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[client] audio: tune target latency

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The target latency is now based upon the device maximum period size
(which may be configured by setting the `PIPEWIRE_LATENCY` environment
variable if using PipeWire), with some allowance for timing jitter from
Spice and the audio device.

PipeWire can change the period size dynamically at any time which must be
taken into account when selecting the target latency to avoid underruns
when the period size is increased. This is explained in detail within the
commit body.
This commit is contained in:
Chris Spencer 2022-01-29 19:23:28 +00:00 committed by Geoffrey McRae
parent ca29fe80a6
commit e1e60fdaa6
4 changed files with 131 additions and 46 deletions
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73
client/audiodevs/PipeWire/pipewire.c
View File

@ -104,16 +104,6 @@ static void pipewire_onPlaybackProcess(void * userdata)
if (pw.playback.rateMatch && pw.playback.rateMatch->size > 0)
frames = min(frames, pw.playback.rateMatch->size);
// stream was started just to get the initial period size
if (pw.playback.startFrames == -1)
{
pw.playback.startFrames = frames;
pw_stream_set_active(pw.playback.stream, false);
sbuf->datas[0].chunk->size = 0;
pw_stream_queue_buffer(pw.playback.stream, pbuf);
return;
}
frames = pw.playback.pullFn(dst, frames);
if (!frames)
{
@ -189,7 +179,7 @@ static void pipewire_playbackStopStream(void)
}
static void pipewire_playbackSetup(int channels, int sampleRate,
LG_AudioPullFn pullFn)
int * maxPeriodFrames, LG_AudioPullFn pullFn)
{
const struct spa_pod * params[1];
uint8_t buffer[1024];
@ -205,38 +195,71 @@ static void pipewire_playbackSetup(int channels, int sampleRate,
if (pw.playback.stream &&
pw.playback.channels == channels &&
pw.playback.sampleRate == sampleRate)
{
*maxPeriodFrames = pw.playback.startFrames;
return;
}
pipewire_playbackStopStream();
int bufferFrames = sampleRate / 10;
int maxLatencyFrames = bufferFrames / 2;
char maxLatency[32];
snprintf(maxLatency, sizeof(maxLatency), "%d/%d", maxLatencyFrames,
sampleRate);
int defaultLatencyFrames = 2048;
char defaultNodeLatency[32];
snprintf(defaultNodeLatency, sizeof(defaultNodeLatency), "%d/%d",
defaultLatencyFrames, sampleRate);
pw.playback.channels = channels;
pw.playback.sampleRate = sampleRate;
pw.playback.stride = sizeof(float) * channels;
pw.playback.pullFn = pullFn;
pw.playback.startFrames = -1;
pw_thread_loop_lock(pw.thread);
pw.playback.stream = pw_stream_new_simple(
pw.loop,
"Looking Glass",
pw_properties_new(
PW_KEY_NODE_NAME , "Looking Glass",
PW_KEY_MEDIA_TYPE , "Audio",
PW_KEY_MEDIA_CATEGORY , "Playback",
PW_KEY_MEDIA_ROLE , "Music",
PW_KEY_NODE_MAX_LATENCY, maxLatency,
PW_KEY_NODE_NAME , "Looking Glass",
PW_KEY_MEDIA_TYPE , "Audio",
PW_KEY_MEDIA_CATEGORY, "Playback",
PW_KEY_MEDIA_ROLE , "Music",
PW_KEY_NODE_LATENCY , defaultNodeLatency,
NULL
),
&events,
NULL
);
// The user can override the default node latency with the PIPEWIRE_LATENCY
// environment variable, so get the actual node latency value from the stream.
// The actual quantum size may be lower than this value depending on what else
// is using the audio device, but we can treat this value as a maximum
const struct pw_properties * properties =
pw_stream_get_properties(pw.playback.stream);
const char *actualNodeLatency =
pw_properties_get(properties, PW_KEY_NODE_LATENCY);
DEBUG_ASSERT(actualNodeLatency != NULL);
unsigned num, denom;
if (sscanf(actualNodeLatency, "%u/%u", &num, &denom) != 2 ||
denom != sampleRate)
{
DEBUG_WARN(
"PIPEWIRE_LATENCY value '%s' is invalid or does not match stream sample "
"rate; defaulting to %d/%d", actualNodeLatency, defaultLatencyFrames,
sampleRate);
struct spa_dict_item items[] = {
{ PW_KEY_NODE_LATENCY, defaultNodeLatency }
};
pw_stream_update_properties(pw.playback.stream,
&SPA_DICT_INIT_ARRAY(items));
pw.playback.startFrames = defaultLatencyFrames;
}
else
pw.playback.startFrames = num;
*maxPeriodFrames = pw.playback.startFrames;
if (!pw.playback.stream)
{
pw_thread_loop_unlock(pw.thread);
@ -257,13 +280,11 @@ 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);
while(pw.playback.startFrames == -1)
pw_thread_loop_wait(pw.thread);
}
static bool pipewire_playbackStart(int framesBuffered)

