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[client] audio: drain buffers on stop instead of just discarding them

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This commit is contained in:
Geoffrey McRae 2022-01-18 23:39:05 +11:00
parent b334f22223
commit 07c92ec2e8
1 changed files with 62 additions and 37 deletions
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99
client/src/audio.c
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@ -28,20 +28,31 @@
#include <string.h>
typedef enum
{
STREAM_STATE_STOP,
STREAM_STATE_SETUP,
STREAM_STATE_RUN,
STREAM_STATE_DRAIN
}
StreamState;
#define STREAM_ACTIVE(state) \
(state == STREAM_STATE_SETUP || state == STREAM_STATE_RUN)
typedef struct
{
struct LG_AudioDevOps * audioDev;
struct
{
bool setup;
bool started;
int volumeChannels;
uint16_t volume[8];
bool mute;
int sampleRate;
int stride;
RingBuffer buffer;
StreamState state;
int volumeChannels;
uint16_t volume[8];
bool mute;
int sampleRate;
int stride;
RingBuffer buffer;
LG_Lock lock;
RingBuffer timings;
@ -107,15 +118,34 @@ static const char * audioGraphFormatFn(const char * name,
return title;
}
void playbackStopNL(void)
{
audio.playback.state = STREAM_STATE_STOP;
audio.audioDev->playback.stop();
ringbuffer_free(&audio.playback.buffer);
if (audio.playback.timings)
{
app_unregisterGraph(audio.playback.graph);
ringbuffer_free(&audio.playback.timings);
}
audio.playback.state = STREAM_STATE_STOP;
}
static int playbackPullFrames(uint8_t ** data, int frames)
{
LG_LOCK(audio.playback.lock);
if (audio.playback.buffer)
*data = ringbuffer_consume(audio.playback.buffer, &frames);
else
frames = 0;
LG_UNLOCK(audio.playback.lock);
if (audio.playback.state == STREAM_STATE_DRAIN && frames == 0)
playbackStopNL();
LG_UNLOCK(audio.playback.lock);
return frames;
}
@ -125,18 +155,18 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
if (!audio.audioDev)
return;
LG_LOCK(audio.playback.lock);
static int lastChannels = 0;
static int lastSampleRate = 0;
if (audio.playback.setup)
if (STREAM_ACTIVE(audio.playback.state))
{
if (channels != lastChannels || sampleRate != lastSampleRate)
audio.audioDev->playback.stop();
else
return;
}
if (channels == lastChannels && sampleRate == lastSampleRate)
goto no_change;
LG_LOCK(audio.playback.lock);
playbackStopNL();
}
const int bufferFrames = sampleRate / 10;
audio.playback.buffer = ringbuffer_new(bufferFrames,
@ -147,7 +177,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
audio.playback.sampleRate = sampleRate;
audio.playback.stride = channels * sizeof(uint16_t);
audio.playback.setup = true;
audio.playback.state = STREAM_STATE_SETUP;
audio.audioDev->playback.setup(channels, sampleRate, playbackPullFrames);
@ -166,28 +196,19 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
audio.playback.graph = app_registerGraph("PLAYBACK",
audio.playback.timings, 0.0f, 100.0f, audioGraphFormatFn);
audio.playback.state = STREAM_STATE_SETUP;
no_change:
LG_UNLOCK(audio.playback.lock);
}
void audio_playbackStop(void)
{
if (!audio.audioDev || !audio.playback.started)
if (!audio.audioDev || audio.playback.state == STREAM_STATE_STOP)
return;
LG_LOCK(audio.playback.lock);
audio.audioDev->playback.stop();
audio.playback.setup = false;
audio.playback.started = false;
ringbuffer_free(&audio.playback.buffer);
if (audio.playback.timings)
{
app_unregisterGraph(audio.playback.graph);
ringbuffer_free(&audio.playback.timings);
}
LG_UNLOCK(audio.playback.lock);
audio.playback.state = STREAM_STATE_DRAIN;
return;
}
void audio_playbackVolume(int channels, const uint16_t volume[])
@ -200,7 +221,7 @@ void audio_playbackVolume(int channels, const uint16_t volume[])
memcpy(audio.playback.volume, volume, sizeof(uint16_t) * channels);
audio.playback.volumeChannels = channels;
if (!audio.playback.setup)
if (!STREAM_ACTIVE(audio.playback.state))
return;
audio.audioDev->playback.volume(channels, volume);
@ -213,7 +234,7 @@ void audio_playbackMute(bool mute)
// store the value so we can restore it if the stream is restarted
audio.playback.mute = mute;
if (!audio.playback.setup)
if (!STREAM_ACTIVE(audio.playback.state))
return;
audio.audioDev->playback.mute(mute);
@ -221,7 +242,10 @@ void audio_playbackMute(bool mute)
void audio_playbackData(uint8_t * data, size_t size)
{
if (!audio.audioDev || !audio.playback.setup)
if (!audio.audioDev)
return;
if (!STREAM_ACTIVE(audio.playback.state))
return;
const int frames = size / audio.playback.stride;
@ -229,10 +253,11 @@ void audio_playbackData(uint8_t * data, size_t size)
// don't start playback until the buffer is sifficiently full to avoid
// glitches
if (!audio.playback.started && ringbuffer_getCount(audio.playback.buffer) >=
if (audio.playback.state == STREAM_STATE_SETUP &&
ringbuffer_getCount(audio.playback.buffer) >=
ringbuffer_getLength(audio.playback.buffer) / 4)
{
audio.playback.started = true;
audio.playback.state = STREAM_STATE_RUN;
audio.audioDev->playback.start();
}
}