diff options
| -rw-r--r-- | ChangeLog | 12 | ||||
| -rw-r--r-- | audio.c | 1564 |
2 files changed, 31 insertions, 1545 deletions
@@ -1,6 +1,18 @@ User johns Date: + Removes old audio code (!USE_AUDIORING). + Use -DOSD_DEBUG to debug OSD. + +User arttupii +Date: Tue Aug 7 16:46:23 2012 +0200 + + Fix bug #909: Subtitles destroy menu. + Fix bug #1003: Subtitles overlapping. + +User johns +Date: Fri Jul 27 19:15:48 CEST 2012 + Free used X11 resources colormap, pixmap, cursor. Fix bug: spelling USE_VAPI wrong, missing functions. @@ -40,7 +40,6 @@ //#define USE_ALSA ///< enable alsa support //#define USE_OSS ///< enable OSS support #define USE_AUDIO_THREAD ///< use thread for audio playback -#define USE_AUDIORING ///< new audio ring code (testing) #include <stdio.h> #include <stdint.h> @@ -106,21 +105,12 @@ typedef struct _audio_module_ const char *Name; ///< audio output module name int (*const Thread) (void); ///< module thread handler -#ifndef USE_AUDIORING - void (*const Enqueue) (const void *, int); ///< enqueue samples for output - void (*const VideoReady) (void); ///< video ready, start audio -#endif void (*const FlushBuffers) (void); ///< flush sample buffers -#ifndef USE_AUDIORING - void (*const Poller) (void); ///< output poller - int (*const FreeBytes) (void); ///< number of bytes free in buffer - int (*const UsedBytes) (void); ///< number of bytes used in buffer -#endif int64_t(*const GetDelay) (void); ///< get current audio delay void (*const SetVolume) (int); ///< set output volume int (*const Setup) (int *, int *, int); ///< setup channels, samplerate - void (*const Play) (void); ///< play - void (*const Pause) (void); ///< pause + void (*const 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; @@ -147,11 +137,6 @@ static volatile char AudioPaused; ///< audio paused static volatile char AudioVideoIsReady; ///< video ready start early static int AudioSkip; ///< skip audio to sync to video -#ifndef USE_AUDIORING -static unsigned AudioSampleRate; ///< audio sample rate in Hz -static unsigned AudioChannels; ///< number of audio channels -static int64_t AudioPTS; ///< audio pts clock -#endif static const int AudioBytesProSample = 2; ///< number of bytes per sample static int AudioBufferTime = 336; ///< audio buffer time in ms @@ -208,8 +193,6 @@ static const unsigned AudioRatesTable[AudioRatesMax] = { 44100, 48000, }; -#ifdef USE_AUDIORING - //---------------------------------------------------------------------------- // filter //---------------------------------------------------------------------------- @@ -633,9 +616,6 @@ typedef struct _audio_ring_ring_ RingBuffer *RingBuffer; ///< sample ring buffer } AudioRingRing; - /// default ring buffer size ~2s 8ch 16bit -//static const unsigned AudioRingBufferSize = 2 * 48000 * 8 * 2; - /// ring of audio ring buffers static AudioRingRing AudioRing[AUDIO_RING_MAX]; static int AudioRingWrite; ///< audio ring write pointer @@ -736,8 +716,6 @@ static void AudioRingExit(void) AudioRingWrite = 0; } -#endif - #ifdef USE_ALSA //============================================================================ @@ -752,23 +730,10 @@ static snd_pcm_t *AlsaPCMHandle; ///< alsa pcm handle static char AlsaCanPause; ///< hw supports pause static int AlsaUseMmap; ///< use mmap -#ifndef USE_AUDIORING - -static RingBuffer *AlsaRingBuffer; ///< audio ring buffer -static unsigned AlsaStartThreshold; ///< start play, if filled - -#ifdef USE_AUDIO_THREAD -static volatile char AlsaFlushBuffer; ///< flag empty buffer -#endif - -#endif - static snd_mixer_t *AlsaMixer; ///< alsa mixer handle static snd_mixer_elem_t *AlsaMixerElem; ///< alsa pcm mixer element static int AlsaRatio; ///< internal -> mixer ratio * 1000 -#ifdef USE_AUDIORING - //---------------------------------------------------------------------------- // alsa pcm //---------------------------------------------------------------------------- @@ -928,394 +893,12 @@ static void AlsaFlushBuffers(void) } } -#else - -//---------------------------------------------------------------------------- -// alsa pcm -//---------------------------------------------------------------------------- - -/** -** Place samples in ringbuffer. -** -** @param samples sample buffer -** @param count number of bytes in sample buffer -** -** @returns true if play should be started. -*/ -static int AlsaAddToRingbuffer(const void *samples, int count) -{ - int n; - - n = RingBufferWrite(AlsaRingBuffer, samples, count); - if (n != count) { - Error(_("audio/alsa: can't place %d samples in ring buffer\n"), count); - // too many bytes are lost - // FIXME: should skip more, longer skip, but less often? - } - - if (!AudioRunning) { - Debug(4, "audio/alsa: start %4zdms\n", - (RingBufferUsedBytes(AlsaRingBuffer) * 1000) - / (AudioSampleRate * AudioChannels * AudioBytesProSample)); - - // forced start - if (AlsaStartThreshold * 2 < RingBufferUsedBytes(AlsaRingBuffer)) { - return 1; - } - // enough video + audio buffered - if (AudioVideoIsReady - && AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) { - // restart play-back - return 1; - } - } - - return 0; -} - -/** -** Play samples from ringbuffer. -*/ -static int AlsaPlayRingbuffer(void) -{ - int first; - int avail; - int n; - int err; - int frames; - const void *p; - - first = 1; - for (;;) { - // how many bytes can be written? - n = snd_pcm_avail_update(AlsaPCMHandle); - if (n < 0) { - if (n == -EAGAIN) { - continue; - } - Error(_("audio/alsa: avail underrun error? '%s'\n"), - snd_strerror(n)); - err = snd_pcm_recover(AlsaPCMHandle, n, 0); - if (err >= 0) { - continue; - } - Error(_("audio/alsa: snd_pcm_avail_update(): %s\n"), - snd_strerror(n)); - return -1; - } - avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, n); - if (avail < 256) { // too much overhead - if (first) { - // happens with broken alsa drivers - if (AudioThread) { - if (!AudioAlsaDriverBroken) { - Error(_("audio/alsa: broken driver %d\n"), avail); - Error("audio/alsa: state %s\n", - snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle))); - } - if (snd_pcm_state(AlsaPCMHandle) - == SND_PCM_STATE_PREPARED) { - if ((err = snd_pcm_start(AlsaPCMHandle)) < 0) { - Error(_("audio/alsa: snd_pcm_start(): %s\n"), - snd_strerror(err)); - } - } - usleep(5 * 1000); - } - } - Debug(4, "audio/alsa: break state %s\n", - snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle))); - break; - } - n = RingBufferGetReadPointer(AlsaRingBuffer, &p); - if (!n) { // ring buffer empty - if (first) { // only error on first loop - Debug(4, "audio/alsa: empty buffers %d\n", avail); - // ring buffer empty - // AlsaLowWaterMark = 1; - return 1; - } - return 0; - } - if (n < avail) { // not enough bytes in ring buffer - avail = n; - } - if (!avail) { // full or buffer empty - break; - } - frames = snd_pcm_bytes_to_frames(AlsaPCMHandle, avail); - - again: - if (AlsaUseMmap) { - err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames); - } else { - err = snd_pcm_writei(AlsaPCMHandle, p, frames); - } - //Debug(3, "audio/alsa: wrote %d/%d frames\n", err, frames); - if (err != frames) { - if (err < 0) { - if (err == -EAGAIN) { - goto again; - } - /* - if (err == -EBADFD) { - goto again; - } - */ - Error(_("audio/alsa: writei underrun error? '%s'\n"), - snd_strerror(err)); - err = snd_pcm_recover(AlsaPCMHandle, err, 0); - if (err >= 0) { - goto again; - } - Error(_("audio/alsa: snd_pcm_writei failed: %s\n"), - snd_strerror(err)); - return -1; - } - // this could happen, if underrun happened - Error(_("audio/alsa: error not all frames written\n")); - avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err); - } - RingBufferReadAdvance(AlsaRingBuffer, avail); - first = 0; - } - - return 0; -} - -/** -** Flush alsa buffers. -*/ -static void AlsaFlushBuffers(void) -{ - int err; - snd_pcm_state_t state; - - if (AlsaRingBuffer && AlsaPCMHandle) { -#ifdef DEBUG - const void *r; - void *w; -#endif - - RingBufferReadAdvance(AlsaRingBuffer, - RingBufferUsedBytes(AlsaRingBuffer)); -#ifdef DEBUG - RingBufferGetWritePointer(AlsaRingBuffer, &w); - RingBufferGetReadPointer(AlsaRingBuffer, &r); - if (r != w) { - Fatal(_("audio/alsa: ringbuffer out of sync %zd-%zd\n"), - RingBufferGetWritePointer(AlsaRingBuffer, &w), - RingBufferGetReadPointer(AlsaRingBuffer, &r)); - abort(); - } -#endif - - state = snd_pcm_state(AlsaPCMHandle); - Debug(3, "audio/alsa: flush state %s\n", snd_pcm_state_name(state)); - if (state != SND_PCM_STATE_OPEN) { - if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) { - Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err)); - } - // ****ing alsa crash, when in open state here - if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) { - Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err)); - } - } - } - AudioRunning = 0; - AudioVideoIsReady = 0; - AudioSkip = 0; - AudioPTS = INT64_C(0x8000000000000000); -} - -/** -** Call back to play audio polled. -*/ -static void AlsaPoller(void) -{ - if (!AlsaPCMHandle) { // setup failure - return; - } - if (!AudioThread && AudioRunning) { - AlsaPlayRingbuffer(); - } -} - -/** -** Get free bytes in audio output. -*/ -static int AlsaFreeBytes(void) -{ - return AlsaRingBuffer ? RingBufferFreeBytes(AlsaRingBuffer) : INT32_MAX; -} - -/** -** Get used bytes in audio output. -*/ -static int AlsaUsedBytes(void) -{ - return AlsaRingBuffer ? RingBufferUsedBytes(AlsaRingBuffer) : 0; -} - -#if 0 - -//---------------------------------------------------------------------------- -// async playback -//---------------------------------------------------------------------------- - -// async playback is broken, don't use it! - -/** -** Alsa async pcm callback function. -** -** @param handler alsa async handler -*/ -static void AlsaAsyncCallback(snd_async_handler_t * handler) -{ - - Debug(3, "audio/%s: %p\n", __FUNCTION__, handler); - - // how many bytes can be written? - for (;;) { - n = snd_pcm_avail_update(AlsaPCMHandle); - if (n < 0) { - Error(_("audio/alsa: snd_pcm_avail_update(): %s\n"), - snd_strerror(n)); - break; - } - avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, n); - if (avail < 512) { // too much overhead - break; - } - - n = RingBufferGetReadPointer(AlsaRingBuffer, &p); - if (!n) { // ring buffer empty - Debug(3, "audio/alsa: ring buffer empty\n"); - break; - } - if (n < avail) { // not enough bytes in ring buffer - avail = n; - } - if (!avail) { // full - break; - } - frames = snd_pcm_bytes_to_frames(AlsaPCMHandle, avail); - - again: - if (AlsaUseMmap) { - err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames); - } else { - err = snd_pcm_writei(AlsaPCMHandle, p, frames); - } - Debug(3, "audio/alsa: %d => %d\n", frames, err); - if (err < 0) { - Error(_("audio/alsa: underrun error?\n")); - err = snd_pcm_recover(AlsaPCMHandle, err, 0); - if (err >= 0) { - goto again; - } - Error(_("audio/alsa: snd_pcm_writei failed: %s\n"), - snd_strerror(err)); - } - if (err != frames) { - Error(_("audio/alsa: error not all frames written\n")); - avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err); - } - RingBufferReadAdvance(AlsaRingBuffer, avail); - } -} - -/** -** Place samples in audio output queue. -** -** @param samples sample buffer -** @param count number of bytes in sample buffer -*/ -static void AlsaEnqueue(const void *samples, int count) -{ - snd_pcm_state_t state; - int n; - - //int err; - - Debug(3, "audio: %6zd + %4d\n", RingBufferUsedBytes(AlsaRingBuffer), - count); - n = RingBufferWrite(AlsaRingBuffer, samples, count); - if (n != count) { - Fatal(_("audio: can't place %d samples in ring buffer\n"), count); - } - // check if running, wait until enough buffered - state = snd_pcm_state(AlsaPCMHandle); - if (state == SND_PCM_STATE_PREPARED) { - Debug(3, "audio/alsa: state %d - %s\n", state, - snd_pcm_state_name(state)); - // FIXME: adjust start ratio - if (RingBufferFreeBytes(AlsaRingBuffer) - < RingBufferUsedBytes(AlsaRingBuffer)) { - // restart play-back -#if 0 - if (AlsaCanPause) { - if ((err = snd_pcm_pause(AlsaPCMHandle, 