/// /// @file audio.c @brief Audio module /// /// Copyright (c) 2009 - 2014 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: 77fa65030b179e78c13d0bf69a7cc417dae89e1a $ ////////////////////////////////////////////////////////////////////////////// /// /// @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! /// /// /// @todo FIXME: there can be problems with little/big endian. /// #ifdef DEBUG #undef DEBUG #endif #define USE_AUDIO_THREAD ///< use thread for audio playback #define USE_AUDIO_MIXER ///< use audio module mixer #include #include #include #include #include #include #include #include #include #define _(str) gettext(str) ///< gettext shortcut #define _N(str) str ///< gettext_noop shortcut #include #ifdef USE_AUDIO_THREAD #include #include #include #endif #include "iatomic.h" // portable atomic_t #include "audio.h" #include "misc.h" #include "ringbuffer.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 void (*const FlushBuffers)(void); ///< flush sample buffers 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 audio void (*const Pause)(void); ///< pause audio 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 AudioAlsaNotest; ///< disable Audio capbility test char AudioAlsaDriverBroken; ///< disable broken driver message char AudioAlsaNoCloseOpen; ///< disable alsa close/open fix char AudioAlsaCloseOpenDelay; ///< enable alsa close/open delay fix 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 *AudioPassthroughDevice; ///< Passthrough device name static char AudioAppendAES; ///< flag automatic append AES 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 static int AudioSkip; ///< skip audio to sync to video 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 static char AudioThreadStop; ///< stop audio thread #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 (3 * 5 * 7 * 8) static const unsigned AudioRingBufferSize = 3 * 5 * 7 * 8 * 2 * 1000; 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 (must be sorted by frequency) static const unsigned AudioRatesTable[AudioRatesMax] = {44100, 48000, 192000}; //---------------------------------------------------------------------------- // 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; } } #ifdef USE_AUDIO_MIXER /** ** 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] * 100; // 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; } } /** ** Upmix @a in_chan channels to @a out_chan. ** ** @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 AudioUpmix(const int16_t *in, int in_chan, int frames, int16_t *out, int out_chan) { while (frames--) { int i; for (i = 0; i < in_chan; ++i) { // copy existing channels *out++ = *in++; } for (; i < out_chan; ++i) { // silents missing channels *out++ = 0; } } } /** ** Resample ffmpeg sample format to hardware format. ** ** FIXME: use libswresample for this and move it to codec. ** FIXME: ffmpeg to alsa conversion is already done in codec.c. ** ** 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 ** @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; case 5 * 8 + 6: case 3 * 8 + 8: case 5 * 8 + 8: case 6 * 8 + 8: AudioUpmix(in, in_chan, frames, out, out_chan); break; default: Error("audio: unsupported %d -> %d channels resample\n", in_chan, out_chan); // play silence memset(out, 0, frames * out_chan * AudioBytesProSample); break; } } #endif //---------------------------------------------------------------------------- // 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 Passthrough; ///< flag: use pass-through (AC-3, ...) int16_t PacketSize; ///< packet size 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; /// 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 channels change to ring. ** ** @param sample_rate sample-rate frequency ** @param channels number of channels ** @param passthrough use /pass-through (AC-3, ...) device ** ** @retval -1 error ** @retval 0 okay ** ** @note this function shouldn't fail. Checks are done during AudoInit. */ static int AudioRingAdd(unsigned sample_rate, int