffer, &r)); abort(); } #endif state = snd_pcm_state(AlsaPCMHandle); Debug(3, "audio/alsa: flush state %s\n", snd_pcm_state_name(state)); if (state != SND_PCM_STATE_OPEN) { if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) { Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err)); } // ****ing alsa crash, when in open state here if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) { Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err)); } } } AudioRunning = 0; AudioVideoIsReady = 0; AudioPTS = INT64_C(0x8000000000000000); } /** ** Call back to play audio polled. */ static void AlsaPoller(void) { if (!AlsaPCMHandle) { // setup failure return; } if (!AudioThread && AudioRunning) { AlsaPlayRingbuffer(); } } /** ** Get free bytes in audio output. */ static int AlsaFreeBytes(void) { return AlsaRingBuffer ? RingBufferFreeBytes(AlsaRingBuffer) : INT32_MAX; } /** ** Get used bytes in audio output. */ static int AlsaUsedBytes(void) { return AlsaRingBuffer ? RingBufferUsedBytes(AlsaRingBuffer) : 0; } #if 0 //---------------------------------------------------------------------------- // async playback //---------------------------------------------------------------------------- // async playback is broken, don't use it! /** ** Alsa async pcm callback function. ** ** @param handler alsa async handler */ static void AlsaAsyncCallback(snd_async_handler_t * handler) { Debug(3, "audio/%s: %p\n", __FUNCTION__, handler); // how many bytes can be written? for (;;) { n = snd_pcm_avail_update(AlsaPCMHandle); if (n < 0) { Error(_("audio/alsa: snd_pcm_avail_update(): %s\n"), snd_strerror(n)); break; } avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, n); if (avail < 512) { // too much overhead break; } n = RingBufferGetReadPointer(AlsaRingBuffer, &p); if (!n) { // ring buffer empty Debug(3, "audio/alsa: ring buffer empty\n"); break; } if (n < avail) { // not enough bytes in ring buffer avail = n; } if (!avail) { // full break; } frames = snd_pcm_bytes_to_frames(AlsaPCMHandle, avail); again: if (AlsaUseMmap) { err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames); } else { err = snd_pcm_writei(AlsaPCMHandle, p, frames); } Debug(3, "audio/alsa: %d => %d\n", frames, err); if (err < 0) { Error(_("audio/alsa: underrun error?\n")); err = snd_pcm_recover(AlsaPCMHandle, err, 0); if (err >= 0) { goto again; } Error(_("audio/alsa: snd_pcm_writei failed: %s\n"), snd_strerror(err)); } if (err != frames) { Error(_("audio/alsa: error not all frames written\n")); avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err); } RingBufferReadAdvance(AlsaRingBuffer, avail); } } /** ** Place samples in audio output queue. ** ** @param samples sample buffer ** @param count number of bytes in sample buffer */ static void AlsaEnqueue(const void *samples, int count) { snd_pcm_state_t state; int n; //int err; Debug(3, "audio: %6zd + %4d\n", RingBufferUsedBytes(AlsaRingBuffer), count); n = RingBufferWrite(AlsaRingBuffer, samples, count); if (n != count) { Fatal(_("audio: can't place %d samples in ring buffer\n"), count); } // check if running, wait until enough buffered state = snd_pcm_state(AlsaPCMHandle); if (state == SND_PCM_STATE_PREPARED) { Debug(3, "audio/alsa: state %d - %s\n", state, snd_pcm_state_name(state)); // FIXME: adjust start ratio if (RingBufferFreeBytes(AlsaRingBuffer) < RingBufferUsedBytes(AlsaRingBuffer)) { // restart play-back #if 0 if (AlsaCanPause) { if ((err = snd_pcm_pause(AlsaPCMHandle, 0))) { Error(_("audio: snd_pcm_pause(): %s\n"), snd_strerror(err)); } } else { if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) { Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err)); } } if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) { Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err)); } Debug(3, "audio/alsa: unpaused\n"); if ((err = snd_pcm_start(AlsaPCMHandle)) < 0) { Error(_("audio: snd_pcm_start(): %s\n"), snd_strerror(err)); } #endif state = snd_pcm_state(AlsaPCMHandle); Debug(3, "audio/alsa: state %s\n", snd_pcm_state_name(state)); Debug(3, "audio/alsa: unpaused\n"); } } } #endif //---------------------------------------------------------------------------- // direct playback //---------------------------------------------------------------------------- // direct play produces underuns on some hardware #ifndef USE_AUDIO_THREAD /** ** Place samples in audio output queue. ** ** @param samples sample buffer ** @param count number of bytes in sample buffer */ static void AlsaEnqueue(const void *samples, int count) { if (AlsaAddToRingbuffer(samples, count)) { AudioRunning = 1; } } #endif #endif #ifdef USE_AUDIO_THREAD //---------------------------------------------------------------------------- // thread playback //---------------------------------------------------------------------------- #ifdef USE_AUDIORING /** ** Alsa thread ** ** Play some samples and return. ** ** @retval -1 error ** @retval 0 underrun ** @retval 1 running */ static int AlsaThread(void) { int err; if (!AlsaPCMHandle) { usleep(24 * 1000); return -1; } for (;;) { pthread_testcancel(); if (AudioPaused) { return 1; } // wait for space in kernel buffers if ((err = snd_pcm_wait(AlsaPCMHandle, 24)) < 0) { Warning(_("audio/alsa: wait underrun error? '%s'\n"), snd_strerror(err)); err = snd_pcm_recover(AlsaPCMHandle, err, 0); if (err >= 0) { continue; } Error(_("audio/alsa: snd_pcm_wait(): %s\n"), snd_strerror(err)); usleep(24 * 1000); return -1; } break; } if (!err || AudioPaused) { // timeout or some commands return 1; } if ((err = AlsaPlayRingbuffer())) { // empty or error snd_pcm_state_t state; if (err < 0) { // underrun error return -1; } state = snd_pcm_state(AlsaPCMHandle); if (state != SND_PCM_STATE_RUNNING) { Debug(3, "audio/alsa: stopping play '%s'\n", snd_pcm_state_name(state)); return 0; } usleep(24 * 1000); // let fill/empty the buffers } return 1; } #else /** ** Alsa thread */ static int AlsaThread(void) { for (;;) { int err; pthread_testcancel(); if (AlsaFlushBuffer) { // we can flush too many, but wo cares Debug(3, "audio/alsa: flushing buffers\n"); AlsaFlushBuffers(); /* if ((err = snd_pcm_prepare(AlsaPCMHandle))) { Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err)); } */ AlsaFlushBuffer = 0; break; } if (AudioPaused) { break; } // wait for space in kernel buffers if ((err = snd_pcm_wait(AlsaPCMHandle, 24)) < 0) { Error(_("audio/alsa: wait underrun error? '%s'\n"), snd_strerror(err)); err = snd_pcm_recover(AlsaPCMHandle, err, 0); if (err >= 0) { continue; } Error(_("audio/alsa: snd_pcm_wait(): %s\n"), snd_strerror(err)); usleep(24 * 1000); continue; } // timeout or some commands if (!err || AlsaFlushBuffer || AudioPaused) { continue; } if ((err = AlsaPlayRingbuffer())) { // empty / error snd_pcm_state_t state; if (err < 0) { // underrun error break; } state = snd_pcm_state(AlsaPCMHandle); if (state != SND_PCM_STATE_RUNNING) { Debug(3, "audio/alsa: stopping play '%s'\n", snd_pcm_state_name(state)); break; } pthread_yield(); usleep(24 * 1000); // let fill/empty the buffers } } return 0; } /** ** Place samples in audio output queue. ** ** @param samples sample buffer ** @param count number of bytes in sample buffer */ static void AlsaThreadEnqueue(const void *samples, int count) { if (!AlsaRingBuffer || !AlsaPCMHandle) { Debug(3, "audio/alsa: enqueue not ready\n"); return; } if (AlsaAddToRingbuffer(samples, count)) { snd_pcm_state_t state; state = snd_pcm_state(AlsaPCMHandle); Debug(3, "audio/alsa: enqueue state %s\n", snd_pcm_state_name(state)); // no lock needed, can wakeup next time AudioRunning = 1; pthread_cond_signal(&AudioStartCond); } } /** ** Video is ready, start audio if possible, */ static void AlsaVideoReady(void) { if (!AudioRunning) { size_t used; used = RingBufferUsedBytes(AlsaRingBuffer); // enough video + audio buffered if (AlsaStartThreshold < used) { // too much audio buffered, skip it if (AlsaStartThreshold * 2 < used) { Debug(3, "audio/alsa: start %4zdms skip ready\n", ((used - AlsaStartThreshold * 2) * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample)); RingBufferReadAdvance(AlsaRingBuffer, used - AlsaStartThreshold * 2); } AudioRunning = 1; pthread_cond_signal(&AudioStartCond); } } if (AudioSampleRate && AudioChannels) { Debug(3, "audio/alsa: start %4zdms video ready\n", (RingBufferUsedBytes(AlsaRingBuffer) * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample)); } } /** ** Flush alsa buffers with thread. */ static void AlsaThreadFlushBuffers(void) { // signal thread to flush buffers if (AudioThread) { AlsaFlushBuffer = 1; do { AudioRunning = 1; // wakeup in case of sleeping pthread_cond_signal(&AudioStartCond); usleep(1 * 1000); } while (AlsaFlushBuffer); // wait until flushed } } #endif #endif //---------------------------------------------------------------------------- /** ** Open alsa pcm device. ** ** @param use_ac3 use ac3/pass-through device */ static snd_pcm_t *AlsaOpenPCM(int use_ac3) { const char *device; snd_pcm_t *handle; int err; // &&|| hell if (!(use_ac3 && ((device = AudioAC3Device) || (device = getenv("ALSA_AC3_DEVICE")))) && !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) { device = "default"; } if (!AudioDoingInit) { Info(_("audio/alsa: using %sdevice '%s'\n"), use_ac3 ? "ac3 " : "", device); } // open none blocking; if device is already used, we don't want wait if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK)) < 0) { Error(_("audio/alsa: playback open '%s' error: %s\n"), device, snd_strerror(err)); return NULL; } if ((err = snd_pcm_nonblock(handle, 0)) < 0) { Error(_("audio/alsa: can't set block mode: %s\n"), snd_strerror(err)); } return handle; } /** ** Initialize alsa pcm device. ** ** @see AudioPCMDevice */ static void AlsaInitPCM(void) { snd_pcm_t *handle; snd_pcm_hw_params_t *hw_params; int err; if (!(handle = AlsaOpenPCM(0))) { return; } // FIXME: pass-through and pcm out can support different features snd_pcm_hw_params_alloca(&hw_params); // choose all parameters if ((err = snd_pcm_hw_params_any(handle, hw_params)) < 0) { Error(_ ("audio: snd_pcm_hw_params_any: no configurations available: %s\n"), snd_strerror(err)); } AlsaCanPause = snd_pcm_hw_params_can_pause(hw_params); Info(_("audio/alsa: supports pause: %s\n"), AlsaCanPause ? "yes" : "no"); AlsaPCMHandle = handle; } //---------------------------------------------------------------------------- // Alsa Mixer //---------------------------------------------------------------------------- /** ** Set alsa mixer volume (0-1000) ** ** @param volume volume (0 .. 1000) */ static void AlsaSetVolume(int volume) { int v; if (AlsaMixer && AlsaMixerElem) { v = (volume * AlsaRatio) / (1000 * 1000); snd_mixer_selem_set_playback_volume(AlsaMixerElem, 0, v); snd_mixer_selem_set_playback_volume(AlsaMixerElem, 1, v); } } /** ** Initialize alsa mixer. */ static void AlsaInitMixer(void) { const char *device; const char *channel; snd_mixer_t *alsa_mixer; snd_mixer_elem_t *alsa_mixer_elem; long alsa_mixer_elem_min; long alsa_mixer_elem_max; if (!(device = AudioMixerDevice)) { if (!(device = getenv("ALSA_MIXER"))) { device = "default"; } } if (!(channel = AudioMixerChannel)) { if (!(channel = getenv("ALSA_MIXER_CHANNEL"))) { channel = "PCM"; } } Debug(3, "audio/alsa: mixer %s - %s open\n", device, channel); snd_mixer_open(&alsa_mixer, 0); if (alsa_mixer && snd_mixer_attach(alsa_mixer, device) >= 0 && snd_mixer_selem_register(alsa_mixer, NULL, NULL) >= 0 && snd_mixer_load(alsa_mixer) >= 0) { const char *const alsa_mixer_elem_name = channel; alsa_mixer_elem = snd_mixer_first_elem(alsa_mixer); while (alsa_mixer_elem) { const char *name; name = snd_mixer_selem_get_name(alsa_mixer_elem); if (!strcasecmp(name, alsa_mixer_elem_name)) { snd_mixer_selem_get_playback_volume_range(alsa_mixer_elem, &alsa_mixer_elem_min, &alsa_mixer_elem_max); AlsaRatio = 1000 * (alsa_mixer_elem_max - alsa_mixer_elem_min); Debug(3, "audio/alsa: PCM mixer found %ld - %ld ratio %d\n", alsa_mixer_elem_min, alsa_mixer_elem_max, AlsaRatio); break; } alsa_mixer_elem = snd_mixer_elem_next(alsa_mixer_elem); } AlsaMixer = alsa_mixer; AlsaMixerElem = alsa_mixer_elem; } else { Error(_("audio/alsa: can't open mixer '%s'\n"), device); } } //---------------------------------------------------------------------------- // Alsa API //---------------------------------------------------------------------------- #ifdef USE_AUDIORING /** ** Get alsa audio delay in time-stamps. ** ** @returns audio delay in time-stamps. ** ** @todo FIXME: handle the case no audio running */ static int64_t AlsaGetDelay(void) { int err; snd_pcm_sframes_t delay; int64_t pts; // setup error if (!AlsaPCMHandle || !AudioRing[AudioRingRead].HwSampleRate) { return 0L; } // delay in frames in alsa + kernel buffers if ((err = snd_pcm_delay(AlsaPCMHandle, &delay)) < 0) { //Debug(3, "audio/alsa: no hw delay\n"); delay = 0L; #ifdef DEBUG } else if (snd_pcm_state(AlsaPCMHandle) != SND_PCM_STATE_RUNNING) { //Debug(3, "audio/alsa: %ld frames delay ok, but not running\n", delay); #endif } //Debug(3, "audio/alsa: %ld frames hw delay\n", delay); // delay can be negative, when underrun occur if (delay < 0) { delay = 0L; } pts = ((int64_t) delay * 90 * 1000) / AudioRing[AudioRingRead].HwSampleRate; return pts; } /** ** Setup alsa audio for requested format. ** ** @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 ** @retval -1 something gone wrong ** ** @todo FIXME: remove pointer for freq + channels */ static int AlsaSetup(int *freq, int *channels, int use_ac3) { snd_pcm_uframes_t buffer_size; snd_pcm_uframes_t period_size; int err; int delay; if (!AlsaPCMHandle) { // alsa not running yet // FIXME: if open fails for ac3, we never recover return -1; } if (1) { // close+open to fix HDMI no sound bug snd_pcm_t *handle; handle = AlsaPCMHandle; // FIXME: need lock AlsaPCMHandle = NULL; // other threads should check handle snd_pcm_close(handle); if (!(handle = AlsaOpenPCM(use_ac3))) { return -1; } AlsaPCMHandle = handle; } for (;;) { if ((err = snd_pcm_set_params(AlsaPCMHandle, SND_PCM_FORMAT_S16, AlsaUseMmap ? SND_PCM_ACCESS_MMAP_INTERLEAVED : SND_PCM_ACCESS_RW_INTERLEAVED, *channels, *freq, 1, 96 * 1000))) { /* if ( err == -EBADFD ) { snd_pcm_close(AlsaPCMHandle); AlsaPCMHandle = NULL; continue; } */ if (!AudioDoingInit) { Error(_("audio/alsa: set params error: %s\n"), snd_strerror(err)); } // FIXME: must stop sound, AudioChannels ... invalid return -1; } break; } // this is disabled, no advantages! if (0) { // no underruns allowed, play silence snd_pcm_sw_params_t *sw_params; snd_pcm_uframes_t boundary; snd_pcm_sw_params_alloca(&sw_params); err = snd_pcm_sw_params_current(AlsaPCMHandle, sw_params); if (err < 0) { Error(_("audio: snd_pcm_sw_params_current failed: %s\n"), snd_strerror(err)); } if ((err = snd_pcm_sw_params_get_boundary(sw_params, &boundary)) < 0) { Error(_("audio: snd_pcm_sw_params_get_boundary failed: %s\n"), snd_strerror(err)); } Debug(4, "audio/alsa: boundary %lu frames\n", boundary); if ((err = snd_pcm_sw_params_set_stop_threshold(AlsaPCMHandle, sw_params, boundary)) < 0) { Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"), snd_strerror(err)); } if ((err = snd_pcm_sw_params_set_silence_size(AlsaPCMHandle, sw_params, boundary)) < 0) { Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"), snd_strerror(err)); } if ((err = snd_pcm_sw_params(AlsaPCMHandle, sw_params)) < 0) { Error(_("audio: snd_pcm_sw_params failed: %s\n"), snd_strerror(err)); } } // update buffer snd_pcm_get_params(AlsaPCMHandle, &buffer_size, &period_size); Debug(3, "audio/alsa: