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Revert "[client] audio: allow the audiodev to return the periodFrames"

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This reverts commit 41884bfcc5.

PipeWire can change it's period size on the fly on us making this
approach invalid.
This commit is contained in:
Geoffrey McRae 2022-01-28 09:58:57 +11:00
parent c2a766c2ee
commit a0477466d2
4 changed files with 49 additions and 69 deletions
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40
client/audiodevs/PipeWire/pipewire.c
View File

@ -104,17 +104,6 @@ static void pipewire_onPlaybackProcess(void * userdata)
if (pw.playback.rateMatch && pw.playback.rateMatch->size > 0)
frames = min(frames, pw.playback.rateMatch->size);
/* pipewire doesn't provide a way to access the quantum, so we start the
* stream and stop it immediately at setup to get this value */
if (pw.playback.startFrames == -1)
{
sbuf->datas[0].chunk->size = 0;
pw_stream_queue_buffer(pw.playback.stream, pbuf);
pw_stream_set_active(pw.playback.stream, false);
pw.playback.startFrames = frames;
return;
}
frames = pw.playback.pullFn(dst, frames);
if (!frames)
{
@ -190,7 +179,7 @@ static void pipewire_playbackStopStream(void)
}
static void pipewire_playbackSetup(int channels, int sampleRate,
LG_AudioPullFn pullFn, int * periodFrames)
LG_AudioPullFn pullFn)
{
const struct spa_pod * params[1];
uint8_t buffer[1024];
@ -220,7 +209,7 @@ static void pipewire_playbackSetup(int channels, int sampleRate,
pw.playback.sampleRate = sampleRate;
pw.playback.stride = sizeof(float) * channels;
pw.playback.pullFn = pullFn;
pw.playback.startFrames = -1;
pw.playback.startFrames = maxLatencyFrames;
pw_thread_loop_lock(pw.thread);
pw.playback.stream = pw_stream_new_simple(
@ -258,22 +247,19 @@ static void pipewire_playbackSetup(int channels, int sampleRate,
PW_ID_ANY,
PW_STREAM_FLAG_AUTOCONNECT |
PW_STREAM_FLAG_MAP_BUFFERS |
PW_STREAM_FLAG_RT_PROCESS,
PW_STREAM_FLAG_RT_PROCESS |
PW_STREAM_FLAG_INACTIVE,
params, 1);
pw_thread_loop_unlock(pw.thread);
/* wait for the stream to start and set this value */
while(pw.playback.startFrames == -1)
pw_thread_loop_wait(pw.thread);
*periodFrames = pw.playback.startFrames;
}
static void pipewire_playbackStart(void)
static bool pipewire_playbackStart(int framesBuffered)
{
if (!pw.playback.stream)
return;
return false;
bool start = false;
if (pw.playback.state != STREAM_STATE_ACTIVE)
{
@ -282,8 +268,12 @@ static void pipewire_playbackStart(void)
switch (pw.playback.state)
{
case STREAM_STATE_INACTIVE:
pw_stream_set_active(pw.playback.stream, true);
pw.playback.state = STREAM_STATE_ACTIVE;
if (framesBuffered >= pw.playback.startFrames)
{
pw_stream_set_active(pw.playback.stream, true);
pw.playback.state = STREAM_STATE_ACTIVE;
start = true;
}
break;
case STREAM_STATE_DRAINING:
@ -297,6 +287,8 @@ static void pipewire_playbackStart(void)
pw_thread_loop_unlock(pw.thread);
}
return start;
}
static void pipewire_playbackStop(void)

14
client/audiodevs/PulseAudio/pulseaudio.c
View File

@ -37,6 +37,7 @@ struct PulseAudio
int sinkIndex;
bool sinkCorked;
bool sinkMuted;
int sinkStart;
int sinkSampleRate;
int sinkChannels;
int sinkStride;
@ -245,7 +246,7 @@ static void pulseaudio_overflow_cb(pa_stream * p, void * userdata)
}
static void pulseaudio_setup(int channels, int sampleRate,
LG_AudioPullFn pullFn, int * periodFrames)
LG_AudioPullFn pullFn)
{
if (pa.sink && pa.sinkChannels == channels && pa.sinkSampleRate == sampleRate)
return;
@ -285,21 +286,26 @@ static void pulseaudio_setup(int channels, int sampleRate,
pa.sinkStride = channels * sizeof(float);
pa.sinkPullFn = pullFn;
pa.sinkStart = attribs.tlength / pa.sinkStride;
pa.sinkCorked = true;
*periodFrames = attribs.tlength / pa.sinkStride;
pa_threaded_mainloop_unlock(pa.loop);
}
static void pulseaudio_start(void)
static bool pulseaudio_start(int framesBuffered)
{
if (!pa.sink)
return;
return false;
if (framesBuffered < pa.sinkStart)
return false;
pa_threaded_mainloop_lock(pa.loop);
pa_stream_cork(pa.sink, 0, NULL, NULL);
pa.sinkCorked = false;
pa_threaded_mainloop_unlock(pa.loop);
return true;
}
static void pulseaudio_stop(void)

