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Split out AudioIO::AllocateBuffers

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This commit is contained in:
Paul Licameli 2018-08-20 09:57:50 -04:00
parent eeb5f1ec20
commit 8e151aba8d
2 changed files with 168 additions and 145 deletions
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303
src/AudioIO.cpp
View File

@ -1995,8 +1995,6 @@ int AudioIO::StartStream(const TransportTracks &tracks,
mCaptureBuffers.reset();
mResample.reset();
double playbackTime = 4.0;
#ifdef EXPERIMENTAL_MIDI_OUT
streamStartTime = 0;
streamStartTime = SystemTime(mUsingAlsa);
@ -2005,32 +2003,6 @@ int AudioIO::StartStream(const TransportTracks &tracks,
mPlaybackSchedule.Init(
t0, t1, options, mCaptureTracks.empty() ? nullptr : &mRecordingSchedule );
const bool scrubbing = mPlaybackSchedule.Interactive();
if (scrubbing)
playbackTime =
lrint(options.pScrubbingOptions->delay * sampleRate) / sampleRate;
//
// The RingBuffer sizes, and the max amount of the buffer to
// fill at a time, both grow linearly with the number of
// tracks. This allows us to scale up to many tracks without
// killing performance.
//
// real playback time to produce with each filling of the buffers
// by the Audio thread (except at the end of playback):
// usually, make fillings fewer and longer for less CPU usage.
// But for useful scrubbing, we can't run too far ahead without checking
// mouse input, so make fillings more and shorter.
// What Audio thread produces for playback is then consumed by the PortAudio
// thread, in many smaller pieces.
wxASSERT( playbackTime >= 0 );
mPlaybackSamplesToCopy = playbackTime * mRate;
// Capacity of the playback buffer.
mPlaybackRingBufferSecs = 10.0;
mCaptureRingBufferSecs = 4.5 + 0.5 * std::min(size_t(16), mCaptureTracks.size());
mMinCaptureSecsToCopy = 0.2 + 0.2 * std::min(size_t(16), mCaptureTracks.size());
unsigned int playbackChannels = 0;
unsigned int captureChannels = 0;
@ -2092,123 +2064,8 @@ int AudioIO::StartStream(const TransportTracks &tracks,
return 0;
}
//
// The (audio) stream has been opened successfully (assuming we tried
// to open it). We now proceed to
// allocate the memory structures the stream will need.
//
bool bDone;
do
{
bDone = true; // assume success
try
{
if( mNumPlaybackChannels > 0 ) {
// Allocate output buffers. For every output track we allocate
// a ring buffer of five seconds
auto playbackBufferSize =
(size_t)lrint(mRate * mPlaybackRingBufferSecs);
auto playbackMixBufferSize =
mPlaybackSamplesToCopy;
mPlaybackBuffers.reinit(mPlaybackTracks.size());
mPlaybackMixers.reinit(mPlaybackTracks.size());
const Mixer::WarpOptions &warpOptions =
#ifdef EXPERIMENTAL_SCRUBBING_SUPPORT
scrubbing
? Mixer::WarpOptions
(ScrubbingOptions::MinAllowedScrubSpeed(),
ScrubbingOptions::MaxAllowedScrubSpeed())
:
#endif
Mixer::WarpOptions(mPlaybackSchedule.mTimeTrack);
for (unsigned int i = 0; i < mPlaybackTracks.size(); i++)
{
// Bug 1763 - We must fade in from zero to avoid a click on starting.
mPlaybackTracks[i]->SetOldChannelGain(0, 0.0);
mPlaybackTracks[i]->SetOldChannelGain(1, 0.0);
mPlaybackBuffers[i] = std::make_unique<RingBuffer>(floatSample, playbackBufferSize);
// use track time for the end time, not real time!
