Mix.cpp

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/**********************************************************************
 
  Audacity: A Digital Audio Editor
 
  Mix.cpp
 
  Dominic Mazzoni
  Markus Meyer
  Vaughan Johnson
 
*******************************************************************//*******************************************************************/
#include "Mix.h"
#include "MixerSource.h"
 
#include <cmath>
#include "EffectStage.h"
#include "SampleTrack.h"
#include "SampleTrackCache.h"
#include "Resample.h"
#include "float_cast.h"
#include <numeric>
 
namespace {
template<typename T, typename F> std::vector<T>
initVector(size_t dim1, const F &f)
{
   std::vector<T> result( dim1 );
   for (auto &row : result)
      f(row);
   return result;
}
 
template<typename T> std::vector<std::vector<T>>
initVector(size_t dim1, size_t dim2)
{
   return initVector<std::vector<T>>(dim1,
      [dim2](auto &row){ row.resize(dim2); });
}
}
 
namespace {
// Find a block size acceptable to all stages; side-effects on instances
size_t FindBufferSize(const Mixer::Inputs &inputs, size_t bufferSize)
{
   size_t blockSize = bufferSize;
   const auto nTracks = inputs.size();
   for (size_t i = 0; i < nTracks;) {
      const auto &input = inputs[i];
      const auto leader = input.pTrack.get();
      const auto nInChannels = TrackList::Channels(leader).size();
      if (!leader || i + nInChannels > nTracks) {
         assert(false);
         break;
      }
      auto increment = finally([&]{ i += nInChannels; });
      for (const auto &stage : input.stages)
         blockSize = std::min(blockSize, stage.mpInstance->SetBlockSize(blockSize));
   }
   return blockSize;
}
}
 
Mixer::Mixer(Inputs inputs,
   const bool mayThrow,
   const WarpOptions &warpOptions,
   const double startTime, const double stopTime,
   const unsigned numOutChannels,
   const size_t outBufferSize, const bool outInterleaved,
   double outRate, sampleFormat outFormat,
   const bool highQuality, MixerSpec *const mixerSpec,
   const bool applyTrackGains
)  : mNumChannels{ numOutChannels }
   , mInputs{ move(inputs) }
   , mBufferSize{ FindBufferSize(mInputs, outBufferSize) }
   , mApplyTrackGains{ applyTrackGains }
   , mHighQuality{ highQuality }
   , mFormat{ outFormat }
   , mInterleaved{ outInterleaved }
 
   , mTimesAndSpeed{ std::make_shared<TimesAndSpeed>( TimesAndSpeed{
      startTime, stopTime, warpOptions.initialSpeed, startTime
   } ) }
 
   // PRL:  Bug2536: see other comments below for the last, padding argument
   // TODO: more-than-two-channels
   , mFloatBuffers{ 2, mBufferSize, 1, 1 }
 
   // non-interleaved
   , mTemp{ initVector<float>(mNumChannels, mBufferSize) }
   , mBuffer{ initVector<SampleBuffer>(mInterleaved ? 1 : mNumChannels,
      [format = mFormat,
         size = mBufferSize * (mInterleaved ? mNumChannels : 1)
      ](auto &buffer){ buffer.Allocate(size, format); }
   )}
{
   assert(BufferSize() <= outBufferSize);
   const auto nTracks =  mInputs.size();
 
   auto pMixerSpec = ( mixerSpec &&
      mixerSpec->GetNumChannels() == mNumChannels &&
      mixerSpec->GetNumTracks() == nTracks
   ) ? mixerSpec : nullptr;
 
   // Reserve vectors first so we can take safe references to pushed elements
   mSources.reserve(nTracks);
   auto nStages = std::accumulate(mInputs.begin(), mInputs.end(), 0,
      [](auto sum, auto &input){ return sum + input.stages.size(); });
   mSettings.reserve(nStages);
   mStageBuffers.reserve(nStages);
 
   for (size_t i = 0; i < nTracks;) {
      const auto &input = mInputs[i];
      const auto leader = input.pTrack.get();
      const auto nInChannels = TrackList::Channels(leader).size();
      if (!leader || i + nInChannels > nTracks) {
         assert(false);
         break;
      }
      auto increment = finally([&]{ i += nInChannels; });
 