0))) { - Error(_("audio: snd_pcm_pause(): %s\n"), - snd_strerror(err)); - } - } else { - if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) { - Error(_("audio: snd_pcm_prepare(): %s\n"), - snd_strerror(err)); - } - } - if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) { - Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err)); - } - - Debug(3, "audio/alsa: unpaused\n"); - if ((err = snd_pcm_start(AlsaPCMHandle)) < 0) { - Error(_("audio: snd_pcm_start(): %s\n"), snd_strerror(err)); - } -#endif - state = snd_pcm_state(AlsaPCMHandle); - Debug(3, "audio/alsa: state %s\n", snd_pcm_state_name(state)); - Debug(3, "audio/alsa: unpaused\n"); - } - } -} - -#endif - -//---------------------------------------------------------------------------- -// direct playback -//---------------------------------------------------------------------------- - -// direct play produces underuns on some hardware - -#ifndef USE_AUDIO_THREAD - -/** -** Place samples in audio output queue. -** -** @param samples sample buffer -** @param count number of bytes in sample buffer -*/ -static void AlsaEnqueue(const void *samples, int count) -{ - if (AlsaAddToRingbuffer(samples, count)) { - AudioRunning = 1; - } -} - -#endif - -#endif - #ifdef USE_AUDIO_THREAD //---------------------------------------------------------------------------- // thread playback //---------------------------------------------------------------------------- -#ifdef USE_AUDIORING - /** ** Alsa thread ** @@ -1375,141 +958,6 @@ static int AlsaThread(void) return 1; } -#else - -/** -** Alsa thread -*/ -static int AlsaThread(void) -{ - for (;;) { - int err; - - pthread_testcancel(); - if (AlsaFlushBuffer) { - // we can flush too many, but wo cares - Debug(3, "audio/alsa: flushing buffers\n"); - AlsaFlushBuffers(); - /* - if ((err = snd_pcm_prepare(AlsaPCMHandle))) { - Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err)); - } - */ - AlsaFlushBuffer = 0; - break; - } - if (AudioPaused) { - break; - } - // wait for space in kernel buffers - if ((err = snd_pcm_wait(AlsaPCMHandle, 24)) < 0) { - Error(_("audio/alsa: wait underrun error? '%s'\n"), - snd_strerror(err)); - err = snd_pcm_recover(AlsaPCMHandle, err, 0); - if (err >= 0) { - continue; - } - Error(_("audio/alsa: snd_pcm_wait(): %s\n"), snd_strerror(err)); - usleep(24 * 1000); - continue; - } - // timeout or some commands - if (!err || AlsaFlushBuffer || AudioPaused) { - continue; - } - if ((err = AlsaPlayRingbuffer())) { // empty / error - snd_pcm_state_t state; - - if (err < 0) { // underrun error - break; - } - state = snd_pcm_state(AlsaPCMHandle); - if (state != SND_PCM_STATE_RUNNING) { - Debug(3, "audio/alsa: stopping play '%s'\n", - snd_pcm_state_name(state)); - break; - } - pthread_yield(); - usleep(24 * 1000); // let fill/empty the buffers - } - } - return 0; -} - -/** -** Place samples in audio output queue. -** -** @param samples sample buffer -** @param count number of bytes in sample buffer -*/ -static void AlsaThreadEnqueue(const void *samples, int count) -{ - if (!AlsaRingBuffer || !AlsaPCMHandle) { - Debug(3, "audio/alsa: enqueue not ready\n"); - return; - } - if (AlsaAddToRingbuffer(samples, count)) { - snd_pcm_state_t state; - - state = snd_pcm_state(AlsaPCMHandle); - Debug(3, "audio/alsa: enqueue state %s\n", snd_pcm_state_name(state)); - - // no lock needed, can wakeup next time - AudioRunning = 1; - pthread_cond_signal(&AudioStartCond); - } -} - -/** -** Video is ready, start audio if possible, -*/ -static void AlsaVideoReady(void) -{ - if (!AudioRunning) { - size_t used; - - used = RingBufferUsedBytes(AlsaRingBuffer); - // enough video + audio buffered - if (AlsaStartThreshold < used) { - // too much audio buffered, skip it - if (AlsaStartThreshold * 2 < used) { - Debug(3, "audio/alsa: start %4zdms skip ready\n", - ((used - AlsaStartThreshold * 2) * 1000) - / (AudioSampleRate * AudioChannels * AudioBytesProSample)); - RingBufferReadAdvance(AlsaRingBuffer, - used - AlsaStartThreshold * 2); - } - AudioRunning = 1; - pthread_cond_signal(&AudioStartCond); - } - } - - if (AudioSampleRate && AudioChannels) { - Debug(3, "audio/alsa: start %4zdms video ready\n", - (RingBufferUsedBytes(AlsaRingBuffer) * 1000) - / (AudioSampleRate * AudioChannels * AudioBytesProSample)); - } - -} - -/** -** Flush alsa buffers with thread. -*/ -static void AlsaThreadFlushBuffers(void) -{ - // signal thread to flush buffers - if (AudioThread) { - AlsaFlushBuffer = 1; - do { - AudioRunning = 1; // wakeup in case of sleeping - pthread_cond_signal(&AudioStartCond); - usleep(1 * 1000); - } while (AlsaFlushBuffer); // wait until flushed - } -} - -#endif - #endif //---------------------------------------------------------------------------- @@ -1531,7 +979,7 @@ static snd_pcm_t *AlsaOpenPCM(int use_ac3) && !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) { device = "default"; } - if (!AudioDoingInit) { + if (!AudioDoingInit) { // reduce blabla during init Info(_("audio/alsa: using %sdevice '%s'\n"), use_ac3 ? "ac3 " : "", device); } @@ -1656,8 +1104,6 @@ static void AlsaInitMixer(void) // Alsa API //---------------------------------------------------------------------------- -#ifdef USE_AUDIORING - /** ** Get alsa audio delay in time-stamps. ** @@ -1826,290 +1272,10 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3) return 0; } -#else - -/** -** Get alsa audio delay in time stamps. -** -** @returns audio delay in time stamps. -** -** @todo FIXME: handle the case no audio running -*/ -static int64_t AlsaGetDelay(void) -{ - int err; - snd_pcm_sframes_t delay; - int64_t pts; - - if (!AlsaPCMHandle || !AudioSampleRate) { - return 0L; - } - if (!AudioRunning) { // audio not running - return 0L; - } - // FIXME: thread safe? __assert_fail_base in snd_pcm_delay - - // delay in frames in alsa + kernel buffers - if ((err = snd_pcm_delay(AlsaPCMHandle, &delay)) < 0) { - //Debug(3, "audio/alsa: no hw delay\n"); - delay = 0L; - } else if (snd_pcm_state(AlsaPCMHandle) != SND_PCM_STATE_RUNNING) { - //Debug(3, "audio/alsa: %ld frames delay ok, but not running\n", delay); - } - //Debug(3, "audio/alsa: %ld frames hw delay\n", delay); - - // delay can be negative when underrun occur - if (delay < 0) { - delay = 0L; - } - - pts = ((int64_t) delay * 90 * 1000) / AudioSampleRate; - pts += ((int64_t) RingBufferUsedBytes(AlsaRingBuffer) * 90 * 1000) - / (AudioSampleRate * AudioChannels * AudioBytesProSample); - Debug(4, "audio/alsa: hw+sw delay %zd %" PRId64 "ms\n", - RingBufferUsedBytes(AlsaRingBuffer), pts / 90); - - return pts; -} - -/** -** Setup alsa audio for requested format. -** -** @param freq sample frequency -** @param channels number of channels -** @param use_ac3 use ac3/pass-through device -** -** @retval 0 everything ok -** @retval 1 didn't support frequency/channels combination -** @retval -1 something gone wrong -** -** @todo audio changes must be queued and done when the buffer is empty -*/ -static int AlsaSetup(int *freq, int *channels, int use_ac3) -{ - snd_pcm_uframes_t buffer_size; - snd_pcm_uframes_t period_size; - int err; - int ret; - int delay; - snd_pcm_t *handle; - - if (!AlsaPCMHandle) { // alsa not running yet - return -1; - } -#if 1 // easy alsa hw setup way - // flush any buffered data - AudioFlushBuffers(); - Debug(3, "audio: %dms flush\n", (AudioUsedBytes() * 1000) - / (!AudioSampleRate + !AudioChannels + - AudioSampleRate * AudioChannels * AudioBytesProSample)); - - if (1) { // close+open to fix hdmi no sound bugs - handle = AlsaPCMHandle; - AlsaPCMHandle = NULL; - snd_pcm_close(handle); - if (!(handle = AlsaOpenPCM(use_ac3))) { - return -1; - } - AlsaPCMHandle = handle; - } - - ret = 0; - try_again: - AudioChannels = *channels; - AudioSampleRate = *freq; - - if ((err = - snd_pcm_set_params(AlsaPCMHandle, SND_PCM_FORMAT_S16, - AlsaUseMmap ? SND_PCM_ACCESS_MMAP_INTERLEAVED : - SND_PCM_ACCESS_RW_INTERLEAVED, *channels, *freq, 1, - 96 * 1000))) { - Error(_("audio/alsa: set params error: %s\n"), snd_strerror(err)); - - /* - if ( err == -EBADFD ) { - snd_pcm_close(AlsaPCMHandle); - AlsaPCMHandle = NULL; - goto try_again; - } - */ - - switch (*channels) { - case 1: - // FIXME: enable channel upmix - ret = 1; - *channels = 2; - goto try_again; - case 2: - return -1; - case 3: - case 4: - case 5: - case 6: - case 7: - case 8: - // FIXME: enable channel downmix - // FIXME: try 8 -> 7 -> 6 -> 5 -> 4 -> 3 -> 2 - ret = 1; - *channels = 2; - goto try_again; - default: - Error(_("audio/alsa: unsupported number of channels\n")); - // FIXME: must stop sound, AudioChannels ... invalid - return -1; - } - } -#else - // - // complex way to setup parameters - // - snd_pcm_hw_params_t *hw_params; - int dir; - unsigned buffer_time; - snd_pcm_uframes_t buffer_size; - - snd_pcm_hw_params_alloca(&hw_params); - // choose all parameters - if ((err = snd_pcm_hw_params_any(AlsaPCMHandle, hw_params)) < 0) { - Error(_ - ("audio: snd_pcm_hw_params_any: no configurations available: %s\n"), - snd_strerror(err)); - } - - if ((err = - snd_pcm_hw_params_set_rate_resample(AlsaPCMHandle, hw_params, 1)) - < 0) { - Error(_("audio: can't set rate resample: %s\n"), snd_strerror(err)); - } - if ((err = - snd_pcm_hw_params_set_format(AlsaPCMHandle, hw_params, - SND_PCM_FORMAT_S16)) < 0) { - Error(_("audio: can't set 16-bit: %s\n"), snd_strerror(err)); - } - if ((err = - snd_pcm_hw_params_set_access(AlsaPCMHandle, hw_params, - SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { - Error(_("audio: can't set interleaved read/write %s\n"), - snd_strerror(err)); - } - if ((err = - snd_pcm_hw_params_set_channels(AlsaPCMHandle, hw_params, - channels)) < 0) { - Error(_("audio: can't set channels: %s\n"), snd_strerror(err)); - } - if ((err = - snd_pcm_hw_params_set_rate(AlsaPCMHandle, hw_params, freq, - 0)) < 0) { - Error(_("audio: can't set rate: %s\n"), snd_strerror(err)); - } - // 500000 - // 170667us - buffer_time = 1000 * 1000 * 1000; - dir = 1; -#if 0 - snd_pcm_hw_params_get_buffer_time_max(hw_params, &buffer_time, &dir); - Info(_("audio/alsa: %dus max buffer time\n"), buffer_time); - - buffer_time = 5 * 200 * 1000; // 1s - if ((err = - snd_pcm_hw_params_set_buffer_time_near(AlsaPCMHandle, hw_params, - &buffer_time, &dir)) < 0) { - Error(_("audio: snd_pcm_hw_params_set_buffer_time_near failed: %s\n"), - snd_strerror(err)); - } - Info(_("audio/alsa: %dus buffer time\n"), buffer_time); -#endif - snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size); - Info(_("audio/alsa: buffer size %lu\n"), buffer_size); - buffer_size = buffer_size < 65536 ? buffer_size : 65536; - if ((err = - snd_pcm_hw_params_set_buffer_size_near(AlsaPCMHandle, hw_params, - &buffer_size))) { - Error(_("audio: can't set buffer size: %s\n"), snd_strerror(err)); - } - Info(_("audio/alsa: buffer size %lu\n"), buffer_size); - - if ((err = snd_pcm_hw_params(AlsaPCMHandle, hw_params)) < 0) { - Error(_("audio: snd_pcm_hw_params failed: %s\n"), snd_strerror(err)); - } - // FIXME: use hw_params for buffer_size period_size -#endif - -#if 1 - if (0) { // no underruns allowed, play silence - snd_pcm_sw_params_t *sw_params; - snd_pcm_uframes_t boundary; - - snd_pcm_sw_params_alloca(&sw_params); - err = snd_pcm_sw_params_current(AlsaPCMHandle, sw_params); - if (err < 0) { - Error(_("audio: snd_pcm_sw_params_current failed: %s\n"), - snd_strerror(err)); - } - if ((err = snd_pcm_sw_params_get_boundary(sw_params, &boundary)) < 0) { - Error(_("audio: snd_pcm_sw_params_get_boundary failed: %s\n"), - snd_strerror(err)); - } - Debug(4, "audio/alsa: boundary %lu frames\n", boundary); - if ((err = - snd_pcm_sw_params_set_stop_threshold(AlsaPCMHandle, sw_params, - boundary)) < 0) { - Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"), - snd_strerror(err)); - } - if ((err = - snd_pcm_sw_params_set_silence_size(AlsaPCMHandle, sw_params, - boundary)) < 0) { - Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"), - snd_strerror(err)); - } - if ((err = snd_pcm_sw_params(AlsaPCMHandle, sw_params)) < 0) { - Error(_("audio: snd_pcm_sw_params failed: %s\n"), - snd_strerror(err)); - } - } -#endif - - // update buffer - - snd_pcm_get_params(AlsaPCMHandle, &buffer_size, &period_size); - Debug(3, "audio/alsa: buffer size %lu %zdms, period size %lu %zdms\n", - buffer_size, snd_pcm_frames_to_bytes(AlsaPCMHandle, - buffer_size) * 1000 / (AudioSampleRate * AudioChannels * - AudioBytesProSample), period_size, - snd_pcm_frames_to_bytes(AlsaPCMHandle, - period_size) * 1000 / (AudioSampleRate * AudioChannels * - AudioBytesProSample)); - Debug(3, "audio/alsa: state %s\n", - snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle))); - - AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size); - // buffer time/delay in ms - delay = AudioBufferTime; - if (VideoAudioDelay > 0) { - delay += VideoAudioDelay / 90; - } - if (AlsaStartThreshold < - (*freq * *channels * AudioBytesProSample * delay) / 1000U) { - AlsaStartThreshold = - (*freq * *channels * AudioBytesProSample * delay) / 1000U; - } - // no bigger, than the buffer - if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) { - AlsaStartThreshold = RingBufferFreeBytes(AlsaRingBuffer); - } - Info(_("audio/alsa: delay %ums\n"), (AlsaStartThreshold * 1000) - / (AudioSampleRate * AudioChannels * AudioBytesProSample)); - - return ret; -} - -#endif - /** ** Play audio. */ -void AlsaPlay(void) +static void AlsaPlay(void) { int err; @@ -2132,7 +1298,7 @@ void AlsaPlay(void) /** ** Pause audio. */ -void AlsaPause(void) +static void AlsaPause(void) { int err; @@ -2164,14 +1330,11 @@ static void AlsaNoopCallback( __attribute__ ((unused)) */ static void AlsaInit(void) { -#ifndef DEBUG - // disable display alsa error messages - snd_lib_error_set_handler(AlsaNoopCallback); -#else +#ifdef DEBUG (void)AlsaNoopCallback; -#endif -#ifndef USE_AUDIORING - AlsaRingBuffer = RingBufferNew(AudioRingBufferSize); +#else + // disable display of alsa error messages + snd_lib_error_set_handler(AlsaNoopCallback); #endif AlsaInitPCM(); @@ -2192,13 +1355,6 @@ static void AlsaExit(void) AlsaMixer = NULL; AlsaMixerElem = NULL; } -#ifndef USE_AUDIORING - if (AlsaRingBuffer) { - RingBufferDel(AlsaRingBuffer); - AlsaRingBuffer = NULL; - } - AlsaFlushBuffer = 0; -#endif } /** @@ -2208,25 +1364,8 @@ static const AudioModule AlsaModule = { .Name = "alsa", #ifdef USE_AUDIO_THREAD .Thread = AlsaThread, -#ifdef USE_AUDIORING - //.Enqueue = AlsaThreadEnqueue, - //.VideoReady = AlsaVideoReady, - .FlushBuffers = AlsaFlushBuffers, -#else - .Enqueue = AlsaThreadEnqueue, - .VideoReady = AlsaVideoReady, - .FlushBuffers = AlsaThreadFlushBuffers, #endif -#else - .Enqueue = AlsaEnqueue, - .VideoReady = AlsaVideoReady, .FlushBuffers = AlsaFlushBuffers, -#endif -#ifndef USE_AUDIORING - .Poller = AlsaPoller, - .FreeBytes = AlsaFreeBytes, - .UsedBytes = AlsaUsedBytes, -#endif .GetDelay = AlsaGetDelay, .SetVolume = AlsaSetVolume, .Setup = AlsaSetup, @@ -2253,17 +1392,6 @@ static int OssMixerFildes = -1; ///< mixer file descriptor static int OssMixerChannel; ///< mixer channel index static int OssFragmentTime; ///< fragment time in ms -#ifndef USE_AUDIORING -static RingBuffer *OssRingBuffer; ///< audio ring buffer -static unsigned OssStartThreshold; ///< start play, if filled -#endif - -#ifdef USE_AUDIO_THREAD -static volatile char OssFlushBuffer; ///< flag empty buffer -#endif - -#ifdef USE_AUDIORING - //---------------------------------------------------------------------------- // OSS pcm //---------------------------------------------------------------------------- @@ -2347,194 +1475,12 @@ static void OssFlushBuffers(void) } } -#else - -//---------------------------------------------------------------------------- -// OSS pcm -//---------------------------------------------------------------------------- - -/** -** Place samples in ringbuffer. -** -** @param samples sample buffer -** @param count number of bytes in sample buffer -** -** @returns true if play should be started. -*/ -static int OssAddToRingbuffer(const void *samples, int count) -{ - int n; - - n = RingBufferWrite(OssRingBuffer, samples, count); - if (n != count) { - Error(_("audio/oss: can't place %d samples in ring buffer\n"), count); - // too many bytes are lost - // FIXME: should skip more, longer skip, but less often? - } - - if (!AudioRunning) { - Debug(4, "audio/oss: start %4zdms\n", - (RingBufferUsedBytes(OssRingBuffer) * 1000) - / (AudioSampleRate * AudioChannels * AudioBytesProSample)); - - // forced start - if (OssStartThreshold * 2 < RingBufferUsedBytes(OssRingBuffer)) { - return 1; - } - // enough video + audio buffered - if (AudioVideoIsReady - && OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) { - // restart play-back - return 1; - } - } - - return 0; -} - -/** -** Play samples from ringbuffer. -*/ -static int OssPlayRingbuffer(void) -{ - int first; - const void *p; - - first = 1; - for (;;) { - audio_buf_info bi; - int n; - - if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) { - Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"), - strerror(errno)); - return -1; - } - Debug(4, "audio/oss: %d bytes free\n", bi.bytes); - - n = RingBufferGetReadPointer(OssRingBuffer, &p); - if (!n) { // ring buffer empty - if (first) { // only error on first loop - return 1; - } - return 0; - } - if (n < bi.bytes) { // not enough bytes in ring buffer - bi.bytes = n; - } - if (bi.bytes <= 0) { // full or buffer empty - break; // bi.bytes could become negative! - } - - n = write(OssPcmFildes, p, bi.bytes); - if (n != bi.bytes) { - if (n < 0) { - Error(_("audio/oss: write error: %s\n"), strerror(errno)); - return 1; - } - Warning(_("audio/oss: error not all bytes written\n")); - } - // advance how many could written - RingBufferReadAdvance(OssRingBuffer, n); - first = 0; - } - - return 0; -} - -/** -** Flush OSS buffers. -*/ -static void OssFlushBuffers(void) -{ - if (OssRingBuffer && OssPcmFildes != -1) { - RingBufferReadAdvance(OssRingBuffer, - RingBufferUsedBytes(OssRingBuffer)); - // flush kernel buffers - if (ioctl(OssPcmFildes, SNDCTL_DSP_HALT_OUTPUT, NULL) < 