channels, int passthrough) { unsigned u; // search supported sample-rates for (u = 0; u < AudioRatesMax; ++u) { if (AudioRatesTable[u] == sample_rate) { goto found; } if (AudioRatesTable[u] > sample_rate) { break; } } Error(_("audio: %dHz sample-rate unsupported\n"), sample_rate); return -1; // unsupported sample-rate found: 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; AudioRing[AudioRingWrite].FlushBuffers = 0; AudioRing[AudioRingWrite].Passthrough = passthrough; AudioRing[AudioRingWrite].PacketSize = 0; AudioRing[AudioRingWrite].InSampleRate = sample_rate; AudioRing[AudioRingWrite].InChannels = channels; AudioRing[AudioRingWrite].HwSampleRate = sample_rate; AudioRing[AudioRingWrite].HwChannels = AudioChannelMatrix[u][channels]; AudioRing[AudioRingWrite].PTS = AV_NOPTS_VALUE; RingBufferReset(AudioRing[AudioRingWrite].RingBuffer); Debug(3, "audio: %d ring buffer prepared\n", atomic_read(&AudioRingFilled) + 1); atomic_inc(&AudioRingFilled); #ifdef USE_AUDIO_THREAD if (AudioThread) { // tell thread, that there is something todo AudioRunning = 1; pthread_cond_signal(&AudioStartCond); Debug(3, "Start on AudioRingAdd\n"); } #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; } //============================================================================ // 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 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 //---------------------------------------------------------------------------- // 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; } // muting pass-through AC-3, can produce disturbance if (AudioMute || (AudioSoftVolume && !AudioRing[AudioRingRead].Passthrough)) { // 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); #ifdef DEBUG if (avail != snd_pcm_frames_to_bytes(AlsaPCMHandle, frames)) { Error(_("audio/alsa: bytes lost -> out of sync\n")); } #endif 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)); } } } } #ifdef USE_AUDIO_THREAD //---------------------------------------------------------------------------- // thread playback //---------------------------------------------------------------------------- /** ** 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 (;;) { 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; } #endif //---------------------------------------------------------------------------- /** ** Open alsa pcm device. ** ** @param passthrough use pass-through (AC-3, ...) device */ static snd_pcm_t *AlsaOpenPCM(int passthrough) { const char *device; snd_pcm_t *handle; int err; char tmp[80]; // &&|| hell if (!(passthrough && ((device = AudioPassthroughDevice) || (device = getenv("ALSA_PASSTHROUGH_DEVICE")))) && !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) { device = "default"; } if (!AudioDoingInit) { // reduce blabla during init Info(_("audio/alsa: using %sdevice '%s'\n"), passthrough ? "pass-through " : "", device); } //printf("audio/alsa: using %sdevice '%s'\n", passthrough ? "pass-through " : "", device); // // for AC3 pass-through try to set the non-audio bit, use AES0=6 // if (passthrough && AudioAppendAES) { if (!(strchr(device, ':'))) { sprintf(tmp, //"AES0=%d,AES1=%d,AES2=0,AES3=%d", "%s:AES0=%d,AES1=%d,AES2=0", device, IEC958_AES0_NONAUDIO | IEC958_AES0_PRO_EMPHASIS_NONE, IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER); //map_iec958_srate(ao->samplerate)); } else { sprintf(tmp, //"AES0=%d,AES1=%d,AES2=0,AES3=%d", "%s,AES0=%d,AES1=%d,AES2=0", device, IEC958_AES0_NONAUDIO | IEC958_AES0_PRO_EMPHASIS_NONE, IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER); //map_iec958_srate(ao->samplerate)); } printf( "opening device '%s' => '%s'\n", device, tmp); if ((err = snd_pcm_open(&handle, tmp, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK)) < 0 ) { Error(_("audio/alsa: playback open '%s' error: %s\n"), device, snd_strerror(err)); return NULL; } } else { // 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 //---------------------------------------------------------------------------- /** ** 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(4, "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 passthrough use pass-through (AC-3, ...) 