WaveTrackConstArray mixTracks;
mixTracks.push_back(mPlaybackTracks[i]);
double endTime;
if (make_iterator_range(tracks.prerollTracks).contains(mPlaybackTracks[i]))
// Stop playing this track after pre-roll
endTime = t0;
else
// Pass t1 -- not mT1 as may have been adjusted for latency
// -- so that overdub recording stops playing back samples
// at the right time, though transport may continue to record
endTime = t1;
mPlaybackMixers[i] = std::make_unique<Mixer>
(mixTracks,
// Don't throw for read errors, just play silence:
false,
warpOptions,
mPlaybackSchedule.mT0,
endTime,
1,
playbackMixBufferSize, false,
mRate, floatSample, false);
mPlaybackMixers[i]->ApplyTrackGains(false);
}
}
if( mNumCaptureChannels > 0 )
{
// Allocate input buffers. For every input track we allocate
// a ring buffer of five seconds
auto captureBufferSize = (size_t)(mRate * mCaptureRingBufferSecs + 0.5);
// In the extraordinarily rare case that we can't even afford 100 samples, just give up.
if(captureBufferSize < 100)
{
StartStreamCleanup();
AudacityMessageBox(_("Out of memory!"));
return 0;
}
mCaptureBuffers.reinit(mCaptureTracks.size());
mResample.reinit(mCaptureTracks.size());
mFactor = sampleRate / mRate;
for( unsigned int i = 0; i < mCaptureTracks.size(); i++ )
{
mCaptureBuffers[i] = std::make_unique<RingBuffer>
( mCaptureTracks[i]->GetSampleFormat(),
captureBufferSize );
mResample[i] = std::make_unique<Resample>(true, mFactor, mFactor); // constant rate resampling
}
}
}
catch(std::bad_alloc&)
{
// Oops! Ran out of memory. This is pretty rare, so we'll just
// try deleting everything, halving our buffer size, and try again.
StartStreamCleanup(true);
mPlaybackRingBufferSecs *= 0.5;
mPlaybackSamplesToCopy /= 2;
mCaptureRingBufferSecs *= 0.5;
mMinCaptureSecsToCopy *= 0.5;
bDone = false;
// In the extraordinarily rare case that we can't even afford 100 samples, just give up.
auto playbackBufferSize = (size_t)lrint(mRate * mPlaybackRingBufferSecs);
auto playbackMixBufferSize = mPlaybackSamplesToCopy;
if(playbackBufferSize < 100 || playbackMixBufferSize < 100)
{
StartStreamCleanup();
AudacityMessageBox(_("Out of memory!"));
return 0;
}
}
} while(!bDone);
if ( ! AllocateBuffers( options, tracks, t0, t1, sampleRate, scrubbing ) )
return 0;
if (mNumPlaybackChannels > 0)
{
@ -2365,6 +2222,162 @@ int AudioIO::StartStream(const TransportTracks &tracks,
return mStreamToken;
}
bool AudioIO::AllocateBuffers(
const AudioIOStartStreamOptions &options,
const TransportTracks &tracks, double t0, double t1, double sampleRate,
bool scrubbing )
{
//
// The (audio) stream has been opened successfully (assuming we tried
// to open it). We now proceed to
// allocate the memory structures the stream will need.
//
//
// The RingBuffer sizes, and the max amount of the buffer to
// fill at a time, both grow linearly with the number of
// tracks. This allows us to scale up to many tracks without
// killing performance.
//
// real playback time to produce with each filling of the buffers
// by the Audio thread (except at the end of playback):
// usually, make fillings fewer and longer for less CPU usage.
// But for useful scrubbing, we can't run too far ahead without checking
// mouse input, so make fillings more and shorter.
// What Audio thread produces for playback is then consumed by the PortAudio
// thread, in many smaller pieces.
double playbackTime = 4.0;
if (scrubbing)
playbackTime =
lrint(options.pScrubbingOptions->delay * sampleRate) / sampleRate;
wxASSERT( playbackTime >= 0 );
mPlaybackSamplesToCopy = playbackTime * mRate;
// Capacity of the playback buffer.
mPlaybackRingBufferSecs = 10.0;
mCaptureRingBufferSecs = 4.5 + 0.5 * std::min(size_t(16), mCaptureTracks.size());
mMinCaptureSecsToCopy = 0.2 + 0.2 * std::min(size_t(16), mCaptureTracks.size());
bool bDone;
do
{
bDone = true; // assume success
try
{
if( mNumPlaybackChannels > 0 ) {
// Allocate output buffers. For every output track we allocate
// a ring buffer of ten seconds
auto playbackBufferSize =
(size_t)lrint(mRate * mPlaybackRingBufferSecs);
auto playbackMixBufferSize =
mPlaybackSamplesToCopy;
mPlaybackBuffers.reinit(mPlaybackTracks.size());
mPlaybackMixers.reinit(mPlaybackTracks.size());
const Mixer::WarpOptions &warpOptions =
#ifdef EXPERIMENTAL_SCRUBBING_SUPPORT
scrubbing
? Mixer::WarpOptions
(ScrubbingOptions::MinAllowedScrubSpeed(),
ScrubbingOptions::MaxAllowedScrubSpeed())
:
#endif
Mixer::WarpOptions(mPlaybackSchedule.mTimeTrack);
for (unsigned int i = 0; i < mPlaybackTracks.size(); i++)
{
// Bug 1763 - We must fade in from zero to avoid a click on starting.