buffer size %lu %zdms, period size %lu %zdms\n", buffer_size, snd_pcm_frames_to_bytes(AlsaPCMHandle, buffer_size) * 1000 / (*freq * *channels * AudioBytesProSample), period_size, snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size) * 1000 / (*freq * *channels * AudioBytesProSample)); Debug(3, "audio/alsa: state %s\n", snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle))); AudioStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size); // buffer time/delay in ms delay = AudioBufferTime; if (VideoAudioDelay > 0) { delay += VideoAudioDelay / 90; } if (AudioStartThreshold < (*freq * *channels * AudioBytesProSample * delay) / 1000U) { AudioStartThreshold = (*freq * *channels * AudioBytesProSample * delay) / 1000U; } // no bigger, than 1/3 the buffer if (AudioStartThreshold > AudioRingBufferSize / 3) { AudioStartThreshold = AudioRingBufferSize / 3; } if (!AudioDoingInit) { Info(_("audio/alsa: start delay %ums\n"), (AudioStartThreshold * 1000) / (*freq * *channels * AudioBytesProSample)); } return 0; } #else /** ** Get alsa audio delay in time stamps. ** ** @returns audio delay in time stamps. ** ** @todo FIXME: handle the case no audio running */ static int64_t AlsaGetDelay(void) { int err; snd_pcm_sframes_t delay; int64_t pts; if (!AlsaPCMHandle || !AudioSampleRate) { return 0L; } if (!AudioRunning) { // audio not running return 0L; } // FIXME: thread safe? __assert_fail_base in snd_pcm_delay // delay in frames in alsa + kernel buffers if ((err = snd_pcm_delay(AlsaPCMHandle, &delay)) < 0) { //Debug(3, "audio/alsa: no hw delay\n"); delay = 0L; } else if (snd_pcm_state(AlsaPCMHandle) != SND_PCM_STATE_RUNNING) { //Debug(3, "audio/alsa: %ld frames delay ok, but not running\n", delay); } //Debug(3, "audio/alsa: %ld frames hw delay\n", delay); // delay can be negative when underrun occur if (delay < 0) { delay = 0L; } pts = ((int64_t) delay * 90 * 1000) / AudioSampleRate; pts += ((int64_t) RingBufferUsedBytes(AlsaRingBuffer) * 90 * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample); Debug(4, "audio/alsa: hw+sw delay %zd %" PRId64 "ms\n", RingBufferUsedBytes(AlsaRingBuffer), pts / 90); return pts; } /** ** Setup alsa audio for requested format. ** ** @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 ** @retval -1 something gone wrong ** ** @todo audio changes must be queued and done when the buffer is empty */ static int AlsaSetup(int *freq, int *channels, int use_ac3) { snd_pcm_uframes_t buffer_size; snd_pcm_uframes_t period_size; int err; int ret; int delay; snd_pcm_t *handle; if (!AlsaPCMHandle) { // alsa not running yet return -1; } #if 1 // easy alsa hw setup way // flush any buffered data AudioFlushBuffers(); Debug(3, "audio: %dms flush\n", (AudioUsedBytes() * 1000) / (!AudioSampleRate + !AudioChannels + AudioSampleRate * AudioChannels * AudioBytesProSample)); if (1) { // close+open to fix hdmi no sound bugs handle = AlsaPCMHandle; AlsaPCMHandle = NULL; snd_pcm_close(handle); if (!(handle = AlsaOpenPCM(use_ac3))) { return -1; } AlsaPCMHandle = handle; } ret = 0; try_again: AudioChannels = *channels; AudioSampleRate = *freq; if ((err = snd_pcm_set_params(AlsaPCMHandle, SND_PCM_FORMAT_S16, AlsaUseMmap ? SND_PCM_ACCESS_MMAP_INTERLEAVED : SND_PCM_ACCESS_RW_INTERLEAVED, *channels, *freq, 1, 96 * 1000))) { Error(_("audio/alsa: set params error: %s\n"), snd_strerror(err)); /* if ( err == -EBADFD ) { snd_pcm_close(AlsaPCMHandle); AlsaPCMHandle = NULL; goto try_again; } */ switch (*channels) { case 1: // FIXME: enable channel upmix ret = 1; *channels = 2; goto try_again; case 2: return -1; case 3: case 4: case 5: case 6: case 7: case 8: // FIXME: enable channel downmix // FIXME: try 8 -> 7 -> 6 -> 5 -> 4 -> 3 -> 2 ret = 1; *channels = 2; goto try_again; default: Error(_("audio/alsa: unsupported number of channels\n")); // FIXME: must stop sound, AudioChannels ... invalid return -1; } } #else // // complex way to setup parameters // snd_pcm_hw_params_t *hw_params; int dir; unsigned buffer_time; snd_pcm_uframes_t buffer_size; snd_pcm_hw_params_alloca(&hw_params); // choose all parameters if ((err = snd_pcm_hw_params_any(AlsaPCMHandle, hw_params)) < 0) { Error(_ ("audio: snd_pcm_hw_params_any: no configurations available: %s\n"), snd_strerror(err)); } if ((err = snd_pcm_hw_params_set_rate_resample(AlsaPCMHandle, hw_params, 1)) < 0) { Error(_("audio: can't set rate resample: %s\n"), snd_strerror(err)); } if ((err = snd_pcm_hw_params_set_format(AlsaPCMHandle, hw_params, SND_PCM_FORMAT_S16)) < 0) { Error(_("audio: can't set 16-bit: %s\n"), snd_strerror(err)); } if ((err = snd_pcm_hw_params_set_access(AlsaPCMHandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { Error(_("audio: can't set interleaved read/write %s\n"), snd_strerror(err)); } if ((err = snd_pcm_hw_params_set_channels(AlsaPCMHandle, hw_params, channels)) < 0) { Error(_("audio: can't set channels: %s\n"), snd_strerror(err)); } if ((err = snd_pcm_hw_params_set_rate(AlsaPCMHandle, hw_params, freq, 0)) < 0) { Error(_("audio: can't set rate: %s\n"), snd_strerror(err)); } // 500000 // 170667us buffer_time = 1000 * 1000 * 1000; dir = 1; #if 0 snd_pcm_hw_params_get_buffer_time_max(hw_params, &buffer_time, &dir); Info(_("audio/alsa: %dus max buffer time\n"), buffer_time); buffer_time = 5 * 200 * 1000; // 1s if ((err = snd_pcm_hw_params_set_buffer_time_near(AlsaPCMHandle, hw_params, &buffer_time, &dir)) < 0) { Error(_("audio: snd_pcm_hw_params_set_buffer_time_near failed: %s\n"), snd_strerror(err)); } Info(_("audio/alsa: %dus buffer time\n"), buffer_time); #endif snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size); Info(_("audio/alsa: buffer size %lu\n"), buffer_size); buffer_size = buffer_size < 65536 ? buffer_size : 65536; if ((err = snd_pcm_hw_params_set_buffer_size_near(AlsaPCMHandle, hw_params, &buffer_size))) { Error(_("audio: can't set buffer size: %s\n"), snd_strerror(err)); } Info(_("audio/alsa: buffer size %lu\n"), buffer_size); if ((err = snd_pcm_hw_params(AlsaPCMHandle, hw_params)) < 0) { Error(_("audio: snd_pcm_hw_params failed: %s\n"), snd_strerror(err)); } // FIXME: use hw_params for buffer_size period_size #endif #if 1 if (0) { // no underruns allowed, play silence snd_pcm_sw_params_t *sw_params; snd_pcm_uframes_t boundary; snd_pcm_sw_params_alloca(&sw_params); err = snd_pcm_sw_params_current(AlsaPCMHandle, sw_params); if (err < 0) { Error(_("audio: snd_pcm_sw_params_current failed: %s\n"), snd_strerror(err)); } if ((err = snd_pcm_sw_params_get_boundary(sw_params, &boundary)) < 0) { Error(_("audio: snd_pcm_sw_params_get_boundary failed: %s\n"), snd_strerror(err)); } Debug(4, "audio/alsa: boundary %lu frames\n", boundary); if ((err = snd_pcm_sw_params_set_stop_threshold(AlsaPCMHandle, sw_params, boundary)) < 0) { Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"), snd_strerror(err)); } if ((err = snd_pcm_sw_params_set_silence_size(AlsaPCMHandle, sw_params, boundary)) < 0) { Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"), snd_strerror(err)); } if ((err = snd_pcm_sw_params(AlsaPCMHandle, sw_params)) < 0) { Error(_("audio: snd_pcm_sw_params failed: %s\n"), snd_strerror(err)); } } #endif // update buffer snd_pcm_get_params(AlsaPCMHandle, &buffer_size, &period_size); Debug(3, "audio/alsa: buffer size %lu %zdms, period size %lu %zdms\n", buffer_size, snd_pcm_frames_to_bytes(AlsaPCMHandle, buffer_size) * 1000 / (AudioSampleRate * AudioChannels * AudioBytesProSample), period_size, snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size) * 1000 / (AudioSampleRate * AudioChannels * AudioBytesProSample)); Debug(3, "audio/alsa: state %s\n", snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle))); AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size); // buffer