0) { - Error(_("audio/oss: ioctl(SNDCTL_DSP_HALT_OUTPUT): %s\n"), - strerror(errno)); - } - } - AudioRunning = 0; - AudioVideoIsReady = 0; - AudioSkip = 0; - AudioPTS = INT64_C(0x8000000000000000); -} - -//---------------------------------------------------------------------------- -// OSS pcm polled -//---------------------------------------------------------------------------- - -#ifndef USE_AUDIO_THREAD - -/** -** Place samples in audio output queue. -** -** @param samples sample buffer -** @param count number of bytes in sample buffer -*/ -static void OssEnqueue(const void *samples, int count) -{ -#ifdef DEBUG - static uint32_t last_tick; - uint32_t tick; - - tick = GetMsTicks(); - Debug(4, "audio/oss: %4d %dms\n", count, tick - last_tick); - last_tick = tick; -#endif - - if (OssPcmFildes == -1) { // setup failure - Debug(3, "audio/oss: not ready\n"); - return; - } - if (OssAddToRingbuffer(samples, count)) { - AudioRunning = 1; - } -} - -#endif - -/** -** Play all samples possible, without blocking. -*/ -static void OssPoller(void) -{ - if (OssPcmFildes == -1) { // setup failure - return; - } - if (!AudioThread && AudioRunning) { - OssPlayRingbuffer(); - } -} - -/** -** Get free bytes in audio output. -*/ -static int OssFreeBytes(void) -{ - return OssRingBuffer ? RingBufferFreeBytes(OssRingBuffer) : INT32_MAX; -} - -/** -** Get used bytes in audio output. -*/ -static int OssUsedBytes(void) -{ - return OssRingBuffer ? RingBufferUsedBytes(OssRingBuffer) : 0; -} - -#endif - #ifdef USE_AUDIO_THREAD //---------------------------------------------------------------------------- // thread playback //---------------------------------------------------------------------------- -#ifdef USE_AUDIORING - /** ** OSS thread ** @@ -2588,110 +1534,6 @@ static int OssThread(void) return 1; } -#else - -/** -** OSS thread -*/ -static int OssThread(void) -{ - for (;;) { - struct pollfd fds[1]; - int err; - - pthread_testcancel(); - if (OssFlushBuffer) { - // we can flush too many, but wo cares - Debug(3, "audio/oss: flushing buffers\n"); - OssFlushBuffers(); - OssFlushBuffer = 0; - break; - } - if (AudioPaused) { - break; - } - - fds[0].fd = OssPcmFildes; - fds[0].events = POLLOUT | POLLERR; - // wait for space in kernel buffers - err = poll(fds, 1, OssFragmentTime); - if (err < 0) { - Error(_("audio/oss: error poll %s\n"), strerror(errno)); - usleep(OssFragmentTime * 1000); - continue; - } - - if (OssFlushBuffer || AudioPaused) { - continue; - } - - if ((err = OssPlayRingbuffer())) { // empty / error - if (err < 0) { // underrun error - break; - } - pthread_yield(); - usleep(OssFragmentTime * 1000); // let fill/empty the buffers - } - } - return 0; -} - -/** -** Place samples in audio output queue. -** -** @param samples sample buffer -** @param count number of bytes in sample buffer -*/ -static void OssThreadEnqueue(const void *samples, int count) -{ - if (!OssRingBuffer || OssPcmFildes == -1) { - Debug(3, "audio/oss: enqueue not ready\n"); - return; - } - if (OssAddToRingbuffer(samples, count)) { - // no lock needed, can wakeup next time - AudioRunning = 1; - pthread_cond_signal(&AudioStartCond); - } -} - -/** -** Video is ready, start audio if possible, -*/ -static void OssVideoReady(void) -{ - if (AudioSampleRate && AudioChannels) { - Debug(3, "audio/oss: start %4zdms video start\n", - (RingBufferUsedBytes(OssRingBuffer) * 1000) - / (AudioSampleRate * AudioChannels * AudioBytesProSample)); - } - - if (!AudioRunning) { - // enough video + audio buffered - if (OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) { - AudioRunning = 1; - pthread_cond_signal(&AudioStartCond); - } - } -} - -/** -** Flush OSS buffers with thread. -*/ -static void OssThreadFlushBuffers(void) -{ - // signal thread to flush buffers - if (AudioThread) { - OssFlushBuffer = 1; - do { - AudioRunning = 1; // wakeup in case of sleeping - pthread_cond_signal(&AudioStartCond); - usleep(1 * 1000); - } while (OssFlushBuffer); // wait until flushed - } -} - -#endif #endif //---------------------------------------------------------------------------- @@ -2822,8 +1664,6 @@ static void OssInitMixer(void) // OSS API //---------------------------------------------------------------------------- -#ifdef USE_AUDIORING - /** ** Get OSS audio delay in time stamps. ** @@ -2985,183 +1825,10 @@ static int OssSetup(int *sample_rate, int *channels, int use_ac3) return ret; } -#else - -/** -** Get OSS audio delay in time stamps. -** -** @returns audio delay in time stamps. -*/ -static int64_t OssGetDelay(void) -{ - int delay; - int64_t pts; - - if (OssPcmFildes == -1) { // setup failure - return 0L; - } - if (!AudioRunning) { // audio not running - return 0L; - } - // delay in bytes in kernel buffers - delay = -1; - if (ioctl(OssPcmFildes, SNDCTL_DSP_GETODELAY, &delay) == -1) { - Error(_("audio/oss: ioctl(SNDCTL_DSP_GETODELAY): %s\n"), - strerror(errno)); - return 0UL; - } - if (delay < 0) { - delay = 0; - } - - pts = ((int64_t) (delay + RingBufferUsedBytes(OssRingBuffer)) * 90 * 1000) - / (AudioSampleRate * AudioChannels * AudioBytesProSample); - Debug(4, "audio/oss: hw+sw delay %zd %" PRId64 "ms\n", - RingBufferUsedBytes(OssRingBuffer), pts / 90); - - return pts; -} - -/** -** Setup OSS audio for requested format. -** -** @param freq sample frequency -** @param channels number of channels -** @param use_ac3 use ac3/pass-through device -** -** @retval 0 everything ok -** @retval 1 didn't support frequency/channels combination -** @retval -1 something gone wrong -** -** @todo audio changes must be queued and done when the buffer is empty -*/ -static int OssSetup(int *freq, int *channels, int use_ac3) -{ - int ret; - int tmp; - int delay; - audio_buf_info bi; - - if (OssPcmFildes == -1) { // OSS not ready - return -1; - } - // flush any buffered data - AudioFlushBuffers(); - - if (1) { // close+open for pcm / ac3 - int fildes; - - fildes = OssPcmFildes; - OssPcmFildes = -1; - close(fildes); - if (!