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 passthrough) { 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 fe. pass-through, we never recover return -1; } if (!AudioAlsaNoCloseOpen) { // close+open to fix HDMI no sound bug snd_pcm_t *handle; handle = AlsaPCMHandle; // no lock needed, thread exit in main loop only // Debug(3, "audio: %s [\n", __FUNCTION__); AlsaPCMHandle = NULL; // other threads should check handle snd_pcm_close(handle); if (AudioAlsaCloseOpenDelay) { usleep(50 * 1000); // 50ms delay for alsa recovery } // FIXME: can use multiple retries if (!(handle = AlsaOpenPCM(passthrough))) { return -1; } AlsaPCMHandle = handle; // Debug(3, "audio: %s ]\n", __FUNCTION__); } 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))) { // try reduced buffer size (needed for sunxi) // FIXME: alternativ make this configurable 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, 72 * 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; } /** ** Play audio. */ static 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. */ static 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) { #ifdef DEBUG (void)AlsaNoopCallback; #else // disable display of alsa error messages snd_lib_error_set_handler(AlsaNoopCallback); #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; } } /** ** Alsa module. */ static const AudioModule AlsaModule = { .Name = "alsa", #ifdef USE_AUDIO_THREAD .Thread = AlsaThread, #endif .FlushBuffers = AlsaFlushBuffers, .GetDelay = AlsaGetDelay, .SetVolume = AlsaSetVolume, .Setup = AlsaSetup, .Play = AlsaPlay, .Pause = AlsaPause, .Init = AlsaInit, .Exit = AlsaExit, }; //============================================================================ // Noop //============================================================================ /** ** 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 ** @param passthrough use pass-through (AC-3, ...) device */ static int NoopSetup(__attribute__((unused)) int *channels, __attribute__((unused)) int *freq, __attribute__((unused)) int passthrough) { return -1; } /** ** Noop void */ static void NoopVoid(void) {} /** ** Noop module. */ static const AudioModule NoopModule = { .Name = "noop", .FlushBuffers = NoopVoid, .GetDelay = NoopGetDelay, .SetVolume = NoopSetVolume, .Setup = NoopSetup, .Play = NoopVoid, .Pause = NoopVoid, .Init = NoopVoid, .Exit = NoopVoid, }; //---------------------------------------------------------------------------- // thread playback //---------------------------------------------------------------------------- #ifdef USE_AUDIO_THREAD /** ** Prepare next ring buffer. */ static int AudioNextRing(void) { int passthrough; int sample_rate; int channels; size_t used; // update audio format // not always needed, but check if needed is too complex passthrough = AudioRing[AudioRingRead].Passthrough; sample_rate = AudioRing[AudioRingRead].HwSampleRate; channels = AudioRing[AudioRingRead].HwChannels; if (AudioUsedModule->Setup(&sample_rate, &channels, passthrough)) { 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(); Debug(3, "audio: a/v next buf(%d,%4zdms)\n", atomic_read(&AudioRingFilled), (RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer) * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample)); // stop, if not enough in next buffer used = RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer); if (AudioStartThreshold * 4 < used || (AudioVideoIsReady && AudioStartThreshold < used)) { return 0; } return 1; } /** ** Audio play thread. ** ** @param dummy unused thread argument */ static void *AudioPlayHandlerThread(void *dummy) { Debug(3, "audio: play thread started\n"); prctl(PR_SET_NAME, "cuvid audio", 0, 0, 0); for (;;) { // check if we should stop the thread if (AudioThreadStop) { Debug(3, "audio: play thread stopped\n"); return PTHREAD_CANCELED; } 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 %d start\n", (AudioUsedBytes() * 1000) / (!AudioRing[AudioRingWrite].HwSampleRate + !AudioRing[AudioRingWrite].HwChannels + AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample), AudioUsedBytes()); do { int filled; int read; int flush; int err; int i; // check if we should stop the thread if (AudioThreadStop) { Debug(3, "audio: play thread stopped\n"); return PTHREAD_CANCELED; } // look if there is a flush command in the queue flush = 0; filled = atomic_read(&AudioRingFilled); read = AudioRingRead; i = filled; while (i--) { read = (read + 1) % AUDIO_RING_MAX; if (AudioRing[read].FlushBuffers) { AudioRing[read].FlushBuffers = 0; AudioRingRead = read; // handle all flush in queue flush = filled - i; } } if (flush) { Debug(3, "audio: flush %d ring buffer(s)\n", flush); AudioUsedModule->FlushBuffers(); atomic_sub(flush, &AudioRingFilled); if (AudioNextRing()) { break; } } // try to play some samples err = 0; if (RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer)) { err = AudioUsedModule->Thread(); } // underrun, check if new ring buffer is available if (!err) { int passthrough; int sample_rate; int channels; int old_passthrough; int old_sample_rate; int old_channels; // underrun, and no new ring buffer, goto sleep. if (!atomic_read(&AudioRingFilled)) { Debug(3, "audio: HandlerThread Underrun with no new data\n"); break; } Debug(3, "audio: next ring buffer\n"); old_passthrough = AudioRing[AudioRingRead].Passthrough; old_sample_rate = AudioRing[AudioRingRead].HwSampleRate; old_channels = AudioRing[AudioRingRead].HwChannels; atomic_dec(&AudioRingFilled); AudioRingRead = (AudioRingRead + 1) % AUDIO_RING_MAX; passthrough = AudioRing[AudioRingRead].Passthrough; sample_rate = AudioRing[AudioRingRead].HwSampleRate; channels = AudioRing[AudioRingRead].HwChannels; Debug(3, "audio: thread channels %d frequency %dHz %s\n", channels, sample_rate, passthrough ? "pass-through" : ""); // audio config changed? if (old_passthrough != passthrough || old_sample_rate != sample_rate || old_channels != channels) { // FIXME: wait for buffer drain if (AudioNextRing()) { Debug(3, "audio: HandlerThread break on nextring"); break; } } else { AudioResetCompressor(); AudioResetNormalizer(); } } // FIXME: check AudioPaused ...Thread() if (AudioPaused) { Debug(3, "audio: HandlerThread break on paused"); break; } } while (AudioRing[AudioRingRead].HwSampleRate); } return dummy; } /** ** Initialize audio thread. */ static void AudioInitThread(void) { AudioThreadStop = 0; pthread_mutex_init(&AudioMutex, NULL); pthread_cond_init(&AudioStartCond, NULL); pthread_create(&AudioThread, NULL, AudioPlayHandlerThread, NULL); pthread_setname_np(AudioThread, "softhddev audio"); } /** ** Cleanup audio thread. */ static void AudioExitThread(void) { void *retval; Debug(3, "audio: %s\n", __FUNCTION__); if (AudioThread) { AudioThreadStop = 1; AudioRunning = 1; // wakeup thread, if needed pthread_cond_signal(&AudioStartCond); 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[] = { &AlsaModule, &NoopModule, }; void AudioDelayms(int delayms) { int count; unsigned char *p; #ifdef DEBUG printf("Try Delay Audio for %d ms Samplerate %d Channels %d bps %d\n", delayms, AudioRing[AudioRingWrite].HwSampleRate, AudioRing[AudioRingWrite].HwChannels, AudioBytesProSample); #endif count = delayms * AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample / 1000; if (delayms < 5000 && delayms > 0) { // not more than 5seconds p = calloc(1, count); RingBufferWrite(AudioRing[AudioRingWrite].RingBuffer, p, count); free(p); } } /** ** 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) { size_t n; int16_t *buffer; #ifdef noDEBUG 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 } // save packet size if (!AudioRing[AudioRingWrite].PacketSize) { AudioRing[AudioRingWrite].PacketSize = count; Debug(3, "audio: a/v packet size %d bytes\n", count); } // audio sample modification allowed and needed? buffer = (void *)samples; if (!AudioRing[AudioRingWrite].Passthrough && (AudioCompression || AudioNormalize || AudioRing[AudioRingWrite].InChannels != AudioRing[AudioRingWrite].HwChannels)) { int frames; // resample into ring-buffer is too complex in the case of a roundabout // just use a temporary buffer frames = count / (AudioRing[AudioRingWrite].InChannels * AudioBytesProSample); buffer = alloca(frames * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample); #ifdef USE_AUDIO_MIXER // Convert / resample input to hardware format AudioResample(samples, AudioRing[AudioRingWrite].InChannels, frames, buffer, AudioRing[AudioRingWrite].HwChannels); #else #ifdef DEBUG if (AudioRing[AudioRingWrite].InChannels != AudioRing[AudioRingWrite].HwChannels) { Debug(3, "audio: internal failure channels mismatch\n"); return; } #endif memcpy(buffer, samples, count); #endif count = frames * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample; 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 int skip; n = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer); skip = AudioSkip; // FIXME: round to packet size Debug(4, "audio: start? %4zdms skip %dms\n", (n * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample), (skip * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample)); if (skip) { if (n < (unsigned)skip) { skip = n; } AudioSkip -= skip; RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, skip); n = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer); } // forced start or enough video + audio buffered // for some exotic channels * 4 too small if (AudioStartThreshold * 4 < n || (AudioVideoIsReady // if ((AudioVideoIsReady && AudioStartThreshold < n)) { // restart play-back // no lock needed, can wakeup next time AudioRunning = 1; pthread_cond_signal(&AudioStartCond); Debug(3, "Start on AudioEnque Threshold %d n %d\n", AudioStartThreshold, n); } } // Update audio clock (stupid gcc developers thinks INT64_C is unsigned) if (AudioRing[AudioRingWrite].PTS != (int64_t)AV_NOPTS_VALUE) { AudioRing[AudioRingWrite].PTS += ((int64_t)count * 90 * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample); } } /** ** Video is ready. ** ** @param pts video presentation timestamp */ void AudioVideoReady(int64_t pts) { int64_t audio_pts; size_t used; if (pts == (int64_t)AV_NOPTS_VALUE) { Debug(3, "audio: a/v start, no valid video\n"); return; } // no valid audio known if (!AudioRing[AudioRingWrite].HwSampleRate || !AudioRing[AudioRingWrite].HwChannels || AudioRing[AudioRingWrite].PTS == (int64_t)AV_NOPTS_VALUE) { Debug(3, "audio: a/v start, no valid audio\n"); AudioVideoIsReady = 1; return; } // Audio.PTS = next written sample time stamp used = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer); audio_pts = AudioRing[AudioRingWrite].PTS - (used * 90 * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample); Debug(3, "audio: a/v sync buf(%d,%4zdms) %s | %s = %dms %s\n", atomic_read(&AudioRingFilled), (used * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample), Timestamp2String(pts), Timestamp2String(audio_pts), (int)(pts - audio_pts) / 90, AudioRunning ? "running" : "ready"); if (!AudioRunning) { int skip; // buffer ~15 video frames // FIXME: HDTV can use smaller video buffer skip = pts - 0 * 20 * 90 - AudioBufferTime * 90 - audio_pts + VideoAudioDelay; #ifdef DEBUG // fprintf(stderr, "a/v-diff %dms a/v-delay %dms skip %dms Audiobuffer //%d\n", (int)(pts - audio_pts) / 90, VideoAudioDelay / 90, skip / // 90,AudioBufferTime); #endif // guard against old PTS if (skip > 0 && skip < 4000 * 90) { skip = (((int64_t)skip * AudioRing[AudioRingWrite].HwSampleRate) / (1000 * 90)) * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample; // FIXME: round to packet size if ((unsigned)skip > used) { AudioSkip = skip - used; skip = used; } Debug(3, "audio: sync advance %dms %d/%zd Rest %d\n", (skip * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample), skip, used, AudioSkip); RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, skip); used = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer); } else { Debug(3, "No