time/delay in ms delay = AudioBufferTime; if (VideoAudioDelay > 0) { delay += VideoAudioDelay / 90; } if (AlsaStartThreshold < (*freq * *channels * AudioBytesProSample * delay) / 1000U) { AlsaStartThreshold = (*freq * *channels * AudioBytesProSample * delay) / 1000U; } // no bigger, than the buffer if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) { AlsaStartThreshold = RingBufferFreeBytes(AlsaRingBuffer); } Info(_("audio/alsa: delay %ums\n"), (AlsaStartThreshold * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample)); return ret; } #endif /** ** Play audio. */ void AlsaPlay(void) { int err; if (AlsaCanPause) { if ((err = snd_pcm_pause(AlsaPCMHandle, 0))) { Error(_("audio/alsa: snd_pcm_pause(): %s\n"), snd_strerror(err)); } } else { if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) { Error(_("audio/alsa: snd_pcm_prepare(): %s\n"), snd_strerror(err)); } } #ifdef DEBUG if (snd_pcm_state(AlsaPCMHandle) == SND_PCM_STATE_PAUSED) { Error(_("audio/alsa: still paused\n")); } #endif } /** ** Pause audio. */ void AlsaPause(void) { int err; if (AlsaCanPause) { if ((err = snd_pcm_pause(AlsaPCMHandle, 1))) { Error(_("snd_pcm_pause(): %s\n"), snd_strerror(err)); } } else { if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) { Error(_("snd_pcm_drop(): %s\n"), snd_strerror(err)); } } } /** ** Empty log callback */ static void AlsaNoopCallback( __attribute__ ((unused)) const char *file, __attribute__ ((unused)) int line, __attribute__ ((unused)) const char *function, __attribute__ ((unused)) int err, __attribute__ ((unused)) const char *fmt, ...) { } /** ** Initialize alsa audio output module. */ static void AlsaInit(void) { #ifndef DEBUG // disable display alsa error messages snd_lib_error_set_handler(AlsaNoopCallback); #else (void)AlsaNoopCallback; #endif #ifndef USE_AUDIORING AlsaRingBuffer = RingBufferNew(AudioRingBufferSize); #endif AlsaInitPCM(); AlsaInitMixer(); } /** ** Cleanup alsa audio output module. */ static void AlsaExit(void) { if (AlsaPCMHandle) { snd_pcm_close(AlsaPCMHandle); AlsaPCMHandle = NULL; } if (AlsaMixer) { snd_mixer_close(AlsaMixer); AlsaMixer = NULL; AlsaMixerElem = NULL; } #ifndef USE_AUDIORING if (AlsaRingBuffer) { RingBufferDel(AlsaRingBuffer); AlsaRingBuffer = NULL; } AlsaFlushBuffer = 0; #endif } /** ** Alsa module. */ static const AudioModule AlsaModule = { .Name = "alsa", #ifdef USE_AUDIO_THREAD .Thread = AlsaThread, #ifdef USE_AUDIORING //.Enqueue = AlsaThreadEnqueue, //.VideoReady = AlsaVideoReady, .FlushBuffers = AlsaFlushBuffers, #else .Enqueue = AlsaThreadEnqueue, .VideoReady = AlsaVideoReady, .FlushBuffers = AlsaThreadFlushBuffers, #endif #else .Enqueue = AlsaEnqueue, .VideoReady = AlsaVideoReady, .FlushBuffers = AlsaFlushBuffers, #endif #ifndef USE_AUDIORING .Poller = AlsaPoller, .FreeBytes = AlsaFreeBytes, .UsedBytes = AlsaUsedBytes, #endif .GetDelay = AlsaGetDelay, .SetVolume = AlsaSetVolume, .Setup = AlsaSetup, .Play = AlsaPlay, .Pause = AlsaPause, .Init = AlsaInit, .Exit = AlsaExit, }; #endif // USE_ALSA #ifdef USE_OSS //============================================================================ // O S S //============================================================================ //---------------------------------------------------------------------------- // OSS variables //---------------------------------------------------------------------------- static int OssPcmFildes = -1; ///< pcm file descriptor static int OssMixerFildes = -1; ///< mixer file descriptor static int OssMixerChannel; ///< mixer channel index 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 //---------------------------------------------------------------------------- /** ** Place samples in ringbuffer. ** ** @param samples sample buffer ** @param count number of bytes in sample buffer ** ** @returns true if play should be started. */ static int OssAddToRingbuffer(const void *samples, int count) { int n; n = RingBufferWrite(OssRingBuffer, samples, count); if (n != count) { Error(_("audio/oss: can't place %d samples in ring buffer\n"), count); // too many bytes are lost // FIXME: should skip more, longer skip, but less often? } if (!AudioRunning) { Debug(4, "audio/oss: start %4zdms\n", (RingBufferUsedBytes(OssRingBuffer) * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample)); // forced start if (OssStartThreshold * 2 < RingBufferUsedBytes(OssRingBuffer)) { return 1; } // enough video + audio buffered if (AudioVideoIsReady && OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) { // restart play-back return 1; } } return 0; } /** ** Play samples from ringbuffer. */ static int OssPlayRingbuffer(void) { int first; const void *p; first = 1; for (;;) { audio_buf_info bi; 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(OssRingBuffer, &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! } n = write(OssPcmFildes, p, bi.bytes); if (n != bi.bytes) { if (n < 0) { Error(_("audio/oss: write error: %s\n"), strerror(errno)); return 1; } Warning(_("audio/oss: error not all bytes written\n")); } // advance how many could written RingBufferReadAdvance(OssRingBuffer, n); first = 0; } return 0; } /** ** Flush OSS buffers. */ static void OssFlushBuffers(void) { if (OssRingBuffer && OssPcmFildes != -1) { RingBufferReadAdvance(OssRingBuffer, RingBufferUsedBytes(OssRingBuffer)); // flush kernel buffers if (ioctl(OssPcmFildes, SNDCTL_DSP_HALT_OUTPUT, NULL) < 0) { Error(_("audio/oss: ioctl(SNDCTL_DSP_HALT_OUTPUT): %s\n"), strerror(errno)); } } AudioRunning = 0; AudioVideoIsReady = 0; AudioPTS = INT64_C(0x8000000000000000); } //---------------------------------------------------------------------------- // OSS pcm polled //---------------------------------------------------------------------------- #ifndef USE_AUDIO_THREAD /** ** Place samples in audio output queue. ** ** @param samples sample buffer ** @param count number of bytes in sample buffer */ static void OssEnqueue(const void *samples, int count) { #ifdef DEBUG static uint32_t last_tick; uint32_t tick; tick = GetMsTicks(); Debug(4, "audio/oss: %4d %dms\n", count, tick - last_tick); last_tick = tick; #endif if (OssPcmFildes == -1) { // setup failure Debug(3, "audio/oss: not ready\n"); return; } if (OssAddToRingbuffer(samples, count)) { AudioRunning = 1; } } #endif /** ** Play all samples possible, without blocking. */ static void OssPoller(void) { if (OssPcmFildes == -1) { // setup failure return; } if (!AudioThread && AudioRunning) { OssPlayRingbuffer(); } } /** ** Get free bytes in audio output. */ static int OssFreeBytes(void) { return OssRingBuffer ? RingBufferFreeBytes(OssRingBuffer) : INT32_MAX; } /** ** Get used bytes in audio output. */ 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 */ static int OssThread(void) { for (;;) { struct pollfd fds[1]; int err; pthread_testcancel(); if (OssFlushBuffer) { // we can flush too many, but wo cares Debug(3, "audio/oss: flushing buffers\n"); OssFlushBuffers(); OssFlushBuffer = 0; break; } if (AudioPaused) { break; } fds[0].fd = OssPcmFildes; fds[0].events = POLLOUT | POLLERR; // wait for space in kernel buffers err = poll(fds, 1, OssFragmentTime); if (err < 0) { Error(_("audio/oss: error poll %s\n"), strerror(errno)); usleep(OssFragmentTime * 1000); continue; } if (OssFlushBuffer || AudioPaused) { continue; } if ((err = OssPlayRingbuffer())) { // empty / error if (err < 0) { // underrun error break; } pthread_yield(); usleep(OssFragmentTime * 1000); // let fill/empty the buffers } } return 0; } /** ** Place samples in audio output queue. ** ** @param samples sample buffer ** @param count number of bytes in sample buffer */ static void OssThreadEnqueue(const void *samples, int count) { if (!OssRingBuffer || OssPcmFildes == -1) { Debug(3, "audio/oss: enqueue not ready\n"); return; } if (OssAddToRingbuffer(samples, count)) { // no lock needed, can wakeup next time AudioRunning = 1; pthread_cond_signal(&AudioStartCond); } } /** ** Video is ready, start audio if possible, */ static void OssVideoReady(void) { if (AudioSampleRate && AudioChannels) { Debug(3, "audio/oss: start %4zdms video start\n", (RingBufferUsedBytes(OssRingBuffer) * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample)); } if (!AudioRunning) { // enough video + audio buffered if (OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) { AudioRunning = 1; pthread_cond_signal(&AudioStartCond); } } } /** ** Flush OSS buffers with thread. */ static void OssThreadFlushBuffers(void) { // signal thread to flush buffers if (AudioThread) { OssFlushBuffer = 1; do { AudioRunning = 1; // wakeup in case of sleeping pthread_cond_signal(&AudioStartCond); usleep(1 * 1000); } while (OssFlushBuffer); // wait until flushed } } #endif #endif //---------------------------------------------------------------------------- /** ** Open OSS pcm device. ** ** @param use_ac3 use ac3/pass-through device */ static int OssOpenPCM(int use_ac3) { const char *device; int fildes; // &&|| hell if (!(use_ac3 && ((device = AudioAC3Device) || (device = getenv("OSS_AC3_AUDIODEV")))) && !(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) { device = "/dev/dsp"; } 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, strerror(errno)); return -1; } return fildes; } /** ** Initialize OSS pcm device. ** ** @see AudioPCMDevice */ static void OssInitPCM(void) { int fildes; fildes = OssOpenPCM(0); OssPcmFildes = fildes; } //---------------------------------------------------------------------------- // OSS Mixer //---------------------------------------------------------------------------- /** ** Set OSS mixer volume (0-1000) ** ** @param volume volume (0 .. 1000) */ static void OssSetVolume(int volume) { int v; if (OssMixerFildes != -1) { v = (volume * 255) / 1000; v &= 0xff; v = (v << 8) | v; if (ioctl(OssMixerFildes, MIXER_WRITE(OssMixerChannel), &v) < 0) { Error(_("audio/oss: ioctl(MIXER_WRITE): %s\n"), strerror(errno)); } } } /** ** Mixer channel name table. */ static const char *OssMixerChannelNames[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES; /** ** Initialize OSS mixer. */ static void OssInitMixer(void) { const char *device; const char *channel; int fildes; int devmask; int i; if (!(device = AudioMixerDevice)) { if (!(device = getenv("OSS_MIXERDEV"))) { device = "/dev/mixer"; } } if (!(channel = AudioMixerChannel)) { if (!(channel = getenv("OSS_MIXER_CHANNEL"))) { channel = "pcm"; } } Debug(3, "audio/oss: mixer %s - %s open\n", device, channel); if ((fildes = open(device, O_RDWR)) < 0) { Error(_("audio/oss: can't open mixer device '%s': %s\n"), device, strerror(errno)); return; } // search channel name if (ioctl(fildes, SOUND_MIXER_READ_DEVMASK, &devmask) < 0) { Error(_("audio/oss: ioctl(SOUND_MIXER_READ_DEVMASK): %s\n"), strerror(errno)); close(fildes); return; } for (i = 0; i < SOUND_MIXER_NRDEVICES; ++i) { if (!strcasecmp(OssMixerChannelNames[i], channel)) { if (devmask & (1 << i)) { OssMixerFildes = fildes; OssMixerChannel = i; return; } Error(_("audio/oss: channel '%s' not supported\n"), channel); break; } } Error(_("audio/oss: channel '%s' not found\n"), channel); close(fildes); } //---------------------------------------------------------------------------- // 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. ** ** @returns audio delay in time stamps. */ static int64_t OssGetDelay(void) { int delay; int64_t pts; if (OssPcmFildes == -1) { // setup failure return 0L; } if (!AudioRunning) { // audio not running 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 0UL; } if (delay < 0) { delay = 0; } pts = ((int64_t) (delay + RingBufferUsedBytes(OssRingBuffer)) * 90 * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample); Debug(4, "audio/oss: hw+sw delay %zd %" PRId64 "ms\n", RingBufferUsedBytes(OssRingBuffer), pts / 90); return pts; } /** ** Setup OSS audio for requested format. ** ** @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 ** @retval -1 something gone wrong ** ** @todo audio changes must be queued and done when the buffer is empty */ static int OssSetup(int *freq, int *channels, int use_ac3) { int ret; int tmp; int delay; audio_buf_info bi; if (OssPcmFildes == -1) { // OSS not ready return -1; } // flush any buffered data AudioFlushBuffers(); 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 = *freq; if (ioctl(OssPcmFildes, SNDCTL_DSP_SPEED, &tmp) == -1) { Error(_("audio/oss: ioctl(SNDCTL_DSP_SPEED): %s\n"), strerror(errno)); return -1; } if (tmp != *freq) { Warning(_("audio/oss: device doesn't support %dHz sample rate.\n"), *freq); *freq = tmp; ret = 1; } AudioChannels = *channels; AudioSampleRate = *freq; // FIXME: setup buffers #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) / (AudioSampleRate * AudioChannels * AudioBytesProSample); Debug(3, "audio/oss: buffer size %d %dms, fragment size %d %dms\n", bi.fragsize * bi.fragstotal, (bi.fragsize * bi.fragstotal * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample), bi.fragsize, OssFragmentTime); // start when enough bytes for initial write OssStartThreshold = (bi.fragsize - 1) * bi.fragstotal; // buffer time/delay in ms delay = AudioBufferTime + 300; if (VideoAudioDelay > 0) { delay += VideoAudioDelay / 90; } if (OssStartThreshold < (AudioSampleRate * AudioChannels * AudioBytesProSample * delay) / 1000U) { OssStartThreshold = (AudioSampleRate * AudioChannels * AudioBytesProSample * delay) / 1000U; } // no bigger, than the buffer if (OssStartThreshold > RingBufferFreeBytes(OssRingBuffer)) { OssStartThreshold = RingBufferFreeBytes(OssRingBuffer); } Info(_("audio/oss: delay %ums\n"), (OssStartThreshold * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample)); return ret; } #endif /** ** Play audio. */ void OssPlay(void) { } /** ** Pause audio. */ void OssPause(void) { } /** ** Initialize OSS audio output module. */ static void OssInit(void) { #ifndef USE_AUDIORING OssRingBuffer = RingBufferNew(AudioRingBufferSize); #endif OssInitPCM(); OssInitMixer(); } /** ** Cleanup OSS audio output module. */ static void OssExit(void) { if (OssPcmFildes != -1) { close(OssPcmFildes); OssPcmFildes = -1; } if (OssMixerFildes != -1) { close(OssMixerFildes); OssMixerFildes = -1; } OssFlushBuffer = 0; } /** ** OSS module. */ 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, .Play = OssPlay, .Pause = OssPause, .Init = OssInit, .Exit = OssExit, }; #endif // USE_OSS //============================================================================ // Noop //============================================================================ #ifndef USE_AUDIORING /** ** Noop enqueue samples. ** ** @param samples sample buffer ** @param count number of bytes in sample buffer */ static void NoopEnqueue( __attribute__ ((unused)) const void *samples, __attribute__ ((unused)) int count) { } /** ** Get free bytes in audio output. */ static int NoopFreeBytes(void) { return INT32_MAX; // no driver, much space } /** ** Get used bytes in audio output. */ static int NoopUsedBytes(void) { return 0; // no driver, nothing used } #endif /** ** Get audio delay in time stamps. ** ** @returns audio delay in time stamps. */ static int64_t NoopGetDelay(void) { return 0L; } /** ** Set mixer volume (0-1000) ** ** @param volume volume (0 .. 