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Mix tracks on the producer side of RingBuffer
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@Paul-Licameli
Paul-Licameli committed May 31, 2024
1 parent 980f77c commit 56f20ce
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Showing 2 changed files with 76 additions and 72 deletions.
137 changes: 74 additions & 63 deletions libraries/lib-audio-io/AudioIO.cpp
View file
Original file line number Diff line number Diff line change
Expand Up @@ -2117,11 +2117,7 @@ void AudioIO::ProcessPlaybackSlices(

// Apply per-track gain and pan and muting and micro-fades, all post-effects
ApplyChannelGains();

// For now just generate sufficient silence
auto &pMaster = mMasterBuffers[0];

pMaster->Put(nullptr, floatSample, 0, orig_demand);
MixChannels(orig_demand);
}

bool AudioIO::ConsumeFromMixers(bool paused, const size_t orig_demand,
Expand Down Expand Up @@ -2313,6 +2309,56 @@ void AudioIO::ApplyChannelGains()
}
}

void AudioIO::MixChannels(const size_t demand)
{
if (mPlaybackTracks.empty()) {
// Just generate sufficient silence
for (size_t iChannel = 0;
iChannel < std::min(mNumPlaybackChannels, MaxPlaybackChannels);
++iChannel
)
mMasterBuffers[iChannel]->Put(nullptr, floatSample, 0, demand);
return;
}
size_t totalLength[MaxPlaybackChannels]{};
for (unsigned iBlock : {0, 1}) {
for (size_t iChannel = 0;
iChannel < std::min(mNumPlaybackChannels, MaxPlaybackChannels);
++iChannel)
{
auto &masterBuffer = *mMasterBuffers[iChannel];
bool first = true;
size_t len{};
float *masterSamples{};
for (auto &track : mPlaybackTracks) {
auto &ringBuffer = *track.mBuffers[iChannel];
auto pair = ringBuffer.GetUnflushed(iBlock);
if (first) {
len = pair.second;
totalLength[iChannel] += len;
// Just copy the first track's samples
masterBuffer.Put(pair.first, floatSample, len, 0);
// Find the pointer to the new samples in the master buffer
pair = masterBuffer.GetUnflushed(iBlock);
assert(len == pair.second);
masterSamples = reinterpret_cast<float*>(pair.first);
}
else {
// Expect equally filled buffers
assert(len == pair.second);
// Accumulate (that is, mix) samples
const auto channelSamples = reinterpret_cast<float*>(pair.first);
for (size_t ii = 0; ii < len; ++ii)
masterSamples[ii] += channelSamples[ii];
}
first = false;
}
}
}
assert(std::all_of(std::begin(totalLength), std::end(totalLength),
[&](auto length){ return length == demand; }));
}

void AudioIO::DrainRecordBuffers()
{
if (mRecordingException || mCaptureSequences.empty())
Expand Down Expand Up @@ -2761,23 +2807,6 @@ void AudioIoCallback::CheckSoundActivatedRecordingLevel(
mLastBelow = below;
}

void AudioIoCallback::AddToOutputChannel(unsigned int chan,
float * outputMeterFloats,
float * outputFloats,
const float * tempBuf,
const unsigned long len)
{
const auto numPlaybackChannels = mNumPlaybackChannels;
for (size_t i = 0; i < len; ++i) {
const auto term = tempBuf[i];
outputFloats[numPlaybackChannels * i + chan] += term;
if (outputMeterFloats != outputFloats)
// The level shown in output meters also sums the track level
// microfaded for gain, pan, and mute, but before applying master gain
outputMeterFloats[numPlaybackChannels * i + chan] += term;
}
};

// Limit values to -1.0..+1.0
void ClampBuffer(float * pBuffer, unsigned long len){
for(unsigned i = 0; i < len; i++)
Expand All @@ -2786,63 +2815,43 @@ void ClampBuffer(float * pBuffer, unsigned long len){


void AudioIoCallback::FillOutputBuffers(
float *outputBuffer, size_t framesPerBuffer, size_t toGet,
float *outputMeterFloats)
float *outputBuffer, size_t framesPerBuffer, size_t toGet)
{
const auto numPlaybackChannels = mNumPlaybackChannels;
float *outputFloats = outputBuffer;

// ------ MEMORY ALLOCATION ----------------------
// These are small structures.
const auto tempBufs = stackAllocate(float *, numPlaybackChannels);

