/// /// @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. /// @todo FIXME: can combine OSS and alsa ring buffer /// //#define USE_ALSA ///< enable alsa support //#define USE_OSS ///< enable OSS support #define USE_AUDIO_THREAD ///< use thread for audio playback #define noUSE_AUDIORING ///< new audio ring code (incomplete) #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 void (*Thread) (void); ///< module thread handler void (*Enqueue) (const void *, int); ///< enqueue samples for output void (*FlushBuffers) (void); ///< flush sample buffers void (*Poller) (void); ///< output poller int (*FreeBytes) (void); ///< number of bytes free in buffer uint64_t(*GetDelay) (void); ///< get current audio delay void (*SetVolume) (int); ///< set output volume int (*Setup) (int *, int *, int); ///< setup channels, samplerate void (*Init) (void); ///< initialize audio output module void (*Exit) (void); ///< cleanup audio output module } AudioModule; static const AudioModule NoopModule; ///< forward definition of noop module //---------------------------------------------------------------------------- // Variables //---------------------------------------------------------------------------- static const char *AudioModuleName; ///< which audio module to use /// Selected audio module. static const AudioModule *AudioUsedModule = &NoopModule; static const char *AudioPCMDevice; ///< alsa/OSS PCM device name static const char *AudioAC3Device; ///< alsa/OSS AC3 device name static const char *AudioMixerDevice; ///< alsa/OSS mixer device name static const char *AudioMixerChannel; ///< alsa/OSS mixer channel name static volatile char AudioRunning; ///< thread running / stopped static int AudioPaused; ///< audio paused static unsigned AudioSampleRate; ///< audio sample rate in hz static unsigned AudioChannels; ///< number of audio channels static const int AudioBytesProSample = 2; ///< number of bytes per sample static int64_t AudioPTS; ///< audio pts clock static const int AudioBufferTime = 350; ///< audio buffer time in ms #ifdef USE_AUDIO_THREAD static pthread_t AudioThread; ///< audio play thread static pthread_mutex_t AudioMutex; ///< audio condition mutex static pthread_cond_t AudioStartCond; ///< condition variable #else static const int AudioThread; ///< dummy audio thread #endif #ifdef USE_AUDIORING //---------------------------------------------------------------------------- // ring buffer //---------------------------------------------------------------------------- // FIXME: use this code, to combine alsa&OSS ring buffers #define AUDIO_RING_MAX 8 ///< number of audio ring buffers /** ** Audio ring buffer. */ typedef struct _audio_ring_ring_ { char FlushBuffers; ///< flag: flush buffers unsigned SampleRate; ///< sample rate in hz unsigned Channels; ///< number of channels } 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 /** ** Add sample rate, number of channel change to ring. ** ** @param freq sample frequency ** @param channels number of channels */ static int AudioRingAdd(int freq, int channels) { int filled; filled = atomic_read(&AudioRingFilled); if (filled == AUDIO_RING_MAX) { // no free slot // FIXME: can wait for ring buffer empty Error(_("audio: out of ring buffers\n")); return -1; } AudioRing[AudioRingWrite].FlushBuffers = 1; AudioRing[AudioRingWrite].SampleRate = freq; AudioRing[AudioRingWrite].Channels = channels; AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX; atomic_inc(&AudioRingFilled); #ifdef USE_AUDIO_THREAD // tell thread, that 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) { // FIXME: //AlsaRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit } // one slot always reservered AudioRingWrite = 1; atomic_set(&AudioRingFilled, 1); } /** ** Cleanup audio ring. */ static void AudioRingExit(void) { int i; for (i = 0; i < AUDIO_RING_MAX; ++i) { // FIXME: //RingBufferDel(AlsaRingBuffer); } } #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 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 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 //---------------------------------------------------------------------------- /** ** 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? } // Update audio clock AudioPTS += ((int64_t) count * 90000) / (AudioSampleRate * AudioChannels * AudioBytesProSample); if (!AudioRunning) { if (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: underrun error?