1000) */ static void NoopSetVolume( __attribute__ ((unused)) int volume) { } /** ** Noop setup. ** ** @param freq sample frequency ** @param channels number of channels */ static int NoopSetup( __attribute__ ((unused)) int *channels, __attribute__ ((unused)) int *freq, __attribute__ ((unused)) int use_ac3) { return -1; } /** ** Noop void */ static void NoopVoid(void) { } /** ** Noop module. */ 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, .Play = NoopVoid, .Pause = NoopVoid, .Init = NoopVoid, .Exit = NoopVoid, }; //---------------------------------------------------------------------------- // thread playback //---------------------------------------------------------------------------- #ifdef USE_AUDIO_THREAD #ifdef USE_AUDIORING /** ** Prepare next ring buffer. */ static int AudioNextRing(void) { int use_ac3; int sample_rate; int channels; // update audio format // not always needed, but check if needed is too complex use_ac3 = AudioRing[AudioRingRead].UseAc3; sample_rate = AudioRing[AudioRingRead].HwSampleRate; channels = AudioRing[AudioRingRead].HwChannels; if (AudioUsedModule->Setup(&sample_rate, &channels, use_ac3)) { Error(_("audio: can't set channels %d sample-rate %dHz\n"), channels, sample_rate); // FIXME: handle error AudioRing[AudioRingRead].HwSampleRate = 0; AudioRing[AudioRingRead].InSampleRate = 0; return -1; } AudioSetVolume(AudioVolume); // update channel delta AudioResetCompressor(); AudioResetNormalizer(); // stop, if not enough in next buffer if (AudioStartThreshold >= RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer)) { return 1; } return 0; } /** ** Audio play thread. ** ** @param dummy unused thread argument */ static void *AudioPlayHandlerThread(void *dummy) { Debug(3, "audio: play thread started\n"); for (;;) { Debug(3, "audio: wait on start condition\n"); pthread_mutex_lock(&AudioMutex); AudioRunning = 0; do { pthread_cond_wait(&AudioStartCond, &AudioMutex); // cond_wait can return, without signal! } while (!AudioRunning); pthread_mutex_unlock(&AudioMutex); Debug(3, "audio: ----> %dms start\n", (AudioUsedBytes() * 1000) / (!AudioRing[AudioRingRead].HwSampleRate + !AudioRing[AudioRingRead].HwChannels + AudioRing[AudioRingRead].HwSampleRate * AudioRing[AudioRingRead].HwChannels * AudioBytesProSample)); do { int filled; int read; int flush; int err; // look if there is a flush command in the queue flush = 0; filled = atomic_read(&AudioRingFilled); read = AudioRingRead; while (filled--) { read = (read + 1) % AUDIO_RING_MAX; if (AudioRing[read].FlushBuffers) { AudioRing[read].FlushBuffers = 0; AudioRingRead = read; atomic_set(&AudioRingFilled, filled); // handle all flush in queue flush = 1; } } if (flush) { AudioUsedModule->FlushBuffers(); if (AudioNextRing()) { break; } } // try to play some samples err = AudioUsedModule->Thread(); // underrun, check if new ring buffer is available if (!err) { int use_ac3; int sample_rate; int channels; int old_use_ac3; int old_sample_rate; int old_channels; // underrun, and no new ring buffer, goto sleep. if (!atomic_read(&AudioRingFilled)) { break; } Debug(3, "audio: next ring buffer\n"); old_use_ac3 = AudioRing[AudioRingRead].UseAc3; old_sample_rate = AudioRing[AudioRingRead].HwSampleRate; old_channels = AudioRing[AudioRingRead].HwChannels; atomic_dec(&AudioRingFilled); AudioRingRead = (AudioRingRead + 1) % AUDIO_RING_MAX; use_ac3 = AudioRing[AudioRingRead].UseAc3; sample_rate = AudioRing[AudioRingRead].HwSampleRate; channels = AudioRing[AudioRingRead].HwChannels; Debug(3, "audio: thread channels %d frequency %dHz %s\n", channels, sample_rate, use_ac3 ? "ac3" : "pcm"); // audio config changed? if (old_use_ac3 != use_ac3 || old_sample_rate != sample_rate || old_channels != channels) { // FIXME: wait for buffer drain if (AudioNextRing()) { break; } } else { AudioResetCompressor(); AudioResetNormalizer(); } } } while (AudioRing[AudioRingRead].HwSampleRate); } return dummy; } #else /** ** Audio play thread. ** ** @param dummy unused thread argument */ static void *AudioPlayHandlerThread(void *dummy) { Debug(3, "audio: play thread started\n"); for (;;) { Debug(3, "audio: wait on start condition\n"); pthread_mutex_lock(&AudioMutex); AudioRunning = 0; do { pthread_cond_wait(&AudioStartCond, &AudioMutex); // cond_wait can return, without signal! } while (!AudioRunning); pthread_mutex_unlock(&AudioMutex); Debug(3, "audio: ----> %dms start\n", (AudioUsedBytes() * 1000) / (!AudioSampleRate + !AudioChannels + AudioSampleRate * AudioChannels * AudioBytesProSample)); AudioUsedModule->Thread(); } return dummy; } #endif /** ** Initialize audio thread. */ static void AudioInitThread(void) { pthread_mutex_init(&AudioMutex, NULL); pthread_cond_init(&AudioStartCond, NULL); pthread_create(&AudioThread, NULL, AudioPlayHandlerThread, NULL); pthread_setname_np(AudioThread, "softhddev audio"); #ifndef USE_AUDIORING pthread_yield(); usleep(5 * 1000); // give thread some time to start #endif } /** ** Cleanup audio thread. */ static void AudioExitThread(void) { void *retval; if (AudioThread) { if (pthread_cancel(AudioThread)) { Error(_("audio: can't queue cancel play thread\n")); } if (pthread_join(AudioThread, &retval) || retval != PTHREAD_CANCELED) { Error(_("audio: can't cancel play thread\n")); } pthread_cond_destroy(&AudioStartCond); pthread_mutex_destroy(&AudioMutex); AudioThread = 0; } } #endif //---------------------------------------------------------------------------- //---------------------------------------------------------------------------- /** ** Table of all audio modules. */ static const AudioModule *AudioModules[] = { #ifdef USE_ALSA &AlsaModule, #endif #ifdef USE_OSS &OssModule, #endif &NoopModule, }; /** ** Place samples in audio output queue. ** ** @param samples sample buffer ** @param count number of bytes in sample buffer */ void AudioEnqueue(const void *samples, int count) { #ifdef USE_AUDIORING size_t n; int16_t *buffer; int frames; #ifdef DEBUG static uint32_t last_tick; uint32_t tick; tick = GetMsTicks(); if (tick - last_tick > 101) { Debug(3, "audio: enqueue %4d %dms\n", count, tick - last_tick); } last_tick = tick; #endif if (!AudioRing[AudioRingWrite].HwSampleRate) { Debug(3, "audio: enqueue not ready\n"); return; // no setup yet } // // Convert / resample input to hardware format // frames = count / (AudioRing[AudioRingWrite].InChannels * AudioBytesProSample); buffer = alloca(frames * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample); AudioResample(samples, AudioRing[AudioRingWrite].InChannels, frames, buffer, AudioRing[AudioRingWrite].HwChannels); count = frames * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample; // resample into ring-buffer is too complex in the case of a roundabout // just use a temporary buffer if (AudioCompression) { // in place operation AudioCompressor(buffer, count); } if (AudioNormalize) { // in place operation AudioNormalizer(buffer, count); } n = RingBufferWrite(AudioRing[AudioRingWrite].RingBuffer, buffer, count); if (n != (size_t) count) { Error(_("audio: can't place %d samples in ring buffer\n"), count); // too many bytes are lost // FIXME: caller checks buffer full. // FIXME: should skip more, longer skip, but less often? // FIXME: round to channel + sample border } if (!AudioRunning) { // check, if we can start the thread //int64_t video_pts; //video_pts = VideoGetClock(); n = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer); Debug(3, "audio: start? %4zdms\n", (n * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample)); // forced start or enough video + audio buffered if (AudioStartThreshold * 2 < n || (AudioVideoIsReady && AudioStartThreshold < n)) { // restart play-back // no lock needed, can wakeup next time AudioRunning = 1; pthread_cond_signal(&AudioStartCond); } } // Update audio clock (stupid gcc developers thinks INT64_C is unsigned) if (AudioRing[AudioRingWrite].PTS != (int64_t) INT64_C(0x8000000000000000)) { AudioRing[AudioRingWrite].PTS += ((int64_t) count * 90 * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample); } #else