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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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94
client/audiodevs/PipeWire/pipewire.c
View File

@ -26,7 +26,6 @@
#include <math.h>
#include "common/debug.h"
#include "common/ringbuffer.h"
#include "common/stringutils.h"
#include "common/util.h"
@ -50,11 +49,11 @@ struct PipeWire
struct pw_stream * stream;
struct spa_io_rate_match * rateMatch;
int channels;
int sampleRate;
int stride;
int channels;
int sampleRate;
int stride;
LG_AudioPullFn pullFn;
RingBuffer buffer;
StreamState state;
}
playback;
@ -63,10 +62,10 @@ struct PipeWire
{
struct pw_stream * stream;
int channels;
int sampleRate;
int stride;
void (*dataFn)(uint8_t * data, size_t size);
int channels;
int sampleRate;
int stride;
LG_AudioPushFn pushFn;
bool active;
}
@ -90,19 +89,6 @@ static void pipewire_onPlaybackProcess(void * userdata)
{
struct pw_buffer * pbuf;
if (!ringbuffer_getCount(pw.playback.buffer))
{
if (pw.playback.state == STREAM_STATE_FLUSHING)
{
pw_thread_loop_lock(pw.thread);
pw_stream_flush(pw.playback.stream, true);
pw.playback.state = STREAM_STATE_DRAINING;
pw_thread_loop_unlock(pw.thread);
}
return;
}
if (!(pbuf = pw_stream_dequeue_buffer(pw.playback.stream)))
{
DEBUG_WARN("out of buffers");
@ -119,8 +105,24 @@ static void pipewire_onPlaybackProcess(void * userdata)
if (pw.playback.rateMatch && pw.playback.rateMatch->size > 0)
frames = min(frames, pw.playback.rateMatch->size);
void * values = ringbuffer_consume(pw.playback.buffer, &frames);
memcpy(dst, values, frames * pw.playback.stride);
uint8_t * data;
frames = pw.playback.pullFn(&data, frames);
if (!frames)
{
if (pw.playback.state == STREAM_STATE_FLUSHING)
{
pw_thread_loop_lock(pw.thread);
pw_stream_flush(pw.playback.stream, true);
pw.playback.state = STREAM_STATE_DRAINING;
pw_thread_loop_unlock(pw.thread);
}
sbuf->datas[0].chunk->size = 0;
pw_stream_queue_buffer(pw.playback.stream, pbuf);
return;
}
memcpy(dst, data, frames * pw.playback.stride);
sbuf->datas[0].chunk->offset = 0;
sbuf->datas[0].chunk->stride = pw.playback.stride;
@ -197,11 +199,11 @@ static void pipewire_playbackStopStream(void)
pw_stream_destroy(pw.playback.stream);
pw.playback.stream = NULL;
pw.playback.rateMatch = NULL;
ringbuffer_free(&pw.playback.buffer);
pw_thread_loop_unlock(pw.thread);
}
static void pipewire_playbackStart(int channels, int sampleRate)
static void pipewire_playbackSetup(int channels, int sampleRate,
LG_AudioPullFn pullFn)
{
const struct spa_pod * params[1];
uint8_t buffer[1024];
@ -226,8 +228,7 @@ static void pipewire_playbackStart(int channels, int sampleRate)
pw.playback.channels = channels;
pw.playback.sampleRate = sampleRate;
pw.playback.stride = sizeof(uint16_t) * channels;
pw.playback.buffer = ringbuffer_new(bufferFrames,