\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) { Error(_("audio/alsa: broken driver %d\n"), avail); 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 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: 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)); 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) { RingBufferReadAdvance(AlsaRingBuffer, RingBufferUsedBytes(AlsaRingBuffer)); state = snd_pcm_state(AlsaPCMHandle); Debug(3, "audio/alsa: state %d - %s\n", state, 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; 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; } #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"); AudioPaused = 0; } } // Update audio clock // AudioPTS += (size * 90000) / (AudioSampleRate * AudioChannels * AudioBytesProSample); } #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 #ifdef USE_AUDIO_THREAD //---------------------------------------------------------------------------- // thread playback //---------------------------------------------------------------------------- /** ** Alsa thread */ static void 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; } // wait for space in kernel buffers if ((err = snd_pcm_wait(AlsaPCMHandle, 100)) < 0) { Error(_("audio/alsa: wait underrun error?\n")); err = snd_pcm_recover(AlsaPCMHandle, err, 0); if (err >= 0) { continue; } Error(_("audio/alsa: snd_pcm_wait(): %s\n"), snd_strerror(err)); usleep(100 * 1000); continue; } if (AlsaFlushBuffer) { 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\n"); break; } pthread_yield(); usleep(20 * 1000); // let fill/empty the buffers } } } /** ** 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 || !AudioSampleRate) { 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); } } /** ** 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 //---------------------------------------------------------------------------- /** ** 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 = getenv("ALSA_PASSTHROUGH_DEVICE")))) && !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) { device = "default"; } Debug(3, "audio/alsa: &&|| hell '%s'\n", 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; snd_pcm_uframes_t buffer_size; if (!(handle = AlsaOpenPCM(0))) { return; } 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"); snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size); Info(_("audio/alsa: max buffer size %lu\n"), buffer_size); AlsaPCMHandle = handle; } //---------------------------------------------------------------------------- // Alsa Mixer //---------------------------------------------------------------------------- /** ** Set alsa mixer volume (0-100) ** ** @param volume volume (0 .. 100) */ static void AlsaSetVolume(int volume) { int v; if (AlsaMixer && AlsaMixerElem) { v = (volume * AlsaRatio) / 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) == 0) { 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)) / 100; 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 uint64_t AlsaGetDelay(void) { int err; snd_pcm_sframes_t delay; uint64_t pts; if (!AlsaPCMHandle || !AudioSampleRate) { return 0UL; } // 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 = ((uint64_t) delay * 90 * 1000) / AudioSampleRate; pts += ((uint64_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; snd_pcm_t *handle; if (!AlsaPCMHandle) { // alsa not running yet return -1; } #if 1 // easy alsa hw setup way // flush any buffered data AudioFlushBuffers(); 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, 125 * 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); Info(_("audio/alsa: buffer size %lu, period size %lu\n"), buffer_size, period_size); 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 if (AlsaStartThreshold < (*freq * *channels * AudioBytesProSample * AudioBufferTime) / 1000U) { AlsaStartThreshold = (*freq * *channels * AudioBytesProSample * AudioBufferTime) / 1000U; } // no bigger, than the buffer if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) { AlsaStartThreshold = RingBufferFreeBytes(AlsaRingBuffer); } Info(_("audio/alsa: delay %u ms\n"), (AlsaStartThreshold * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample)); return ret; } /** ** 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 AlsaRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit 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; } if (AlsaRingBuffer) { RingBufferDel(AlsaRingBuffer); AlsaRingBuffer = NULL; } AlsaFlushBuffer = 0; } /** ** Alsa module. */ static const AudioModule AlsaModule = { .Name = "alsa", #ifdef USE_AUDIO_THREAD .Thread = AlsaThread, .Enqueue = AlsaThreadEnqueue, .FlushBuffers = AlsaThreadFlushBuffers, #else .Enqueue = AlsaEnqueue, .FlushBuffers = AlsaFlushBuffers, #endif .Poller = AlsaPoller, .FreeBytes = AlsaFreeBytes, .GetDelay = AlsaGetDelay, .SetVolume = AlsaSetVolume, .Setup = AlsaSetup, .