if (!AudioSampleRate || !AudioChannels) { return; // not setup } if (0) { static uint32_t last; static uint32_t tick; static uint32_t max = 101; int64_t delay; delay = AudioGetDelay(); tick = GetMsTicks(); if ((last && tick - last > max) && AudioRunning) { //max = tick - last; Debug(3, "audio: packet delta %d %lu\n", tick - last, delay / 90); } last = tick; } AudioUsedModule->Enqueue(samples, count); // Update audio clock (stupid gcc developers thinks INT64_C is unsigned) if (AudioPTS != (int64_t) INT64_C(0x8000000000000000)) { AudioPTS += ((int64_t) count * 90 * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample); } #endif } /** ** Video is ready. ** ** @param pts video presentation timestamp */ void AudioVideoReady(int64_t pts) { #ifdef USE_AUDIORING if (AudioRing[AudioRingWrite].HwSampleRate && AudioRing[AudioRingWrite].HwChannels) { if (pts != (int64_t) INT64_C(0x8000000000000000) && AudioRing[AudioRingWrite].PTS != (int64_t) INT64_C(0x8000000000000000)) { Debug(3, "audio: a/v %d %s\n", (int)(pts - AudioRing[AudioRingWrite].PTS) / 90, AudioRunning ? "running" : "stopped"); } Debug(3, "audio: start %4zdms %s|%s video ready\n", (RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer) * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample), Timestamp2String(pts), Timestamp2String(AudioRing[AudioRingWrite].PTS)); if (!AudioRunning) { size_t used; used = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer); // enough video + audio buffered if (AudioStartThreshold < used) { // too much audio buffered, skip it if (AudioStartThreshold * 2 < used) { Debug(3, "audio: start %4zdms skip video ready\n", ((used - AudioStartThreshold * 2) * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample)); RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, used - AudioStartThreshold * 2); } AudioRunning = 1; pthread_cond_signal(&AudioStartCond); } } } AudioVideoIsReady = 1; #else (void)pts; AudioVideoIsReady = 1; AudioUsedModule->VideoReady(); #endif } /** ** Flush audio buffers. */ void AudioFlushBuffers(void) { #ifdef USE_AUDIORING int old; old = AudioRingWrite; AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX; AudioRing[AudioRingWrite].FlushBuffers = 1; AudioRing[AudioRingWrite].UseAc3 = AudioRing[old].UseAc3; AudioRing[AudioRingWrite].HwSampleRate = AudioRing[old].HwSampleRate; AudioRing[AudioRingWrite].HwChannels = AudioRing[old].HwChannels; AudioRing[AudioRingWrite].InSampleRate = AudioRing[old].InSampleRate; AudioRing[AudioRingWrite].InChannels = AudioRing[old].InChannels; AudioRing[AudioRingWrite].PTS = INT64_C(0x8000000000000000); RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer)); Debug(3, "audio: reset video ready\n"); AudioVideoIsReady = 0; atomic_inc(&AudioRingFilled); if (!AudioRunning) { // wakeup thread to flush buffers AudioRunning = 1; pthread_cond_signal(&AudioStartCond); } #else AudioUsedModule->FlushBuffers(); #endif } /** ** Call back to play audio polled. */ void AudioPoller(void) { #ifndef USE_AUDIORING AudioUsedModule->Poller(); #endif } /** ** Get free bytes in audio output. */ int AudioFreeBytes(void) { #ifdef USE_AUDIORING return AudioRing[AudioRingWrite]. RingBuffer ? RingBufferFreeBytes(AudioRing[AudioRingWrite]. RingBuffer) : INT32_MAX; #else return AudioUsedModule->FreeBytes(); #endif } /** ** Get used bytes in audio output. */ int AudioUsedBytes(void) { #ifdef USE_AUDIORING return AudioRing[AudioRingWrite]. RingBuffer ? RingBufferUsedBytes(AudioRing[AudioRingWrite]. RingBuffer) : 0; #else return AudioUsedModule->UsedBytes(); #endif } /** ** Get audio delay in time stamps. ** ** @returns audio delay in time stamps. */ int64_t AudioGetDelay(void) { #ifdef USE_AUDIORING int64_t pts; if (!AudioRunning) { return 0L; // audio not running } if (!AudioRing[AudioRingRead].HwSampleRate) { return 0L; // audio not setup } if (atomic_read(&AudioRingFilled)) { return 0L; // invalid delay } pts = AudioUsedModule->GetDelay(); pts += ((int64_t) RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer) * 90 * 1000) / (AudioRing[AudioRingRead].HwSampleRate * AudioRing[AudioRingRead].HwChannels * AudioBytesProSample); Debug(4, "audio/alsa: hw+sw delay %zd %" PRId64 "ms\n", RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer), pts / 90); return pts; #else return AudioUsedModule->GetDelay(); #endif } /** ** Set audio clock base. ** ** @param pts audio presentation timestamp */ void AudioSetClock(int64_t pts) { #ifdef USE_AUDIORING if (AudioRing[AudioRingWrite].PTS != pts) { Debug(4, "audio: set clock %s -> %s pts\n", Timestamp2String(AudioRing[AudioRingWrite].PTS), Timestamp2String(pts)); } AudioRing[AudioRingWrite].PTS = pts; #else #ifdef DEBUG if (AudioPTS != pts) { Debug(4, "audio: set clock %s -> %s pts\n", Timestamp2String(AudioPTS), Timestamp2String(pts)); } #endif AudioPTS = pts; #endif } /** ** Get current audio clock. ** ** @returns the audio clock in time stamps. */ int64_t AudioGetClock(void) { #ifdef USE_AUDIORING // (cast) needed for the evil gcc if (AudioRing[AudioRingRead].PTS != (int64_t) INT64_C(0x8000000000000000)) { int64_t delay; // delay zero, if no valid time stamp if ((delay = AudioGetDelay())) { return AudioRing[AudioRingRead].PTS - delay; } } return INT64_C(0x8000000000000000); #else // (cast) needed for the evil gcc if (AudioPTS != (int64_t) INT64_C(0x8000000000000000)) { int64_t delay; if ((delay = AudioGetDelay())) { return AudioPTS - delay; } } return INT64_C(0x8000000000000000); #endif } /** ** Set mixer volume (0-1000) ** ** @param volume volume (0 .. 1000) */ void AudioSetVolume(int volume) { AudioVolume = volume; #ifdef USE_AUDIORING // reduce loudness for stereo output if (AudioStereoDescent && AudioRing[AudioRingRead].InChannels == 2 && !AudioRing[AudioRingRead].UseAc3) { volume -= AudioStereoDescent; if (volume < 0) { volume = 0; } else if (volume > 1000) { volume = 1000; } } #endif AudioAmplifier = volume; printf("volume %d\n", volume); if (!AudioSoftVolume) { AudioUsedModule->SetVolume(volume); } } /** ** Setup audio for requested format. ** ** @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 ** @retval -1 something gone wrong */ int AudioSetup(int *freq, int *channels, int use_ac3) { Debug(3, "audio: setup channels %d frequency %dHz %s\n", *channels, *freq, use_ac3 ? "ac3" : "pcm"); // invalid parameter if (!freq || !channels || !*freq || !*channels) { Debug(3, "audio: bad channels or frequency parameters\n"); // FIXME: set flag invalid setup return -1; } #ifdef USE_AUDIORING return AudioRingAdd(*freq, *channels, use_ac3); #else return AudioUsedModule->Setup(freq, channels, use_ac3); #endif } /** ** Play audio. */ void AudioPlay(void) { if (!AudioPaused) { Debug(3, "audio: not paused, check the code\n"); return; } Debug(3, "audio: resumed\n"); AudioPaused = 0; AudioEnqueue(NULL, 0); // wakeup thread } /** ** Pause audio. */ void AudioPause(void) { if (AudioPaused) { Debug(3, "audio: already paused, check the code\n"); return; } Debug(3, "audio: paused\n"); AudioPaused = 1; } /** ** Set audio buffer time. ** ** PES audio packets have a max distance of 300 ms. ** TS audio packet have a max distance of 100 ms. ** The period size of the audio buffer is 24 ms. ** With streamdev sometimes extra +100ms are needed. */ void AudioSetBufferTime(int delay) { if (!delay) { delay = 336; } AudioBufferTime = delay; } /** ** Enable/disable software volume. ** ** @param onoff -1 toggle, true turn on, false turn off */ void AudioSetSoftvol(int onoff) { if (onoff < 0) { AudioSoftVolume ^= 1; } else { AudioSoftVolume = onoff; } } /** ** Set normalize volume parameters. ** ** @param onoff -1 toggle, true turn on, false turn off ** @param maxfac max. factor of normalize /1000 */ void AudioSetNormalize(int