(fildes = OssOpenPCM(use_ac3))) { - return -1; - } - OssPcmFildes = fildes; - } - - ret = 0; - - tmp = AFMT_S16_NE; // native 16 bits - if (ioctl(OssPcmFildes, SNDCTL_DSP_SETFMT, &tmp) == -1) { - Error(_("audio/oss: ioctl(SNDCTL_DSP_SETFMT): %s\n"), strerror(errno)); - // FIXME: stop player, set setup failed flag - return -1; - } - if (tmp != AFMT_S16_NE) { - Error(_("audio/oss: device doesn't support 16 bit sample format.\n")); - // FIXME: stop player, set setup failed flag - return -1; - } - - tmp = *channels; - if (ioctl(OssPcmFildes, SNDCTL_DSP_CHANNELS, &tmp) == -1) { - Error(_("audio/oss: ioctl(SNDCTL_DSP_CHANNELS): %s\n"), - strerror(errno)); - return -1; - } - if (tmp != *channels) { - Warning(_("audio/oss: device doesn't support %d channels.\n"), - *channels); - *channels = tmp; - ret = 1; - } - - tmp = *freq; - if (ioctl(OssPcmFildes, SNDCTL_DSP_SPEED, &tmp) == -1) { - Error(_("audio/oss: ioctl(SNDCTL_DSP_SPEED): %s\n"), strerror(errno)); - return -1; - } - if (tmp != *freq) { - Warning(_("audio/oss: device doesn't support %dHz sample rate.\n"), - *freq); - *freq = tmp; - ret = 1; - } - - AudioChannels = *channels; - AudioSampleRate = *freq; - - // FIXME: setup buffers - -#ifdef SNDCTL_DSP_POLICY - tmp = 3; - if (ioctl(OssPcmFildes, SNDCTL_DSP_POLICY, &tmp) == -1) { - Error(_("audio/oss: ioctl(SNDCTL_DSP_POLICY): %s\n"), strerror(errno)); - } else { - Info("audio/oss: set policy to %d\n", tmp); - } -#endif - - if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) { - Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"), - strerror(errno)); - bi.fragsize = 4096; - bi.fragstotal = 16; - } else { - Debug(3, "audio/oss: %d bytes buffered\n", bi.bytes); - } - - OssFragmentTime = (bi.fragsize * 1000) - / (AudioSampleRate * AudioChannels * AudioBytesProSample); - - Debug(3, "audio/oss: buffer size %d %dms, fragment size %d %dms\n", - bi.fragsize * bi.fragstotal, (bi.fragsize * bi.fragstotal * 1000) - / (AudioSampleRate * AudioChannels * AudioBytesProSample), bi.fragsize, - OssFragmentTime); - - // start when enough bytes for initial write - OssStartThreshold = (bi.fragsize - 1) * bi.fragstotal; - - // buffer time/delay in ms - delay = AudioBufferTime + 300; - if (VideoAudioDelay > 0) { - delay += VideoAudioDelay / 90; - } - if (OssStartThreshold < - (AudioSampleRate * AudioChannels * AudioBytesProSample * delay) / - 1000U) { - OssStartThreshold = - (AudioSampleRate * AudioChannels * AudioBytesProSample * delay) / - 1000U; - } - // no bigger, than the buffer - if (OssStartThreshold > RingBufferFreeBytes(OssRingBuffer)) { - OssStartThreshold = RingBufferFreeBytes(OssRingBuffer); - } - - Info(_("audio/oss: delay %ums\n"), (OssStartThreshold * 1000) - / (AudioSampleRate * AudioChannels * AudioBytesProSample)); - - return ret; -} - -#endif - /** ** Play audio. */ -void OssPlay(void) +static void OssPlay(void) { } @@ -3177,10 +1844,6 @@ void OssPause(void) */ static void OssInit(void) { -#ifndef USE_AUDIORING - OssRingBuffer = RingBufferNew(AudioRingBufferSize); -#endif - OssInitPCM(); OssInitMixer(); } @@ -3198,7 +1861,6 @@ static void OssExit(void) close(OssMixerFildes); OssMixerFildes = -1; } - OssFlushBuffer = 0; } /** @@ -3208,25 +1870,8 @@ static const AudioModule OssModule = { .Name = "oss", #ifdef USE_AUDIO_THREAD .Thread = OssThread, -#ifdef USE_AUDIORING - //.Enqueue = OssThreadEnqueue, - //.VideoReady = OssVideoReady, - .FlushBuffers = OssFlushBuffers, -#else - .Enqueue = OssThreadEnqueue, - .VideoReady = OssVideoReady, - .FlushBuffers = OssThreadFlushBuffers, #endif -#else - .Enqueue = OssEnqueue, - .VideoReady = OssVideoReady, .FlushBuffers = OssFlushBuffers, -#endif -#ifndef USE_AUDIORING - .Poller = OssPoller, - .FreeBytes = OssFreeBytes, - .UsedBytes = OssUsedBytes, -#endif .GetDelay = OssGetDelay, .SetVolume = OssSetVolume, .Setup = OssSetup, @@ -3242,38 +1887,6 @@ static const AudioModule OssModule = { // Noop //============================================================================ -#ifndef USE_AUDIORING - -/** -** Noop enqueue samples. -** -** @param samples sample buffer -** @param count number of bytes in sample buffer -*/ -static void NoopEnqueue( __attribute__ ((unused)) - const void *samples, __attribute__ ((unused)) - int count) -{ -} - -/** -** Get free bytes in audio output. -*/ -static int NoopFreeBytes(void) -{ - return INT32_MAX; // no driver, much space -} - -/** -** Get used bytes in audio output. -*/ -static int NoopUsedBytes(void) -{ - return 0; // no driver, nothing used -} - -#endif - /** ** Get audio delay in time stamps. ** @@ -3320,16 +1933,7 @@ static void NoopVoid(void) */ static const AudioModule NoopModule = { .Name = "noop", -#ifndef USE_AUDIORING - .Enqueue = NoopEnqueue, - .VideoReady = NoopVoid, -#endif .FlushBuffers = NoopVoid, -#ifndef USE_AUDIORING - .Poller = NoopVoid, - .FreeBytes = NoopFreeBytes, - .UsedBytes = NoopUsedBytes, -#endif .GetDelay = NoopGetDelay, .SetVolume = NoopSetVolume, .Setup = NoopSetup, @@ -3345,8 +1949,6 @@ static const AudioModule NoopModule = { #ifdef USE_AUDIO_THREAD -#ifdef USE_AUDIORING - /** ** Prepare next ring buffer. */ @@ -3483,38 +2085,6 @@ static void *AudioPlayHandlerThread(void *dummy) return dummy; } -#else - -/** -** Audio play thread. -** -** @param dummy unused thread argument -*/ -static void *AudioPlayHandlerThread(void *dummy) -{ - Debug(3, "audio: play thread started\n"); - for (;;) { - Debug(3, "audio: wait on start condition\n"); - pthread_mutex_lock(&AudioMutex); - AudioRunning = 0; - do { - pthread_cond_wait(&AudioStartCond, &AudioMutex); - // cond_wait can return, without signal! - } while (!AudioRunning); - pthread_mutex_unlock(&AudioMutex); - - Debug(3, "audio: ----> %dms start\n", (AudioUsedBytes() * 1000) - / (!AudioSampleRate + !AudioChannels + - AudioSampleRate * AudioChannels * AudioBytesProSample)); - - AudioUsedModule->Thread(); - } - - return dummy; -} - -#endif - /** ** Initialize audio thread. */ @@ -3524,10 +2094,6 @@ static void AudioInitThread(void) pthread_cond_init(&AudioStartCond, NULL); pthread_create(&AudioThread, NULL, AudioPlayHandlerThread, NULL); pthread_setname_np(AudioThread, "softhddev