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 RingBuffer *OssRingBuffer; ///< audio ring buffer static unsigned OssStartThreshold; ///< start play, if filled #ifdef USE_AUDIO_THREAD static volatile char OssFlushBuffer; ///< flag empty buffer #endif //---------------------------------------------------------------------------- // 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? } // Update audio clock AudioPTS += ((int64_t) count * 90000) / (AudioSampleRate * AudioChannels * AudioBytesProSample); if (!AudioRunning) { if (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; } Error(_("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; 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 %d ms\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; } #ifdef USE_AUDIO_THREAD //---------------------------------------------------------------------------- // thread playback //---------------------------------------------------------------------------- /** ** OSS thread */ static void 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; } fds[0].fd = OssPcmFildes; fds[0].events = POLLOUT | POLLERR; // wait for space in kernel buffers err = poll(fds, 1, 100); if (err < 0) { Error(_("audio/oss: error poll %s\n"), strerror(errno)); usleep(100 * 1000); continue; } if (OssFlushBuffer) { continue; } if ((err = OssPlayRingbuffer())) { // empty / error if (err < 0) { // underrun error break; } pthread_yield(); usleep(20 * 1000); // let fill/empty the buffers } } } /** ** 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 || !AudioSampleRate) { 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); } } /** ** 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 //---------------------------------------------------------------------------- /** ** 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"; } Debug(3, "audio/oss: &&|| hell '%s'\n", 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 (0-100) ** ** @param volume volume (0 .. 100) */ static void OssSetVolume(int volume) { int v; if (OssMixerFildes != -1) { v = (volume * 255) / 100; 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 //---------------------------------------------------------------------------- /** ** Get OSS audio delay in time stamps. ** ** @returns audio delay in time stamps. */ static uint64_t OssGetDelay(void) { int delay; uint64_t pts; if (OssPcmFildes == -1) { // setup failure return 0UL; } if (!AudioRunning) { return 0UL; } // 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 == -1) { delay = 0UL; } pts = ((uint64_t) delay * 90 * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample); pts += ((uint64_t) RingBufferUsedBytes(OssRingBuffer) * 90 * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample); if (pts > 600 * 90) { 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; 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 %d Hz sample rate.\n"), *freq); *freq = tmp; ret = 1; } AudioChannels = *channels; AudioSampleRate = *freq; // FIXME: setup buffers if (1) { audio_buf_info bi; if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) { Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"), strerror(errno)); } else { Debug(3, "audio/oss: %d bytes buffered\n", bi.bytes); } tmp = -1; if (ioctl(OssPcmFildes, SNDCTL_DSP_GETODELAY, &tmp) == -1) { Error(_("audio/oss: ioctl(SNDCTL_DSP_GETODELAY): %s\n"), strerror(errno)); // FIXME: stop player, set setup failed flag return -1; } if (tmp == -1) { tmp = 0; } // start when enough bytes for initial write OssStartThreshold = bi.bytes + tmp; // buffer time/delay in ms if (OssStartThreshold < (*freq * *channels * AudioBytesProSample * AudioBufferTime) / 1000U) { OssStartThreshold = (*freq * *channels * AudioBytesProSample * AudioBufferTime) / 1000U; } // no bigger, than the buffer if (OssStartThreshold > RingBufferFreeBytes(OssRingBuffer)) { OssStartThreshold = RingBufferFreeBytes(OssRingBuffer); } Info(_("audio/oss: delay %u ms\n"), (OssStartThreshold * 1000) / (AudioSampleRate * AudioChannels * AudioBytesProSample)); } return ret; } /** ** Initialize OSS audio output module. */ static void OssInit(void) { OssRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit 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, .Enqueue = OssThreadEnqueue, .FlushBuffers = OssThreadFlushBuffers, #else .Enqueue = OssEnqueue, .FlushBuffers = OssFlushBuffers, #endif .Poller = OssPoller, .FreeBytes = OssFreeBytes, .GetDelay = OssGetDelay, .SetVolume = OssSetVolume, .Setup = OssSetup, .Init = OssInit, .Exit = OssExit, }; #endif // USE_OSS //============================================================================ // Noop //============================================================================ /** ** 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 audio delay in time stamps. ** ** @returns audio delay in time stamps. */ static uint64_t NoopGetDelay(void) { return 0UL; } /** ** Set mixer volume (0-100) ** ** @param volume volume (0 .. 