8
client/include/interface/audiodev.h
View File

@ -47,11 +47,11 @@ struct LG_AudioDevOps
/* setup the stream for playback but don't start it yet
* Note: the pull function returns f32 samples
*/
void (*setup)(int channels, int sampleRate, LG_AudioPullFn pullFn,
int * periodFrames);
void (*setup)(int channels, int sampleRate, LG_AudioPullFn pullFn);
/* called when there is data available to start playback */
void (*start)();
/* called when there is data available to start playback
* return true if playback should start */
bool (*start)(int framesBuffered);
/* called when SPICE reports the audio stream has stopped */
void (*stop)(void);

56
client/src/audio.c
View File

@ -109,7 +109,6 @@ typedef struct
// avoid false sharing
alignas(64) PlaybackDeviceData deviceData;
alignas(64) PlaybackSpiceData spiceData;
int targetLatencyFrames;
}
playback;
@ -221,16 +220,15 @@ static int playbackPullFrames(uint8_t * dst, int frames)
if (audio.playback.buffer)
{
static bool first = true;
// Measure the device clock and post to the Spice thread
if (frames != data->periodFrames || first)
if (frames != data->periodFrames)
{
if (first)
{
bool init = data->periodFrames == 0;
if (init)
data->nextTime = now;
first = false;
}
data->periodFrames = frames;
data->periodSec = (double) frames / audio.playback.sampleRate;
data->nextTime += llrint(data->periodSec * 1.0e9);
data->nextPosition += frames;
@ -319,6 +317,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
audio.playback.stride = channels * sizeof(float);
audio.playback.state = STREAM_STATE_SETUP;
audio.playback.deviceData.periodFrames = 0;
audio.playback.deviceData.nextPosition = 0;
audio.playback.spiceData.periodFrames = 0;
@ -329,14 +328,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
audio.playback.spiceData.offsetErrorIntegral = 0.0;
audio.playback.spiceData.ratioIntegral = 0.0;
int frames;
audio.audioDev->playback.setup(channels, sampleRate, playbackPullFrames,
&frames);
audio.playback.deviceData.periodFrames = frames;
audio.playback.targetLatencyFrames = frames;
audio.playback.deviceData.periodSec =
(double)frames / audio.playback.sampleRate;
audio.audioDev->playback.setup(channels, sampleRate, playbackPullFrames);
// if a volume level was stored, set it before we return
if (audio.playback.volumeChannels)
@ -405,8 +397,7 @@ void audio_playbackData(uint8_t * data, size_t size)
if (!STREAM_ACTIVE(audio.playback.state))
return;
PlaybackSpiceData * spiceData = &audio.playback.spiceData;
PlaybackDeviceData * devData = &audio.playback.deviceData;
PlaybackSpiceData * spiceData = &audio.playback.spiceData;
int64_t now = nanotime();
// Convert from s16 to f32 samples
@ -455,15 +446,6 @@ void audio_playbackData(uint8_t * data, size_t size)
spiceData->devNextPosition = deviceTick.nextPosition;
}
// If the buffer is getting too empty increase the target latency
static bool checkFill = false;
if (checkFill && audio.playback.state == STREAM_STATE_RUN &&
ringbuffer_getCount(audio.playback.buffer) < devData->periodFrames)
{
audio.playback.targetLatencyFrames += devData->periodFrames;
checkFill = false;
}
// Measure the Spice audio clock
int64_t curTime;
int64_t curPosition;
@ -525,8 +507,17 @@ void audio_playbackData(uint8_t * data, size_t size)
((double) (curTime - spiceData->devLastTime) /
(spiceData->devNextTime - spiceData->devLastTime));
// Target latency derived experimentally to avoid underruns. This could be
// reduced with more tuning. We could adjust on the fly based upon the
// device period size, but that would result in underruns if the period size
// suddenly increases. It may be better instead to just reduce the maximum
// latency on the audio devices, which currently is set quite high
int targetLatencyMs = 70;
int targetLatencyFrames =
targetLatencyMs * audio.playback.sampleRate / 1000;
double actualOffset = curPosition - devPosition;
double actualOffsetError = -(actualOffset - audio.playback.targetLatencyFrames);
double actualOffsetError = -(actualOffset - targetLatencyFrames);
double error = actualOffsetError - offsetError;
spiceData->offsetError += spiceData->b * error +
@ -577,18 +568,9 @@ void audio_playbackData(uint8_t * data, size_t size)
if (audio.playback.state == STREAM_STATE_SETUP)
{
frames = ringbuffer_getCount(audio.playback.buffer);
if (frames >= max(devData->periodFrames,
ringbuffer_getLength(audio.playback.buffer) / 20))
{
if (audio.audioDev->playback.start(frames))
audio.playback.state = STREAM_STATE_RUN;
audio.audioDev->playback.start();
}
}
// re-arm the buffer fill check if we have buffered enough
if (!checkFill && ringbuffer_getCount(audio.playback.buffer) >=
audio.playback.targetLatencyFrames)
checkFill = true;
}
bool audio_supportsRecord(void)