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[client] audio: rework audiodevs to be pull model from a common buffer

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This commit is contained in:
Geoffrey McRae
2022-01-18 09:02:44 +11:00
parent aad65c1cab
commit b334f22223
4 changed files with 128 additions and 104 deletions
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89
client/src/audio.c
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@@ -22,6 +22,7 @@
#include "main.h"
#include "common/array.h"
#include "common/util.h"
#include "common/ringbuffer.h"
#include "dynamic/audiodev.h"
@@ -33,10 +34,14 @@ typedef struct
struct
{
bool started;
int volumeChannels;
uint16_t volume[8];
bool mute;
bool setup;
bool started;
int volumeChannels;
uint16_t volume[8];
bool mute;
int sampleRate;
int stride;
RingBuffer buffer;
LG_Lock lock;
RingBuffer timings;
@@ -50,6 +55,7 @@ typedef struct
int volumeChannels;
uint16_t volume[8];
bool mute;
int stride;
uint32_t time;
}
record;
@@ -101,6 +107,18 @@ static const char * audioGraphFormatFn(const char * name,
return title;
}
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);
return frames;
}
void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
uint32_t time)
{
@@ -110,7 +128,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
static int lastChannels = 0;
static int lastSampleRate = 0;
if (audio.playback.started)
if (audio.playback.setup)
{
if (channels != lastChannels || sampleRate != lastSampleRate)
audio.audioDev->playback.stop();
@@ -120,11 +138,18 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
LG_LOCK(audio.playback.lock);
const int bufferFrames = sampleRate / 10;
audio.playback.buffer = ringbuffer_new(bufferFrames,
channels * sizeof(uint16_t));
lastChannels = channels;
lastSampleRate = sampleRate;
audio.playback.started = true;
audio.audioDev->playback.start(channels, sampleRate);
audio.playback.sampleRate = sampleRate;
audio.playback.stride = channels * sizeof(uint16_t);
audio.playback.setup = true;
audio.audioDev->playback.setup(channels, sampleRate, playbackPullFrames);
// if a volume level was stored, set it before we return
if (audio.playback.volumeChannels)
@@ -137,12 +162,9 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
audio.audioDev->playback.mute(audio.playback.mute);
// if the audio dev can report it's latency setup a timing graph
if (audio.audioDev->playback.latency)
{
audio.playback.timings = ringbuffer_new(1200, sizeof(float));
audio.playback.graph = app_registerGraph("PLAYBACK",
audio.playback.timings, 0.0f, 100.0f, audioGraphFormatFn);
}
audio.playback.timings = ringbuffer_new(1200, sizeof(float));
audio.playback.graph = app_registerGraph("PLAYBACK",
audio.playback.timings, 0.0f, 100.0f, audioGraphFormatFn);
LG_UNLOCK(audio.playback.lock);
}
@@ -155,7 +177,9 @@ void audio_playbackStop(void)
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)
{
@@ -176,7 +200,7 @@ 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.started)
if (!audio.playback.setup)
return;
audio.audioDev->playback.volume(channels, volume);
@@ -189,7 +213,7 @@ 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.started)
if (!audio.playback.setup)
return;
audio.audioDev->playback.mute(mute);
@@ -197,10 +221,20 @@ void audio_playbackMute(bool mute)
void audio_playbackData(uint8_t * data, size_t size)
{
if (!audio.audioDev || !audio.playback.started)
if (!audio.audioDev || !audio.playback.setup)
return;
audio.audioDev->playback.play(data, size);
const int frames = size / audio.playback.stride;
ringbuffer_append(audio.playback.buffer, data, frames);
// don't start playback until the buffer is sifficiently full to avoid
// glitches
if (!audio.playback.started && ringbuffer_getCount(audio.playback.buffer) >=
ringbuffer_getLength(audio.playback.buffer) / 4)
{
audio.playback.started = true;
audio.audioDev->playback.start();
}
}
bool audio_supportsRecord(void)
@@ -208,9 +242,9 @@ bool audio_supportsRecord(void)
return audio.audioDev && audio.audioDev->record.start;
}
static void recordData(uint8_t * data, size_t size)
static void recordPushFrames(uint8_t * data, int frames)
{
purespice_writeAudio(data, size, 0);
purespice_writeAudio(data, frames * audio.record.stride, 0);
}
void audio_recordStart(int channels, int sampleRate, PSAudioFormat format)
@@ -232,8 +266,9 @@ void audio_recordStart(int channels, int sampleRate, PSAudioFormat format)
lastChannels = channels;
lastSampleRate = sampleRate;
audio.record.started = true;
audio.record.stride = channels * sizeof(uint16_t);
audio.audioDev->record.start(channels, sampleRate, recordData);
audio.audioDev->record.start(channels, sampleRate, recordPushFrames);
// if a volume level was stored, set it before we return
if (audio.record.volumeChannels)
@@ -287,15 +322,21 @@ void audio_recordMute(bool mute)
void audio_tick(unsigned long long tickCount)
{
LG_LOCK(audio.playback.lock);
if (!audio.playback.timings)
if (!audio.playback.buffer)
{
LG_UNLOCK(audio.playback.lock);
return;
}
const uint64_t latency = audio.audioDev->playback.latency();
const float flatency = latency > 0 ? (float)latency / 1000.0f : 0.0f;
ringbuffer_push(audio.playback.timings, &flatency);
int frames = ringbuffer_getCount(audio.playback.buffer);
if (audio.audioDev->playback.latency)
frames += audio.audioDev->playback.latency();
const float latency = frames > 0
? audio.playback.sampleRate / (float)frames
: 0.0f;
ringbuffer_push(audio.playback.timings, &latency);
LG_UNLOCK(audio.playback.lock);