7
client/audiodevs/PulseAudio/pulseaudio.c
View File

@ -246,10 +246,13 @@ static void pulseaudio_overflow_cb(pa_stream * p, void * userdata)
}
static void pulseaudio_setup(int channels, int sampleRate,
LG_AudioPullFn pullFn)
int * maxPeriodFrames, LG_AudioPullFn pullFn)
{
if (pa.sink && pa.sinkChannels == channels && pa.sinkSampleRate == sampleRate)
{
*maxPeriodFrames = pa.sinkStart;
return;
}
//TODO: be smarter about this
const int PERIOD_LEN = 80;
@ -289,6 +292,8 @@ static void pulseaudio_setup(int channels, int sampleRate,
pa.sinkStart = attribs.tlength / pa.sinkStride;
pa.sinkCorked = true;
*maxPeriodFrames = pa.sinkStart;
pa_threaded_mainloop_unlock(pa.loop);
}

3
client/include/interface/audiodev.h
View File

@ -47,7 +47,8 @@ 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);
void (*setup)(int channels, int sampleRate, int * maxPeriodFrames,
LG_AudioPullFn pullFn);
/* called when there is data available to start playback
* return true if playback should start */

94
client/src/audio.c
View File

@ -69,9 +69,9 @@ typedef struct
double b;
double c;
int devPeriodFrames;
int64_t devLastTime;
int64_t devNextTime;
int64_t devLastPosition;
int64_t devNextPosition;
@ -97,6 +97,7 @@ typedef struct
int channels;
int sampleRate;
int stride;
int deviceMaxPeriodFrames;
RingBuffer buffer;
RingBuffer deviceTiming;
@ -128,6 +129,7 @@ static AudioState audio = { 0 };
typedef struct
{
int periodFrames;
int64_t nextTime;
int64_t nextPosition;
}
@ -217,13 +219,24 @@ static int playbackPullFrames(uint8_t * dst, int frames)
// Measure the device clock and post to the Spice thread
if (frames != data->periodFrames)
{
double newPeriodSec = (double) frames / audio.playback.sampleRate;
bool init = data->periodFrames == 0;
if (init)
data->nextTime = now;
data->nextTime = now + llrint(newPeriodSec * 1.0e9);
else
// Due to the double-buffered nature of audio playback, we are filling
// in the next buffer while the device is playing the previous buffer.
// This results in slightly unintuitive behaviour when the period size
// changes. The device will request enough samples for the new period
// size, but won't call us again until the previous buffer at the old
// size has finished playing. So, to avoid a blip in the timing
// calculations, we must set the estimated next wakeup time based upon
// the previous period size, not the new one
data->nextTime += llrint(data->periodSec * 1.0e9);
data->periodFrames = frames;
data->periodSec = (double) frames / audio.playback.sampleRate;
data->nextTime += llrint(data->periodSec * 1.0e9);
data->periodSec = newPeriodSec;
data->nextPosition += frames;
double bandwidth = 0.05;
@ -256,6 +269,7 @@ static int playbackPullFrames(uint8_t * dst, int frames)
PlaybackDeviceTick tick =
{
.periodFrames = data->periodFrames,
.nextTime = data->nextTime,
.nextPosition = data->nextPosition
};
@ -317,7 +331,10 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
audio.playback.spiceData.offsetErrorIntegral = 0.0;
audio.playback.spiceData.ratioIntegral = 0.0;
audio.audioDev->playback.setup(channels, sampleRate, playbackPullFrames);
audio.playback.deviceMaxPeriodFrames = 0;
audio.audioDev->playback.setup(channels, sampleRate,
&audio.playback.deviceMaxPeriodFrames, playbackPullFrames);
DEBUG_ASSERT(audio.playback.deviceMaxPeriodFrames > 0);
// if a volume level was stored, set it before we return
if (audio.playback.volumeChannels)
@ -332,7 +349,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
// if the audio dev can report it's latency setup a timing graph
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, 0.0f, 200.0f, audioGraphFormatFn);
audio.playback.state = STREAM_STATE_SETUP;
}
@ -426,6 +443,7 @@ void audio_playbackData(uint8_t * data, size_t size)
PlaybackDeviceTick deviceTick;
while (ringbuffer_consume(audio.playback.deviceTiming, &deviceTick, 1))
{
spiceData->devPeriodFrames = deviceTick.periodFrames;
spiceData->devLastTime = spiceData->devNextTime;
spiceData->devLastPosition = spiceData->devNextPosition;
