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Make AudioIOBufferHelper.h cache friendly.

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Improves performance of project loading substantively.

Signed-off-by: Emily Mabrey <emabrey@tenacityaudio.org>
Helped-by: Alex Disibio <alexdisibio@gmail.com>
This commit is contained in:
Emily Mabrey 2021-07-31 16:51:24 -04:00
parent 8c4278cc64
commit 15c4f546f3
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GPG Key ID: 6F4EF47256A1B7DC
2 changed files with 145 additions and 145 deletions
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247
src/AudioIO.cpp
View File

@ -3843,149 +3843,144 @@ bool AudioIoCallback::FillOutputBuffers(
return true;
}
// ------ MEMORY ALLOCATION ----------------------
std::unique_ptr<AudioIOBufferHelper> bufHelper = std::make_unique<AudioIOBufferHelper>(numPlaybackChannels, framesPerBuffer);
// ------ End of MEMORY ALLOCATION ---------------
auto & em = RealtimeEffectManager::Get();
em.RealtimeProcessStart();
bool selected = false;
int group = 0;
int chanCnt = 0;
// Choose a common size to take from all ring buffers
const auto toGet =
std::min<size_t>(framesPerBuffer, GetCommonlyReadyPlayback());
// The drop and dropQuickly booleans are so named for historical reasons.
// JKC: The original code attempted to be faster by doing nothing on silenced audio.
// This, IMHO, is 'premature optimisation'. Instead clearer and cleaner code would
// simply use a gain of 0.0 for silent audio and go on through to the stage of
// applying that 0.0 gain to the data mixed into the buffer.
// Then (and only then) we would have if needed fast paths for:
// - Applying a uniform gain of 0.0.
// - Applying a uniform gain of 1.0.
// - Applying some other uniform gain.
// - Applying a linearly interpolated gain.
// I would expect us not to need the fast paths, since linearly interpolated gain
// is very cheap to process.
bool drop = false; // Track should become silent.
bool dropQuickly = false; // Track has already been faded to silence.
for (unsigned t = 0; t < numPlaybackTracks; t++)
//Real time process section
{
WaveTrack *vt = mPlaybackTracks[t].get();
bufHelper.get()->chans[chanCnt] = vt;
std::unique_ptr<AudioIOBufferHelper> bufHelper = std::make_unique<AudioIOBufferHelper>(numPlaybackChannels, framesPerBuffer);
auto& em = RealtimeEffectManager::Get();
em.RealtimeProcessStart();
// TODO: more-than-two-channels
auto nextTrack =
t + 1 < numPlaybackTracks
? mPlaybackTracks[t + 1].get()
: nullptr;
bool selected = false;
int group = 0;
int chanCnt = 0;
// First and last channel in this group (for example left and right
// channels of stereo).
bool firstChannel = vt->IsLeader();
bool lastChannel = !nextTrack || nextTrack->IsLeader();
// Choose a common size to take from all ring buffers
const auto toGet = std::min<size_t>(framesPerBuffer, GetCommonlyReadyPlayback());
if ( firstChannel )
{
selected = vt->GetSelected();
// IF mono THEN clear 'the other' channel.
if ( lastChannel && (numPlaybackChannels>1)) {
// TODO: more-than-two-channels
memset(bufHelper.get()->tempBufs[1], 0, framesPerBuffer * sizeof(float));
}
drop = TrackShouldBeSilent( *vt );
dropQuickly = drop;
}
// The drop and dropQuickly booleans are so named for historical reasons.
// JKC: The original code attempted to be faster by doing nothing on silenced audio.
// This, IMHO, is 'premature optimisation'. Instead clearer and cleaner code would
// simply use a gain of 0.0 for silent audio and go on through to the stage of
// applying that 0.0 gain to the data mixed into the buffer.
// Then (and only then) we would have if needed fast paths for:
// - Applying a uniform gain of 0.0.
// - Applying a uniform gain of 1.0.
// - Applying some other uniform gain.
// - Applying a linearly interpolated gain.
// I would expect us not to need the fast paths, since linearly interpolated gain
// is very cheap to process.
if( mbMicroFades )
dropQuickly = dropQuickly && TrackHasBeenFadedOut( *vt );
decltype(framesPerBuffer) len = 0;
bool drop = false; // Track should become silent.
bool dropQuickly = false; // Track has already been faded to silence.
for (unsigned t = 0; t < numPlaybackTracks; t++) {
WaveTrack* vt = mPlaybackTracks[t].get();
bufHelper.get()->chans[chanCnt] = vt;
if (dropQuickly)
{
len = mPlaybackBuffers[t]->Discard(toGet);
// keep going here.
// we may still need to issue a paComplete.
}
else
{
len = mPlaybackBuffers[t]->Get((samplePtr)bufHelper.get()->tempBufs[chanCnt],
floatSample,
toGet);
// wxASSERT( len == toGet );
if (len < framesPerBuffer)
// This used to happen normally at the end of non-looping
// plays, but it can also be an anomalous case where the
// supply from FillBuffers fails to keep up with the
// real-time demand in this thread (see bug 1932). We
// must supply something to the sound card, so pad it with
// zeroes and not random garbage.
memset((void*)&bufHelper.get()->tempBufs[chanCnt][len], 0,
(framesPerBuffer - len) * sizeof(float));
chanCnt++;
}
// TODO: more-than-two-channels
auto nextTrack =
t + 1 < numPlaybackTracks
? mPlaybackTracks[t + 1].get()
: nullptr;
// PRL: Bug1104:
// There can be a difference of len in different loop passes if one channel
// of a stereo track ends before the other! Take a max!
// First and last channel in this group (for example left and right