100) */ 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", .Enqueue = NoopEnqueue, .FlushBuffers = NoopVoid, .Poller = NoopVoid, .FreeBytes = NoopFreeBytes, .GetDelay = NoopGetDelay, .SetVolume = NoopSetVolume, .Setup = NoopSetup, .Init = NoopVoid, .Exit = NoopVoid, }; //---------------------------------------------------------------------------- // thread playback //---------------------------------------------------------------------------- #ifdef USE_AUDIO_THREAD /** ** Audio play thread. */ 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); #ifdef USE_AUDIORING if (atomic_read(&AudioRingFilled) > 1) { int sample_rate; int channels; // skip all sample changes between while (atomic_read(&AudioRingFilled) > 1) { Debug(3, "audio: skip ring buffer\n"); AudioRingRead = (AudioRingRead + 1) % AUDIO_RING_MAX; atomic_dec(&AudioRingFilled); } #ifdef USE_ALSA // FIXME: flush only if there is something to flush AlsaFlushBuffers(); sample_rate = AudioRing[AudioRingRead].SampleRate; channels = AudioRing[AudioRingRead].Channels; Debug(3, "audio: thread channels %d sample-rate %d hz\n", channels, sample_rate); if (AlsaSetup(&sample_rate, &channels)) { Error(_("audio: can't set channels %d sample-rate %d hz\n"), channels, sample_rate); } Debug(3, "audio: thread channels %d sample-rate %d hz\n", AudioChannels, AudioSampleRate); #endif } #endif Debug(3, "audio: play start\n"); AudioUsedModule->Thread(); } return dummy; } /** ** 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"); pthread_yield(); usleep(5 * 1000); // give thread some time to start } /** ** 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) { AudioUsedModule->Enqueue(samples, count); } /** ** Flush audio buffers. */ void AudioFlushBuffers(void) { AudioUsedModule->FlushBuffers(); } /** ** Call back to play audio polled. */ void AudioPoller(void) { AudioUsedModule->Poller(); } /** ** Get free bytes in audio output. */ int AudioFreeBytes(void) { return AudioUsedModule->FreeBytes(); } /** ** Get audio delay in time stamps. ** ** @returns audio delay in time stamps. */ uint64_t AudioGetDelay(void) { return AudioUsedModule->GetDelay(); } /** ** Set audio clock base. ** ** @param pts audio presentation timestamp */ void AudioSetClock(int64_t pts) { #ifdef DEBUG if (AudioPTS != pts) { Debug(4, "audio: set clock to %#012" PRIx64 " %#012" PRIx64 " pts\n", AudioPTS, pts); } #endif AudioPTS = pts; } /** ** Get current audio clock. ** ** @returns the audio clock in time stamps. */ int64_t AudioGetClock(void) { if ((uint64_t) AudioPTS != INT64_C(0x8000000000000000)) { int64_t delay; if ((delay = AudioGetDelay())) { return AudioPTS - delay; } } return INT64_C(0x8000000000000000); } /** ** Set mixer volume (0-100) ** ** @param volume volume (0 .. 100) */ void AudioSetVolume(int volume) { #ifdef USE_ALSA AlsaSetVolume(volume); #endif #ifdef USE_OSS OssSetVolume(volume); #endif (void)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 ** ** @todo audio changes must be queued and done when the buffer is empty */ int AudioSetup(int *freq, int *channels, int use_ac3) { Debug(3, "audio: channels %d frequency %d hz %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 // FIXME: need to store possible combination and report this return AudioRingAdd(*freq, *channels, use_ac3); #endif return AudioUsedModule->Setup(freq, channels, use_ac3); } /** ** 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; } /** ** Initialize audio output module. ** ** @todo FIXME: make audio output module selectable. */ void AudioInit(void) { int freq; int chan; unsigned u; const char *name; 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: #ifdef USE_AUDIORING AudioRingInit(); #endif AudioUsedModule->Init(); freq = 48000; chan = 2; if (AudioSetup(&freq, &chan, 0)) { // set default parameters Error(_("audio: can't do initial setup\n")); } #ifdef USE_AUDIO_THREAD if (AudioUsedModule->Thread) { // supports threads AudioInitThread(); } #endif AudioPaused = 1; } /** ** Cleanup audio output module. */ void AudioExit(void) { #ifdef USE_AUDIO_THREAD AudioExitThread(); #endif AudioUsedModule->Exit(); AudioUsedModule = &NoopModule; #ifdef USE_AUDIORING AudioRingExit(); #endif AudioRunning = 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