audio skip -> should skip %d\n", skip / 90); } // FIXME: skip<0 we need bigger audio buffer // enough video + audio buffered if (AudioStartThreshold < used) { AudioRunning = 1; pthread_cond_signal(&AudioStartCond); Debug(3, "Start on AudioVideoReady\n"); } } AudioVideoIsReady = 1; } /** ** Flush audio buffers. */ void AudioFlushBuffers(void) { int old; int i; if (atomic_read(&AudioRingFilled) >= AUDIO_RING_MAX) { // wait for space in ring buffer, should never happen for (i = 0; i < 24 * 2; ++i) { if (atomic_read(&AudioRingFilled) < AUDIO_RING_MAX) { break; } Debug(3, "audio: flush out of ring buffers\n"); usleep(1 * 1000); // avoid hot polling } if (atomic_read(&AudioRingFilled) >= AUDIO_RING_MAX) { // FIXME: We can set the flush flag in the last wrote ring buffer Error(_("audio: flush out of ring buffers\n")); return; } } old = AudioRingWrite; AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX; AudioRing[AudioRingWrite].FlushBuffers = 1; AudioRing[AudioRingWrite].Passthrough = AudioRing[old].Passthrough; 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 = AV_NOPTS_VALUE; RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer)); Debug(3, "audio: reset video ready\n"); AudioVideoIsReady = 0; AudioSkip = 0; atomic_inc(&AudioRingFilled); // FIXME: wait for flush complete needed? for (i = 0; i < 24 * 2; ++i) { if (!AudioRunning) { // wakeup thread to flush buffers AudioRunning = 1; pthread_cond_signal(&AudioStartCond); Debug(3, "Start on Flush\n"); } // FIXME: waiting on zero isn't correct, but currently works if (!atomic_read(&AudioRingFilled)) { break; } usleep(1 * 1000); // avoid hot polling } Debug(3, "audio: audio flush %dms\n", i); } /** ** Call back to play audio polled. */ void AudioPoller(void) { // FIXME: write poller } /** ** Get free bytes in audio output. */ int AudioFreeBytes(void) { return AudioRing[AudioRingWrite].RingBuffer ? RingBufferFreeBytes(AudioRing[AudioRingWrite].RingBuffer) : INT32_MAX; } /** ** Get used bytes in audio output. */ int AudioUsedBytes(void) { // FIXME: not correct, if multiple buffer are in use return AudioRing[AudioRingWrite].RingBuffer ? RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer) : 0; } /** ** Get audio delay in time stamps. ** ** @returns audio delay in time stamps. */ int64_t AudioGetDelay(void) { 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; // multiple buffers, invalid delay } pts = AudioUsedModule->GetDelay(); pts += ((int64_t)RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer) * 90 * 1000) / (AudioRing[AudioRingRead].HwSampleRate * AudioRing[AudioRingRead].HwChannels * AudioBytesProSample); Debug(4, "audio: hw+sw delay %zd %" PRId64 "ms\n", RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer), pts / 90); return pts; } /** ** Set audio clock base. ** ** @param pts audio presentation timestamp */ void AudioSetClock(int64_t pts) { if (AudioRing[AudioRingWrite].PTS != pts) { Debug(4, "audio: set clock %s -> %s pts\n", Timestamp2String(AudioRing[AudioRingWrite].PTS), Timestamp2String(pts)); } // printf("Audiosetclock pts %#012" PRIx64 " // %d\n",pts,RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer)); AudioRing[AudioRingWrite].PTS = pts; } /** ** Get current audio clock. ** ** @returns the audio clock in time stamps. */ int64_t AudioGetClock(void) { // (cast) needed for the evil gcc if (AudioRing[AudioRingRead].PTS != (int64_t)AV_NOPTS_VALUE) { int64_t delay; // delay zero, if no valid time stamp if ((delay = AudioGetDelay())) { if (AudioRing[AudioRingRead].Passthrough) { return AudioRing[AudioRingRead].PTS + 0 * 90 - delay; } return AudioRing[AudioRingRead].PTS + 0 * 90 - delay; } } return AV_NOPTS_VALUE; } /** ** Set mixer volume (0-1000) ** ** @param volume volume (0 .. 