// channels of stereo).
bool firstChannel = vt->IsLeader();
bool lastChannel = !nextTrack || nextTrack->IsLeader();
// PRL: More recent rewrites of FillBuffers should guarantee a
// padding out of the ring buffers so that equal lengths are
// available, so maxLen ought to increase from 0 only once
mMaxFramesOutput = std::max(mMaxFramesOutput, len);
if (firstChannel) {
selected = vt->GetSelected();
// IF mono THEN clear 'the other' channel.
if (lastChannel && (numPlaybackChannels > 1)) {
// TODO: more-than-two-channels
memset(bufHelper.get()->tempBufs[1], 0, framesPerBuffer * sizeof(float));
}
drop = TrackShouldBeSilent(*vt);
dropQuickly = drop;
}
if ( !lastChannel )
continue;
if (mbMicroFades)
dropQuickly = dropQuickly && TrackHasBeenFadedOut(*vt);
// Last channel of a track seen now
len = mMaxFramesOutput;
decltype(framesPerBuffer) len = 0;
if( !dropQuickly && selected )
len = em.RealtimeProcess(group, chanCnt, bufHelper.get()->tempBufs, len);
group++;
if (dropQuickly) {
len = mPlaybackBuffers[t]->Discard(toGet);
// keep going here.
// we may still need to issue a paComplete.
} else {
len = mPlaybackBuffers[t]->Get((samplePtr)bufHelper.get()->tempBufs[chanCnt],
floatSample,
toGet);
// wxASSERT( len == toGet );
if (len < framesPerBuffer)
// This used to happen normally at the end of non-looping
// plays, but it can also be an anomalous case where the
// supply from FillBuffers fails to keep up with the
// real-time demand in this thread (see bug 1932). We
// must supply something to the sound card, so pad it with
// zeroes and not random garbage.
memset((void*)&bufHelper.get()->tempBufs[chanCnt][len], 0,
(framesPerBuffer - len) * sizeof(float));
chanCnt++;
}
CallbackCheckCompletion(mCallbackReturn, len);
if (dropQuickly) // no samples to process, they've been discarded
continue;
// PRL: Bug1104:
// There can be a difference of len in different loop passes if one channel
// of a stereo track ends before the other! Take a max!
// Our channels aren't silent. We need to pass their data on.
//
// Note that there are two kinds of channel count.
// c and chanCnt are counting channels in the Tracks.
// chan (and numPlayBackChannels) is counting output channels on the device.
// chan = 0 is left channel
// chan = 1 is right channel.
//
// Each channel in the tracks can output to more than one channel on the device.
// For example mono channels output to both left and right output channels.
if (len > 0) for (int c = 0; c < chanCnt; c++)
{
vt = bufHelper.get()->chans[c];
// PRL: More recent rewrites of FillBuffers should guarantee a
// padding out of the ring buffers so that equal lengths are
// available, so maxLen ought to increase from 0 only once
mMaxFramesOutput = std::max(mMaxFramesOutput, len);
if (vt->GetChannelIgnoringPan() == Track::LeftChannel || vt->GetChannelIgnoringPan() == Track::MonoChannel )
AddToOutputChannel( 0, outputMeterFloats, outputFloats, bufHelper.get()->tempBufs[c], drop, len, vt);
if (!lastChannel)
continue;
if (vt->GetChannelIgnoringPan() == Track::RightChannel || vt->GetChannelIgnoringPan() == Track::MonoChannel )
AddToOutputChannel( 1, outputMeterFloats, outputFloats, bufHelper.get()->tempBufs[c], drop, len, vt);
}
// Last channel of a track seen now
len = mMaxFramesOutput;
chanCnt = 0;
if (!dropQuickly && selected)
len = em.RealtimeProcess(group, chanCnt, bufHelper.get()->tempBufs, len);
group++;
CallbackCheckCompletion(mCallbackReturn, len);
if (dropQuickly) // no samples to process, they've been discarded
continue;
// Our channels aren't silent. We need to pass their data on.
//
// Note that there are two kinds of channel count.
// c and chanCnt are counting channels in the Tracks.
// chan (and numPlayBackChannels) is counting output channels on the device.
// chan = 0 is left channel
// chan = 1 is right channel.
//
// Each channel in the tracks can output to more than one channel on the device.
// For example mono channels output to both left and right output channels.
if (len > 0) for (int c = 0; c < chanCnt; c++) {
vt = bufHelper.get()->chans[c];
if (vt->GetChannelIgnoringPan() == Track::LeftChannel || vt->GetChannelIgnoringPan() == Track::MonoChannel)
AddToOutputChannel(0, outputMeterFloats, outputFloats, bufHelper.get()->tempBufs[c], drop, len, vt);
if (vt->GetChannelIgnoringPan() == Track::RightChannel || vt->GetChannelIgnoringPan() == Track::MonoChannel)
AddToOutputChannel(1, outputMeterFloats, outputFloats, bufHelper.get()->tempBufs[c], drop, len, vt);
}
chanCnt = 0;
}
// Poke: If there are no playback tracks, then the earlier check
// about the time indicator being past the end won't happen;
// do it here instead (but not if looping or scrubbing)
if (numPlaybackTracks == 0)
CallbackCheckCompletion(mCallbackReturn, 0);
// wxASSERT( maxLen == toGet );
em.RealtimeProcessEnd();
delete bufHelper.release();
}
// Poke: If there are no playback tracks, then the earlier check
// about the time indicator being past the end won't happen;
// do it here instead (but not if looping or scrubbing)
if (numPlaybackTracks == 0)
CallbackCheckCompletion(mCallbackReturn, 0);
// wxASSERT( maxLen == toGet );
em.RealtimeProcessEnd();
mLastPlaybackTimeMillis = ::wxGetUTCTimeMillis();
ClampBuffer( outputFloats, framesPerBuffer*numPlaybackChannels );