mPlaybackTracks[i]->SetOldChannelGain(0, 0.0);
mPlaybackTracks[i]->SetOldChannelGain(1, 0.0);
mPlaybackBuffers[i] = std::make_unique<RingBuffer>(floatSample, playbackBufferSize);
// use track time for the end time, not real time!
WaveTrackConstArray mixTracks;
mixTracks.push_back(mPlaybackTracks[i]);
double endTime;
if (make_iterator_range(tracks.prerollTracks).contains(mPlaybackTracks[i]))
// Stop playing this track after pre-roll
endTime = t0;
else
// Pass t1 -- not mT1 as may have been adjusted for latency
// -- so that overdub recording stops playing back samples
// at the right time, though transport may continue to record
endTime = t1;
mPlaybackMixers[i] = std::make_unique<Mixer>
(mixTracks,
// Don't throw for read errors, just play silence:
false,
warpOptions,
mPlaybackSchedule.mT0,
endTime,
1,
playbackMixBufferSize, false,
mRate, floatSample, false);
mPlaybackMixers[i]->ApplyTrackGains(false);
}
}
if( mNumCaptureChannels > 0 )
{
// Allocate input buffers. For every input track we allocate
// a ring buffer of five seconds
auto captureBufferSize = (size_t)(mRate * mCaptureRingBufferSecs + 0.5);
// In the extraordinarily rare case that we can't even afford 100 samples, just give up.
if(captureBufferSize < 100)
{
StartStreamCleanup();
AudacityMessageBox(_("Out of memory!"));
return false;
}
mCaptureBuffers.reinit(mCaptureTracks.size());
mResample.reinit(mCaptureTracks.size());
mFactor = sampleRate / mRate;
for( unsigned int i = 0; i < mCaptureTracks.size(); i++ )
{
mCaptureBuffers[i] = std::make_unique<RingBuffer>
( mCaptureTracks[i]->GetSampleFormat(),
captureBufferSize );
mResample[i] = std::make_unique<Resample>(true, mFactor, mFactor); // constant rate resampling
}
}
}
catch(std::bad_alloc&)
{
// Oops! Ran out of memory. This is pretty rare, so we'll just
// try deleting everything, halving our buffer size, and try again.
StartStreamCleanup(true);
mPlaybackRingBufferSecs *= 0.5;
mPlaybackSamplesToCopy /= 2;
mCaptureRingBufferSecs *= 0.5;
mMinCaptureSecsToCopy *= 0.5;
bDone = false;
// In the extraordinarily rare case that we can't even afford 100 samples, just give up.
auto playbackBufferSize = (size_t)lrint(mRate * mPlaybackRingBufferSecs);
auto playbackMixBufferSize = mPlaybackSamplesToCopy;
if(playbackBufferSize < 100 || playbackMixBufferSize < 100)
{
StartStreamCleanup();
AudacityMessageBox(_("Out of memory!"));
return false;
}
}
} while(!bDone);
return true;
}
void AudioIO::StartStreamCleanup(bool bOnlyBuffers)
{
if (mNumPlaybackChannels > 0)

10
src/AudioIO.h
View File

@ -581,6 +581,16 @@ private:
/** \brief How many sample rates to try */
static const int NumRatesToTry;
/** \brief Allocate RingBuffer structures, and others, needed for playback
* and recording.
*
* Returns true iff successful.
*/
bool AllocateBuffers(
const AudioIOStartStreamOptions &options,
const TransportTracks &tracks, double t0, double t1, double sampleRate,
bool scrubbing );
/** \brief Clean up after StartStream if it fails.
*
* If bOnlyBuffers is specified, it only cleans up the buffers. */