diff --git a/client/audiodevs/PipeWire/pipewire.c b/client/audiodevs/PipeWire/pipewire.c
index be121e3c..660a8509 100644
--- a/client/audiodevs/PipeWire/pipewire.c
+++ b/client/audiodevs/PipeWire/pipewire.c
@@ -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,
diff --git a/client/audiodevs/PulseAudio/pulseaudio.c b/client/audiodevs/PulseAudio/pulseaudio.c
index 474d9120..1721c9e4 100644
--- a/client/audiodevs/PulseAudio/pulseaudio.c
+++ b/client/audiodevs/PulseAudio/pulseaudio.c
@@ -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
diff --git a/client/include/interface/audiodev.h b/client/include/interface/audiodev.h
index fb2d423b..496e3962 100644
--- a/client/include/interface/audiodev.h
+++ b/client/include/interface/audiodev.h
@@ -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);
diff --git a/client/src/audio.c b/client/src/audio.c
index 53c18845..a9c5844f 100644
--- a/client/src/audio.c
+++ b/client/src/audio.c
@@ -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);