channels * sizeof(uint16_t));
pw.playback.pullFn = pullFn;
int maxLatencyFrames = bufferFrames / 2;
char maxLatency[32];
@ -277,28 +278,21 @@ static void pipewire_playbackStart(int channels, int sampleRate)
pw_thread_loop_unlock(pw.thread);
}
static void pipewire_playbackPlay(uint8_t * data, size_t size)
static void pipewire_playbackStart(void)
{
if (!pw.playback.stream)
return;
ringbuffer_append(pw.playback.buffer, data, size / pw.playback.stride);
if (pw.playback.state != STREAM_STATE_ACTIVE &&
pw.playback.state != STREAM_STATE_RESTARTING)
{
pw_thread_loop_lock(pw.thread);
switch (pw.playback.state) {
switch (pw.playback.state)
{
case STREAM_STATE_INACTIVE:
// Don't start playback until the buffer is sufficiently full to avoid
// glitches
if (ringbuffer_getCount(pw.playback.buffer) >=
ringbuffer_getLength(pw.playback.buffer) / 4)
{
pw_stream_set_active(pw.playback.stream, true);
pw.playback.state = STREAM_STATE_ACTIVE;
}
pw_stream_set_active(pw.playback.stream, true);
pw.playback.state = STREAM_STATE_ACTIVE;
break;
case STREAM_STATE_FLUSHING:
@ -370,14 +364,9 @@ static void pipewire_playbackMute(bool mute)
pw_thread_loop_unlock(pw.thread);
}
static uint64_t pipewire_playbackLatency(void)
static size_t pipewire_playbackLatency(void)
{
const int frames = ringbuffer_getCount(pw.playback.buffer);
if (frames == 0)
return 0;
// TODO: we should really include the hw latency here too
return (uint64_t)pw.playback.sampleRate * 1000ULL / frames;
return 0;
}
static void pipewire_recordStopStream(void)
@ -408,17 +397,17 @@ static void pipewire_onRecordProcess(void * userdata)
return;
dst += sbuf->datas[0].chunk->offset;
pw.record.dataFn(dst,
pw.record.pushFn(dst,
min(
sbuf->datas[0].chunk->size,
sbuf->datas[0].maxsize - sbuf->datas[0].chunk->offset)
sbuf->datas[0].maxsize - sbuf->datas[0].chunk->offset) / pw.record.stride
);
pw_stream_queue_buffer(pw.record.stream, pbuf);
}
static void pipewire_recordStart(int channels, int sampleRate,
void (*dataFn)(uint8_t * data, size_t size))
LG_AudioPushFn pushFn)
{
const struct spa_pod * params[1];
uint8_t buffer[1024];
@ -439,7 +428,7 @@ static void pipewire_recordStart(int channels, int sampleRate,
pw.record.channels = channels;
pw.record.sampleRate = sampleRate;
pw.record.stride = sizeof(uint16_t) * channels;
pw.record.dataFn = dataFn;
pw.record.pushFn = pushFn;
pw_thread_loop_lock(pw.thread);
pw.record.stream = pw_stream_new_simple(
@ -526,7 +515,6 @@ static void pipewire_free(void)
pw.loop = NULL;
pw.thread = NULL;
ringbuffer_free(&pw.playback.buffer);
pw_deinit();
}
@ -537,8 +525,8 @@ struct LG_AudioDevOps LGAD_PipeWire =
.free = pipewire_free,
.playback =
{
.setup = pipewire_playbackSetup,
.start = pipewire_playbackStart,
.play = pipewire_playbackPlay,
.stop = pipewire_playbackStop,
.volume = pipewire_playbackVolume,
.mute = pipewire_playbackMute,

