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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
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@@ -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,