43
src/AudioIOBufferHelper.h
View File

@ -9,34 +9,39 @@ class AudioIOBufferHelper
private:
unsigned int numPlaybackChannels;
unsigned long framesPerBuffer;
unsigned int numPlaybackChannels;
unsigned long framesPerBuffer;
public:
WaveTrack** chans;
float** tempBufs;
WaveTrack** chans;
float** tempBufs;
AudioIOBufferHelper(const unsigned int numPlaybackChannels, const unsigned long framesPerBuffer) {
this->numPlaybackChannels = numPlaybackChannels;
this->framesPerBuffer = framesPerBuffer;
AudioIOBufferHelper(const unsigned int numPlaybackChannels, const unsigned long framesPerBuffer) {
this->numPlaybackChannels = numPlaybackChannels;
this->framesPerBuffer = framesPerBuffer;
this->chans = safenew WaveTrack * [numPlaybackChannels];
this->tempBufs = safenew float* [numPlaybackChannels];
this->chans = safenew WaveTrack * [numPlaybackChannels];
this->tempBufs = safenew float* [numPlaybackChannels];
for (unsigned int c = 0; c < numPlaybackChannels; c++) {
tempBufs[c] = safenew float[framesPerBuffer];
}
tempBufs[0] = safenew float[(size_t)numPlaybackChannels * framesPerBuffer];
for (unsigned int c = 1; c < numPlaybackChannels; c++) {
tempBufs[c] = tempBufs[c - 1] + framesPerBuffer;
}
}
~AudioIOBufferHelper() {
for (unsigned int c = 0; c < numPlaybackChannels; c++) {
delete[] tempBufs[c];
}
~AudioIOBufferHelper() {
delete[] tempBufs;
delete[] tempBufs[0];
delete[] chans;
}
delete[] tempBufs;
tempBufs = nullptr;
delete[] chans;
chans = nullptr;
}
};
#endif