33
client/audiodevs/PulseAudio/pulseaudio.c
View File

@ -25,7 +25,6 @@
#include <math.h>
#include "common/debug.h"
#include "common/ringbuffer.h"
struct PulseAudio
{
@ -42,7 +41,7 @@ struct PulseAudio
int sinkSampleRate;
int sinkChannels;
int sinkStride;
RingBuffer sinkBuffer;
LG_AudioPullFn sinkPullFn;
};
static struct PulseAudio pa = {0};
@ -221,14 +220,14 @@ static void pulseaudio_free(void)
static void pulseaudio_write_cb(pa_stream * p, size_t nbytes, void * userdata)
{
uint8_t * dst;
uint8_t * dst, * src;
pa_stream_begin_write(p, (void **)&dst, &nbytes);
int frames = nbytes / pa.sinkStride;
void * values = ringbuffer_consume(pa.sinkBuffer, &frames);
int frames = nbytes / pa.sinkStride;
frames = pa.sinkPullFn(&src, frames);
memcpy(dst, values, frames * pa.sinkStride);
memcpy(dst, src, frames * pa.sinkStride);
pa_stream_write(p, dst, frames * pa.sinkStride, NULL, 0, PA_SEEK_RELATIVE);
}
@ -242,7 +241,8 @@ static void pulseaudio_overflow_cb(pa_stream * p, void * userdata)
DEBUG_WARN("Overflow");
}
static void pulseaudio_start(int channels, int sampleRate)
static void pulseaudio_setup(int channels, int sampleRate,
LG_AudioPullFn pullFn)
{
if (pa.sink && pa.sinkChannels == channels && pa.sinkSampleRate == sampleRate)
return;
@ -281,27 +281,22 @@ static void pulseaudio_start(int channels, int sampleRate)
NULL, NULL);
pa.sinkStride = channels * sizeof(uint16_t);
pa.sinkPullFn = pullFn;
pa.sinkStart = attribs.tlength / pa.sinkStride;
pa.sinkBuffer = ringbuffer_new(pa.sinkStart * 2, pa.sinkStride);
pa.sinkCorked = true;
pa_threaded_mainloop_unlock(pa.loop);
}
static void pulseaudio_play(uint8_t * data, size_t size)
static void pulseaudio_start(void)
{
if (!pa.sink)
return;
ringbuffer_append(pa.sinkBuffer, data, size / pa.sinkStride);
if (pa.sinkCorked && ringbuffer_getCount(pa.sinkBuffer) >= pa.sinkStart)
{
pa_threaded_mainloop_lock(pa.loop);
pa_stream_cork(pa.sink, 0, NULL, NULL);
pa.sinkCorked = false;
pa_threaded_mainloop_unlock(pa.loop);
}
pa_threaded_mainloop_lock(pa.loop);
pa_stream_cork(pa.sink, 0, NULL, NULL);
pa.sinkCorked = false;
pa_threaded_mainloop_unlock(pa.loop);
}
static void pulseaudio_stop(void)
@ -348,8 +343,8 @@ struct LG_AudioDevOps LGAD_PulseAudio =
.free = pulseaudio_free,
.playback =
{
.setup = pulseaudio_setup,
.start = pulseaudio_start,
.play = pulseaudio_play,
.stop = pulseaudio_stop,
.volume = pulseaudio_volume,
.mute = pulseaudio_mute

16
client/include/interface/audiodev.h
View File

@ -25,6 +25,9 @@
#include <stdint.h>
#include <stddef.h>
typedef int (*LG_AudioPullFn)(uint8_t ** data, int frames);
typedef void (*LG_AudioPushFn)(uint8_t * data, int frames);
struct LG_AudioDevOps
{
/* internal name of the audio for debugging */
@ -41,15 +44,13 @@ struct LG_AudioDevOps
struct
{
/* start the playback audio stream
/* setup the stream for playback but don't start it yet
* Note: currently SPICE only supports S16 samples so always assume so
*/
void (*start)(int channels, int sampleRate);
void (*setup)(int channels, int sampleRate, LG_AudioPullFn pullFn);
/* called for each packet of output audio to play
* Note: size is the size of data in bytes, not frames/samples
*/
void (*play)(uint8_t * data, size_t size);
/* called when playback is about to start */
void (*start)(void);
/* called when SPICE reports the audio stream has stopped */
void (*stop)(void);
@ -70,8 +71,7 @@ struct LG_AudioDevOps
/* start the record stream
* Note: currently SPICE only supports S16 samples so always assume so
*/
void (*start)(int channels, int sampleRate,
void (*dataFn)(uint8_t * data, size_t size));
void (*start)(int channels, int sampleRate, LG_AudioPushFn pushFn);
/* called when SPICE reports the audio stream has stopped */
void (*stop)(void);

89
client/src/audio.c
View File

@ -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);