audio"); -#ifndef USE_AUDIORING - pthread_yield(); - usleep(5 * 1000); // give thread some time to start -#endif } /** @@ -3576,7 +2142,6 @@ static const AudioModule *AudioModules[] = { */ void AudioEnqueue(const void *samples, int count) { -#ifdef USE_AUDIORING size_t n; int16_t *buffer; int frames; @@ -3678,34 +2243,6 @@ void AudioEnqueue(const void *samples, int count) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample); } -#else - if (!AudioSampleRate || !AudioChannels) { - return; // not setup - } - if (0) { - static uint32_t last; - static uint32_t tick; - static uint32_t max = 101; - int64_t delay; - - delay = AudioGetDelay(); - tick = GetMsTicks(); - if ((last && tick - last > max) && AudioRunning) { - - //max = tick - last; - Debug(3, "audio: packet delta %d %lu\n", tick - last, delay / 90); - } - last = tick; - } - AudioUsedModule->Enqueue(samples, count); - - // Update audio clock (stupid gcc developers thinks INT64_C is unsigned) - if (AudioPTS != (int64_t) INT64_C(0x8000000000000000)) { - AudioPTS += - ((int64_t) count * 90 * 1000) / (AudioSampleRate * AudioChannels * - AudioBytesProSample); - } -#endif } /** @@ -3715,7 +2252,6 @@ void AudioEnqueue(const void *samples, int count) */ void AudioVideoReady(int64_t pts) { -#ifdef USE_AUDIORING int64_t audio_pts; size_t used; @@ -3826,11 +2362,6 @@ void AudioVideoReady(int64_t pts) } AudioVideoIsReady = 1; #endif -#else - (void)pts; - AudioVideoIsReady = 1; - AudioUsedModule->VideoReady(); -#endif } /** @@ -3838,7 +2369,6 @@ void AudioVideoReady(int64_t pts) */ void AudioFlushBuffers(void) { -#ifdef USE_AUDIORING int old; int i; @@ -3871,9 +2401,6 @@ void AudioFlushBuffers(void) usleep(1 * 1000); // avoid hot polling } Debug(3, "audio: audio flush %dms\n", i); -#else - AudioUsedModule->FlushBuffers(); -#endif } /** @@ -3881,9 +2408,7 @@ void AudioFlushBuffers(void) */ void AudioPoller(void) { -#ifndef USE_AUDIORING - AudioUsedModule->Poller(); -#endif + // FIXME: write poller } /** @@ -3891,13 +2416,9 @@ void AudioPoller(void) */ int AudioFreeBytes(void) { -#ifdef USE_AUDIORING - return AudioRing[AudioRingWrite]. - RingBuffer ? RingBufferFreeBytes(AudioRing[AudioRingWrite]. - RingBuffer) : INT32_MAX; -#else - return AudioUsedModule->FreeBytes(); -#endif + return AudioRing[AudioRingWrite].RingBuffer ? + RingBufferFreeBytes(AudioRing[AudioRingWrite].RingBuffer) + : INT32_MAX; } /** @@ -3905,13 +2426,9 @@ int AudioFreeBytes(void) */ int AudioUsedBytes(void) { -#ifdef USE_AUDIORING - return AudioRing[AudioRingWrite]. - RingBuffer ? RingBufferUsedBytes(AudioRing[AudioRingWrite]. - RingBuffer) : 0; -#else - return AudioUsedModule->UsedBytes(); -#endif + // FIXME: not correct, if multiple buffer are in use + return AudioRing[AudioRingWrite].RingBuffer ? + RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer) : 0; } /** @@ -3921,7 +2438,6 @@ int AudioUsedBytes(void) */ int64_t AudioGetDelay(void) { -#ifdef USE_AUDIORING int64_t pts; if (!AudioRunning) { @@ -3931,19 +2447,16 @@ int64_t AudioGetDelay(void) return 0L; // audio not setup } if (atomic_read(&AudioRingFilled)) { - return 0L; // invalid delay + 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/alsa: hw+sw delay %zd %" PRId64 "ms\n", + Debug(4, "audio: hw+sw delay %zd %" PRId64 "ms\n", RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer), pts / 90); return pts; -#else - return AudioUsedModule->GetDelay(); -#endif } /** @@ -3953,22 +2466,12 @@ int64_t AudioGetDelay(void) */ void AudioSetClock(int64_t pts) { -#ifdef USE_AUDIORING if (AudioRing[AudioRingWrite].PTS != pts) { Debug(4, "audio: set clock %s -> %s pts\n", Timestamp2String(AudioRing[AudioRingWrite].PTS), Timestamp2String(pts)); } AudioRing[AudioRingWrite].PTS = pts; -#else -#ifdef DEBUG - if (AudioPTS != pts) { - Debug(4, "audio: set clock %s -> %s pts\n", Timestamp2String(AudioPTS), - Timestamp2String(pts)); - } -#endif - AudioPTS = pts; -#endif } /** @@ -3978,7 +2481,6 @@ void AudioSetClock(int64_t pts) */ int64_t AudioGetClock(void) { -#ifdef USE_AUDIORING // (cast) needed for the evil gcc if (AudioRing[AudioRingRead].PTS != (int64_t) INT64_C(0x8000000000000000)) { int64_t delay; @@ -3992,17 +2494,6 @@ int64_t AudioGetClock(void) } } return INT64_C(0x8000000000000000); -#else - // (cast) needed for the evil gcc - if (AudioPTS != (int64_t) INT64_C(0x8000000000000000)) { - int64_t delay; - - if ((delay = AudioGetDelay())) { - return AudioPTS - delay; - } - } - return INT64_C(0x8000000000000000); -#endif } /** @@ -4014,7 +2505,6 @@ void AudioSetVolume(int volume) { AudioVolume = volume; AudioMute = !volume; -#ifdef USE_AUDIORING // reduce loudness for stereo output if (AudioStereoDescent && AudioRing[AudioRingRead].InChannels == 2 && !AudioRing[AudioRingRead].UseAc3) { @@ -4025,7 +2515,6 @@ void AudioSetVolume(int volume) volume = 1000; } } -#endif AudioAmplifier = volume; if (!AudioSoftVolume) { AudioUsedModule->SetVolume(volume); @@ -4054,11 +2543,7 @@ int AudioSetup(int *freq, int *channels, int use_ac3) // FIXME: set flag invalid setup return -1; } -#ifdef USE_AUDIORING return AudioRingAdd(*freq, *channels, use_ac3); -#else - return AudioUsedModule->Setup(freq, channels, use_ac3); -#endif } /** @@ -4257,7 +2742,6 @@ void AudioInit(void) found: AudioDoingInit = 1; -#ifdef USE_AUDIORING AudioRingInit(); AudioUsedModule->Init(); // @@ -4355,14 +2839,6 @@ void AudioInit(void) AudioChannelMatrix[u][6], AudioChannelMatrix[u][7], AudioChannelMatrix[u][8]); } -#else - AudioUsedModule->Init(); - freq = 48000; - chan = 2; - if (AudioSetup(&freq, &chan, 0)) { // set default parameters - Error(_("audio: can't do initial setup\n")); - } -#endif #ifdef USE_AUDIO_THREAD if (AudioUsedModule->Thread) { // supports threads AudioInitThread(); @@ -4383,9 +2859,7 @@ void AudioExit(void) #endif AudioUsedModule->Exit(); AudioUsedModule = &NoopModule; -#ifdef USE_AUDIORING AudioRingExit(); -#endif AudioRunning = 0; AudioPaused = 0; } |