// And these are larger structures....
for (unsigned int c = 0; c < numPlaybackChannels; c++)
tempBufs[c] = stackAllocate(float, framesPerBuffer);
// ------ End of MEMORY ALLOCATION ---------------
const auto endOutputFloats =
outputFloats + numPlaybackChannels * framesPerBuffer;

for (auto &track : mPlaybackTracks) {
auto vt = track.mpSequence.get();

decltype(framesPerBuffer) len = 0;

assert(numPlaybackChannels > 0); // pre

// Consume from per-track RingBuffers. For now, just discard.
// In future, per-track data will be forwarded inter-thread for updating
// meter display, synchronizing as closely as possible with the actual
// playback.
for (size_t c = 0; c < numPlaybackChannels; ++c) {
len = track.mBuffers[c]
->Get((samplePtr)tempBufs[c], floatSample, toGet);
len = track.mBuffers[c]->Discard(toGet);
// This should be guaranteed by the producer thread, populating
// the master buffer with the minumum of available lengths
assert(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 SequenceBufferExchange 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*)&tempBufs[c][len], 0,
(framesPerBuffer - len) * sizeof(float));
}
assert(len == toGet);

// Realtime effect transformation of the sound used to happen here
// but it is now done already on the producer side of the RingBuffer

// Our channels aren't silent. We need to pass their data on.
//
// Each channel in the sequences 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 (size_t iChannel = 0;
iChannel < std::min(numPlaybackChannels, MaxPlaybackChannels);
++iChannel)
AddToOutputChannel(iChannel, outputMeterFloats, outputFloats,
tempBufs[iChannel], len);
}
}

mMasterBuffers[0]->Discard(toGet);
// Consume from the master buffers, and interleave
for (size_t c = 0;
c < std::min(MaxPlaybackChannels, mNumPlaybackChannels); ++c) {
const auto buffer = reinterpret_cast<samplePtr>(&outputFloats[c]);
mMasterBuffers[c]->Get(buffer, floatSample, toGet,
numPlaybackChannels /* stride */);
}
// In case the RingBuffer was underfilled
std::fill(outputFloats + toGet * numPlaybackChannels,
endOutputFloats, 0.0f);
}

void AudioIoCallback::ApplyMasterGain(bool paused,
Expand Down Expand Up @@ -3287,8 +3296,10 @@ int AudioIoCallback::AudioCallback(
const bool paused = !UpdateTimePosition(frames);

OtherSynchronization(paused, framesPerBuffer, timeInfo);
FillOutputBuffers(outputBuffer,
framesPerBuffer, frames, outputMeterFloats);
FillOutputBuffers(outputBuffer, framesPerBuffer, frames);
if (outputMeterFloats != outputBuffer)
memcpy(outputMeterFloats, outputBuffer,
mNumPlaybackChannels * frames * sizeof(float));
ApplyMasterGain(paused, outputBuffer, frames);
ClampBuffer(outputBuffer, framesPerBuffer * numPlaybackChannels);
if (outputMeterFloats != outputBuffer)
Expand Down
11 changes: 2 additions & 9 deletions libraries/lib-audio-io/AudioIO.h
View file
Original file line number Diff line number Diff line change
Expand Up @@ -185,21 +185,13 @@ class AUDIO_IO_API AudioIoCallback /* not final */
unsigned long framesPerBuffer
);

void AddToOutputChannel(
unsigned int chan,
float * outputMeterFloats,
float * outputFloats,
const float * tempBuf,
unsigned long len
);
//! Mix and copy to PortAudio's output buffer
//! from our intermediate playback buffers
/*!
@pre mNumPlaybackChannels > 0
*/
void FillOutputBuffers(
float *outputBuffer, size_t framesPerBuffer, size_t toGet,
float *outputMeterFloats);
float *outputBuffer, size_t framesPerBuffer, size_t toGet);
void ApplyMasterGain(bool paused, float *outputFloats, size_t len);
//! @return whether recording is finished
bool DrainInputBuffers(
Expand Down Expand Up @@ -641,6 +633,7 @@ class AUDIO_IO_API AudioIO final
const ChannelGroup *pGroup,
std::unique_ptr<RingBuffer> playbackBuffers[]);
void ApplyChannelGains();
void MixChannels(size_t demand);

//! Second part of SequenceBufferExchange
void DrainRecordBuffers();
Expand Down

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