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[client] audio: drain buffers on stop instead of just discarding them
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parent
b334f22223
commit
07c92ec2e8
1 changed files with 62 additions and 37 deletions
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@ -28,20 +28,31 @@
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#include <string.h>
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typedef enum
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{
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STREAM_STATE_STOP,
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STREAM_STATE_SETUP,
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STREAM_STATE_RUN,
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STREAM_STATE_DRAIN
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}
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StreamState;
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#define STREAM_ACTIVE(state) \
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(state == STREAM_STATE_SETUP || state == STREAM_STATE_RUN)
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typedef struct
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{
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struct LG_AudioDevOps * audioDev;
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struct
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{
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bool setup;
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bool started;
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int volumeChannels;
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uint16_t volume[8];
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bool mute;
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int sampleRate;
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int stride;
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RingBuffer buffer;
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StreamState state;
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int volumeChannels;
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uint16_t volume[8];
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bool mute;
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int sampleRate;
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int stride;
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RingBuffer buffer;
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LG_Lock lock;
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RingBuffer timings;
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@ -107,15 +118,34 @@ static const char * audioGraphFormatFn(const char * name,
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return title;
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}
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void playbackStopNL(void)
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{
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audio.playback.state = STREAM_STATE_STOP;
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audio.audioDev->playback.stop();
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ringbuffer_free(&audio.playback.buffer);
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if (audio.playback.timings)
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{
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app_unregisterGraph(audio.playback.graph);
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ringbuffer_free(&audio.playback.timings);
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}
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audio.playback.state = STREAM_STATE_STOP;
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}
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static int playbackPullFrames(uint8_t ** data, int frames)
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{
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LG_LOCK(audio.playback.lock);
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if (audio.playback.buffer)
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*data = ringbuffer_consume(audio.playback.buffer, &frames);
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else
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frames = 0;
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LG_UNLOCK(audio.playback.lock);
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if (audio.playback.state == STREAM_STATE_DRAIN && frames == 0)
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playbackStopNL();
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LG_UNLOCK(audio.playback.lock);
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return frames;
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}
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@ -125,18 +155,18 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
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if (!audio.audioDev)
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return;
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LG_LOCK(audio.playback.lock);
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static int lastChannels = 0;
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static int lastSampleRate = 0;
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if (audio.playback.setup)
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if (STREAM_ACTIVE(audio.playback.state))
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{
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if (channels != lastChannels || sampleRate != lastSampleRate)
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audio.audioDev->playback.stop();
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else
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return;
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}
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if (channels == lastChannels && sampleRate == lastSampleRate)
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goto no_change;
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LG_LOCK(audio.playback.lock);
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playbackStopNL();
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}
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const int bufferFrames = sampleRate / 10;
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audio.playback.buffer = ringbuffer_new(bufferFrames,
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@ -147,7 +177,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
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audio.playback.sampleRate = sampleRate;
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audio.playback.stride = channels * sizeof(uint16_t);
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audio.playback.setup = true;
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audio.playback.state = STREAM_STATE_SETUP;
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audio.audioDev->playback.setup(channels, sampleRate, playbackPullFrames);
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@ -166,28 +196,19 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
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audio.playback.graph = app_registerGraph("PLAYBACK",
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audio.playback.timings, 0.0f, 100.0f, audioGraphFormatFn);
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audio.playback.state = STREAM_STATE_SETUP;
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no_change:
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LG_UNLOCK(audio.playback.lock);
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}
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void audio_playbackStop(void)
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{
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if (!audio.audioDev || !audio.playback.started)
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if (!audio.audioDev || audio.playback.state == STREAM_STATE_STOP)
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return;
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LG_LOCK(audio.playback.lock);
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audio.audioDev->playback.stop();
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audio.playback.setup = false;
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audio.playback.started = false;
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ringbuffer_free(&audio.playback.buffer);
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if (audio.playback.timings)
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{
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app_unregisterGraph(audio.playback.graph);
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ringbuffer_free(&audio.playback.timings);
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}
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LG_UNLOCK(audio.playback.lock);
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audio.playback.state = STREAM_STATE_DRAIN;
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return;
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}
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void audio_playbackVolume(int channels, const uint16_t volume[])
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@ -200,7 +221,7 @@ void audio_playbackVolume(int channels, const uint16_t volume[])
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memcpy(audio.playback.volume, volume, sizeof(uint16_t) * channels);
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audio.playback.volumeChannels = channels;
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if (!audio.playback.setup)
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if (!STREAM_ACTIVE(audio.playback.state))
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return;
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audio.audioDev->playback.volume(channels, volume);
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@ -213,7 +234,7 @@ void audio_playbackMute(bool mute)
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// store the value so we can restore it if the stream is restarted
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audio.playback.mute = mute;
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if (!audio.playback.setup)
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if (!STREAM_ACTIVE(audio.playback.state))
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return;
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audio.audioDev->playback.mute(mute);
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@ -221,7 +242,10 @@ void audio_playbackMute(bool mute)
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void audio_playbackData(uint8_t * data, size_t size)
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{
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if (!audio.audioDev || !audio.playback.setup)
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if (!audio.audioDev)
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return;
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if (!STREAM_ACTIVE(audio.playback.state))
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return;
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const int frames = size / audio.playback.stride;
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@ -229,10 +253,11 @@ void audio_playbackData(uint8_t * data, size_t size)
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// don't start playback until the buffer is sifficiently full to avoid
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// glitches
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if (!audio.playback.started && ringbuffer_getCount(audio.playback.buffer) >=
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if (audio.playback.state == STREAM_STATE_SETUP &&
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ringbuffer_getCount(audio.playback.buffer) >=
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ringbuffer_getLength(audio.playback.buffer) / 4)
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{
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audio.playback.started = true;
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audio.playback.state = STREAM_STATE_RUN;
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audio.audioDev->playback.start();
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}
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}
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