onoff, int maxfac) { if (onoff < 0) { AudioNormalize ^= 1; } else { AudioNormalize = onoff; } AudioMaxNormalize = maxfac; } /** ** Set volume compression parameters. ** ** @param onoff -1 toggle, true turn on, false turn off ** @param maxfac max. factor of compression /1000 */ void AudioSetCompression(int onoff, int maxfac) { if (onoff < 0) { AudioCompression ^= 1; } else { AudioCompression = onoff; } AudioMaxCompression = maxfac; if (!AudioCompressionFactor) { AudioCompressionFactor = 1000; } if (AudioCompressionFactor > AudioMaxCompression) { AudioCompressionFactor = AudioMaxCompression; } } /** ** Set stereo loudness descent. ** ** @param delta value (/1000) to reduce stereo volume */ void AudioSetStereoDescent(int delta) { AudioStereoDescent = delta; AudioSetVolume(AudioVolume); // update channel delta } /** ** Set pcm audio device. ** ** @param device name of pcm device (fe. "hw:0,9" or "/dev/dsp") ** ** @note this is currently used to select alsa/OSS output module. */ void AudioSetDevice(const char *device) { if (!AudioModuleName) { AudioModuleName = "alsa"; // detect alsa/OSS if (!device[0]) { AudioModuleName = "noop"; } else if (device[0] == '/') { AudioModuleName = "oss"; } } AudioPCMDevice = device; } /** ** Set pass-through audio device. ** ** @param device name of pass-through device (fe. "hw:0,1") ** ** @note this is currently usable with alsa only. */ void AudioSetDeviceAC3(const char *device) { if (!AudioModuleName) { AudioModuleName = "alsa"; // detect alsa/OSS if (!device[0]) { AudioModuleName = "noop"; } else if (device[0] == '/') { AudioModuleName = "oss"; } } AudioAC3Device = device; } /** ** Set pcm audio mixer channel. ** ** @param channel name of the mixer channel (fe. PCM or Master) ** ** @note this is currently used to select alsa/OSS output module. */ void AudioSetChannel(const char *channel) { AudioMixerChannel = channel; } /** ** Initialize audio output module. ** ** @todo FIXME: make audio output module selectable. */ void AudioInit(void) { unsigned u; const char *name; int freq; int chan; name = "noop"; #ifdef USE_OSS name = "oss"; #endif #ifdef USE_ALSA name = "alsa"; #endif if (AudioModuleName) { name = AudioModuleName; } // // search selected audio module. // for (u = 0; u < sizeof(AudioModules) / sizeof(*AudioModules); ++u) { if (!strcasecmp(name, AudioModules[u]->Name)) { AudioUsedModule = AudioModules[u]; Info(_("audio: '%s' output module used\n"), AudioUsedModule->Name); goto found; } } Error(_("audio: '%s' output module isn't supported\n"), name); AudioUsedModule = &NoopModule; return; found: AudioDoingInit = 1; #ifdef USE_AUDIORING AudioRingInit(); AudioUsedModule->Init(); // // Check which channels/rates/formats are supported // FIXME: we force 44.1Khz and 48Khz must be supported equal // FIXME: should use bitmap of channels supported in RatesInHw freq = 44100; AudioRatesInHw[Audio44100] = 0; for (chan = 1; chan < 9; ++chan) { if (AudioUsedModule->Setup(&freq, &chan, 0)) { AudioChannelsInHw[chan] = 0; } else { AudioChannelsInHw[chan] = chan; AudioRatesInHw[Audio44100] |= (1 << chan); } } freq = 48000; AudioRatesInHw[Audio48000] = 0; for (chan = 1; chan < 9; ++chan) { if (!AudioChannelsInHw[chan]) { continue; } if (AudioUsedModule->Setup(&freq, &chan, 0)) { AudioChannelsInHw[chan] = 0; } else { AudioChannelsInHw[chan] = chan; AudioRatesInHw[Audio48000] |= (1 << chan); } } // build channel support and conversion table for (u = 0; u < AudioRatesMax; ++u) { for (chan = 1; chan < 9; ++chan) { AudioChannelMatrix[u][chan] = 0; if (!AudioRatesInHw[u]) { // rate unsupported continue; } if (AudioChannelsInHw[chan]) { AudioChannelMatrix[u][chan] = chan; } else { switch (chan) { case 1: if (AudioChannelsInHw[2]) { AudioChannelMatrix[u][chan] = 2; } break; case 2: case 3: if (AudioChannelsInHw[4]) { AudioChannelMatrix[u][chan] = 4; break; } case 4: if (AudioChannelsInHw[5]) { AudioChannelMatrix[u][chan] = 5; break; } case 5: if (AudioChannelsInHw[6]) { AudioChannelMatrix[u][chan] = 6; break; } case 6: if (AudioChannelsInHw[7]) { AudioChannelMatrix[u][chan] = 7; break; } case 7: if (AudioChannelsInHw[8]) { AudioChannelMatrix[u][chan] = 8; break; } case 8: if (AudioChannelsInHw[6]) { AudioChannelMatrix[u][chan] = 6; break; } if (AudioChannelsInHw[2]) { AudioChannelMatrix[u][chan] = 2; break; } if (AudioChannelsInHw[1]) { AudioChannelMatrix[u][chan] = 1; break; } break; } } } } for (u = 0; u < AudioRatesMax; ++u) { Info(_("audio: %6dHz supports %d %d %d %d %d %d %d %d channels\n"), AudioRatesTable[u], AudioChannelMatrix[u][1], AudioChannelMatrix[u][2], AudioChannelMatrix[u][3], AudioChannelMatrix[u][4], AudioChannelMatrix[u][5], AudioChannelMatrix[u][6], AudioChannelMatrix[u][7], AudioChannelMatrix[u][8]); } #else AudioUsedModule->Init(); freq = 48000; chan = 2; if (AudioSetup(&freq, &chan, 0)) { // set default parameters Error(_("audio: can't do initial setup\n")); } #endif #ifdef USE_AUDIO_THREAD if (AudioUsedModule->Thread) { // supports threads AudioInitThread(); } #endif AudioDoingInit = 0; } /** ** Cleanup audio output module. */ void AudioExit(void) { #ifdef USE_AUDIO_THREAD if (AudioUsedModule->Thread) { // supports threads AudioExitThread(); } #endif AudioUsedModule->Exit(); AudioUsedModule = &NoopModule; #ifdef USE_AUDIORING AudioRingExit(); #endif AudioRunning = 0; AudioPaused = 0; } #ifdef AUDIO_TEST //---------------------------------------------------------------------------- // Test //---------------------------------------------------------------------------- void AudioTest(void) { for (;;) { unsigned u; uint8_t buffer[16 * 1024]; // some random data int i; for (u = 0; u < sizeof(buffer); u++) { buffer[u] = random() & 0xffff; } Debug(3, "audio/test: loop\n"); for (i = 0; i < 100; ++i) { while (RingBufferFreeBytes(AlsaRingBuffer) > sizeof(buffer)) { AlsaEnqueue(buffer, sizeof(buffer)); } usleep(20 * 1000); } break; } } #include int SysLogLevel; ///< show additional debug informations /** ** Print version. */ static void PrintVersion(void) { printf("audio_test: audio tester Version " VERSION #ifdef GIT_REV "(GIT-" GIT_REV ")" #endif ",\n\t(c) 2009 - 2012 by Johns\n" "\tLicense AGPLv3: GNU Affero General Public License version 3\n"); } /** ** Print usage. */ static void PrintUsage(void) { printf("Usage: audio_test [-?dhv]\n" "\t-d\tenable debug, more -d increase the verbosity\n" "\t-? -h\tdisplay this message\n" "\t-v\tdisplay version information\n" "Only idiots print usage on stderr!\n"); } /** ** Main entry point. ** ** @param argc number of arguments ** @param argv arguments vector ** ** @returns -1 on failures, 0 clean exit. */ int main(int argc, char *const argv[]) { SysLogLevel = 0; // // Parse command line arguments // for (;;) { switch (getopt(argc, argv, "hv?-c:d")) { case 'd': // enabled debug ++SysLogLevel; continue; case EOF: break; case 'v': // print version PrintVersion(); return 0; case '?': case 'h': // help usage PrintVersion(); PrintUsage(); return 0; case '-': PrintVersion(); PrintUsage(); fprintf(stderr, "\nWe need no long options\n"); return -1; case ':': PrintVersion(); fprintf(stderr, "Missing argument for option '%c'\n", optopt); return -1; default: PrintVersion(); fprintf(stderr, "Unkown option '%c'\n", optopt); return -1; } break; } if (optind < argc) { PrintVersion(); while (optind < argc) { fprintf(stderr, "Unhandled argument '%s'\n", argv[optind++]); } return -1; } // // main loop // AudioInit(); for (;;) { unsigned u; uint8_t buffer[16 * 1024]; // some random data for (u = 0; u < sizeof(buffer); u++) { buffer[u] = random() & 0xffff; } Debug(3, "audio/test: loop\n"); for (;;) { while (RingBufferFreeBytes(AlsaRingBuffer) > sizeof(buffer)) { AlsaEnqueue(buffer, sizeof(buffer)); } } } AudioExit(); return 0; } #endif