spiceData->devNextTime = deviceTick.nextTime;
@ -484,6 +502,7 @@ void audio_playbackData(uint8_t * data, size_t size)
// the playback speed to bring them back in line. This value can change
// quite rapidly, particularly at the start of playback, so filter it to
// avoid sudden pitch shifts which will be noticeable to the user.
double actualOffset = 0.0;
double offsetError = spiceData->offsetError;
if (spiceData->devLastTime != INT64_MIN)
{
@ -493,16 +512,56 @@ 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;
// Determine the target latency. Ideally, this would be precisely equal to
// the maximum device period size. However, we need to allow for some timing
// jitter to avoid underruns. Packets from Spice in particular can sometimes
// be delayed by an entire period or more, so include a fixed amount of
// latency to absorb these gaps. For device jitter use a multiplier, so
// timing requirements get progressively stricter as the period size is
// reduced
int spiceJitterMs = 13;
double targetLatencyFrames =
spiceJitterMs * audio.playback.sampleRate / 1000.0 +
audio.playback.deviceMaxPeriodFrames * 1.1;
double actualOffset = curPosition - devPosition;
// If the device is currently at a lower period size than its maximum (which
// can happen, for example, if another application has requested a lower
// latency) then we need to take that into account in our target latency.
//
// The reason to do this is not necessarily obvious, since we already set
// the target latency based upon the maximum period size. The problem stems
// from the way the device changes the period size. When the period size is
// reduced, there will be a transitional period where `playbackPullFrames`
// is invoked with the new smaller period size, but the time until the next
// invocation is based upon the previous size. This happens because the
// device is preparing the next small buffer while still playing back the
// previous large buffer. The result of this is that we end up with a
// surplus of data in the ring buffer. The overall latency is unchanged, but
// the balance has shifted: there is more data in our ring buffer and less
// in the device buffer.
//
// Unaccounted for, this would be detected as an offset error and playback
// would be sped up to bring things back in line. In isolation, this is not
// inherently problematic, and may even be desirable because it would reduce
// the overall latency. The real problem occurs when the period size goes
// back up.
//
// When the period size increases, the exact opposite happens. The device
// will suddenly request data at the new period size, but the timing
// interval will be based upon the previous period size during the
// transition. If there is not enough data to satisfy this then playback
// will start severely underrunning until the timing loop can correct for
// the error.
//
// To counteract this issue, if the current period size is smaller than the
// maximum period size then we increase the target latency by the
// difference. This keeps the offset error stable and ensures we have
// enough data in the buffer to absorb rate increases.
if (spiceData->devPeriodFrames < audio.playback.deviceMaxPeriodFrames)
targetLatencyFrames +=
audio.playback.deviceMaxPeriodFrames - spiceData->devPeriodFrames;
actualOffset = curPosition - devPosition;
double actualOffsetError = -(actualOffset - targetLatencyFrames);
double error = actualOffsetError - offsetError;
@ -558,12 +617,11 @@ void audio_playbackData(uint8_t * data, size_t size)
audio.playback.state = STREAM_STATE_RUN;
}
int latencyFrames = ringbuffer_getCount(audio.playback.buffer);
double latencyFrames = actualOffset;
if (audio.audioDev->playback.latency)
latencyFrames += audio.audioDev->playback.latency();
const float latency = latencyFrames /
(float)(audio.playback.sampleRate / 1000);
const float latency = latencyFrames * 1000.0 / audio.playback.sampleRate;
ringbuffer_push(audio.playback.timings, &latency);
app_invalidateGraph(audio.playback.graph);
}