1000) */ void AudioSetVolume(int volume) { AudioVolume = volume; AudioMute = !volume; // reduce loudness for stereo output if (AudioStereoDescent && AudioRing[AudioRingRead].InChannels == 2 && !AudioRing[AudioRingRead].Passthrough) { volume -= AudioStereoDescent; if (volume < 0) { volume = 0; } else if (volume > 1000) { volume = 1000; } } AudioAmplifier = volume; if (!AudioSoftVolume) { AudioUsedModule->SetVolume(volume); } } /** ** Setup audio for requested format. ** ** @param freq sample frequency ** @param channels number of channels ** @param passthrough use pass-through (AC-3, ...) device ** ** @retval 0 everything ok ** @retval 1 didn't support frequency/channels combination ** @retval -1 something gone wrong ** ** @todo add support to report best fitting format. */ int AudioSetup(int *freq, int *channels, int passthrough) { Debug(3, "audio: setup channels %d frequency %dHz %s\n", *channels, *freq, passthrough ? "pass-through" : ""); // invalid parameter if (!freq || !channels || !*freq || !*channels) { Debug(3, "audio: bad channels or frequency parameters\n"); // FIXME: set flag invalid setup return -1; } return AudioRingAdd(*freq, *channels, passthrough); } /** ** 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 AudioSetPassthroughDevice(const char *device) { if (!AudioModuleName) { AudioModuleName = "alsa"; // detect alsa/OSS if (!device[0]) { AudioModuleName = "noop"; } else if (device[0] == '/') { AudioModuleName = "oss"; } } AudioPassthroughDevice = 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; } /** ** Set automatic AES flag handling. ** ** @param onoff turn setting AES flag on or off */ void AudioSetAutoAES(int onoff) { if (onoff < 0) { AudioAppendAES ^= 1; } else { AudioAppendAES = onoff; } } /** ** 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"; name = "alsa"; 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; AudioRingInit(); AudioUsedModule->Init(); if (AudioAlsaNotest) { for (u = 0; u < AudioRatesMax; ++u) { AudioChannelMatrix[u][1]=AudioChannelMatrix[u][2]=2; AudioChannelMatrix[u][3]=AudioChannelMatrix[u][4]=4; AudioChannelMatrix[u][5]=AudioChannelMatrix[u][6]=6; AudioChannelMatrix[u][7]=AudioChannelMatrix[u][8]=8; printf("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]); } AudioChannelsInHw[1]=AudioChannelsInHw[3]=AudioChannelsInHw[4]=AudioChannelsInHw[5]=AudioChannelsInHw[6]=AudioChannelsInHw[7]=AudioChannelsInHw[8]=0; AudioChannelsInHw[2]=2; } else { // // 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 // FIXME: use loop over sample-rates freq = 44100; AudioRatesInHw[Audio44100] = 0; for (chan = 1; chan < 9; ++chan) { int tchan; int tfreq; tchan = chan; tfreq = freq; if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) { AudioChannelsInHw[chan] = 0; } else { AudioChannelsInHw[chan] = chan; AudioRatesInHw[Audio44100] |= (1 << chan); } } freq = 48000; AudioRatesInHw[Audio48000] = 0; for (chan = 1; chan < 9; ++chan) { int tchan; int tfreq; if (!AudioChannelsInHw[chan]) { continue; } tchan = chan; tfreq = freq; if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) { // AudioChannelsInHw[chan] = 0; } else { AudioChannelsInHw[chan] = chan; AudioRatesInHw[Audio48000] |= (1 << chan); } } freq = 192000; AudioRatesInHw[Audio192000] = 0; for (chan = 1; chan < 9; ++chan) { int tchan; int tfreq; if (!AudioChannelsInHw[chan]) { continue; } tchan = chan; tfreq = freq; if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) { // AudioChannelsInHw[chan] = 0; } else { AudioChannelsInHw[chan] = chan; AudioRatesInHw[Audio192000] |= (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) { Debug(3,"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]); } } #ifdef USE_AUDIO_THREAD if (AudioUsedModule->Thread) { // supports threads AudioInitThread(); } #endif AudioDoingInit = 0; } /** ** Cleanup audio output module. */ void AudioExit(void) { const AudioModule *module; Debug(3, "audio: %s\n", __FUNCTION__); #ifdef USE_AUDIO_THREAD if (AudioUsedModule->Thread) { // supports threads AudioExitThread(); } #endif module = AudioUsedModule; AudioUsedModule = &NoopModule; module->Exit(); AudioRingExit(); 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 - 2013 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, "Unknown 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