      auto &source = mSources.emplace_back( *leader, BufferSize(), outRate,
         warpOptions, highQuality, mayThrow, mTimesAndSpeed,
         (pMixerSpec ? &pMixerSpec->mMap[i] : nullptr));
      AudioGraph::Source *pDownstream = &source;
      for (const auto &stage : input.stages) {
         // Make a mutable copy of stage.settings
         auto &settings = mSettings.emplace_back(stage.settings);
         // TODO: more-than-two-channels
         // Like mFloatBuffers but padding not needed for soxr
         auto &stageInput = mStageBuffers.emplace_back(2, mBufferSize, 1);
         pDownstream =
         mStages.emplace_back(std::make_unique<AudioGraph::EffectStage>(
            *pDownstream, stageInput,
            *stage.mpInstance, settings, outRate, std::nullopt,
            stage.map
         )).get();
      }
      mDecoratedSources.emplace_back(Source{ source, *pDownstream });
   }
}
 
Mixer::~Mixer()
{
}
 
void Mixer::Clear()
{
   for (auto &buffer: mTemp)
      std::fill(buffer.begin(), buffer.end(), 0);
}
 
static void MixBuffers(unsigned numChannels,
   const unsigned char *channelFlags, const float *gains,
   const float &src, std::vector<std::vector<float>> &dests, int len)
{
   const auto pSrc = &src;
   for (unsigned int c = 0; c < numChannels; c++) {
      if (!channelFlags[c])
         continue;
      float *dest = dests[c].data();
      float gain = gains[c];
      for (int j = 0; j < len; ++j)
         *dest++ += pSrc[j] * gain;   // the actual mixing process
   }
}
 
#define stackAllocate(T, count) static_cast<T*>(alloca(count * sizeof(T)))
 
size_t Mixer::Process(const size_t maxToProcess)
{
   assert(maxToProcess <= BufferSize());
 
   // MB: this is wrong! mT represented warped time, and mTime is too inaccurate to use
   // it here. It's also unnecessary I think.
   //if (mT >= mT1)
   //   return 0;
 
   size_t maxOut = 0;
   const auto channelFlags = stackAllocate(unsigned char, mNumChannels);
   const auto gains = stackAllocate(float, mNumChannels);
   if (!mApplyTrackGains)
      std::fill(gains, gains + mNumChannels, 1.0f);
 
   // Decides which output buffers an input channel accumulates into
   auto findChannelFlags = [&channelFlags, numChannels = mNumChannels]
   (const bool *map, Track::ChannelType channel){
      const auto end = channelFlags + numChannels;
      std::fill(channelFlags, end, 0);
      if (map)
         // ignore left and right when downmixing is customized
         std::copy(map, map + numChannels, channelFlags);
      else switch(channel) {
      case Track::MonoChannel:
      default:
         std::fill(channelFlags, end, 1);
         break;
      case Track::LeftChannel:
         channelFlags[0] = 1;
         break;
      case Track::RightChannel:
         if (numChannels >= 2)
            channelFlags[1] = 1;
         else
            channelFlags[0] = 1;
         break;
      }
      return channelFlags;
   };
 
   auto &[mT0, mT1, _, mTime] = *mTimesAndSpeed;
   auto oldTime = mTime;
   // backwards (as possibly in scrubbing)
   const auto backwards = (mT0 > mT1);
 
   Clear();
   // TODO: more-than-two-channels
   auto maxChannels = mFloatBuffers.Channels();
 
   for (auto &[ upstream, downstream ] : mDecoratedSources) {
      auto oResult = downstream.Acquire(mFloatBuffers, maxToProcess);
      if (!oResult)
         return 0;
      auto result = *oResult;
      maxOut = std::max(maxOut, result);
 
      // Insert effect stages here!  Passing them all channels of the track
 
      const auto limit = std::min<size_t>(upstream.Channels(), maxChannels);
      for (size_t j = 0; j < limit; ++j) {
         const auto pFloat = (const float *)mFloatBuffers.GetReadPosition(j);
         const auto track = upstream.GetChannel(j);
         if (mApplyTrackGains)
            for (size_t c = 0; c < mNumChannels; ++c)
               gains[c] = track->GetChannelGain(c);
         const auto flags =
            findChannelFlags(upstream.MixerSpec(j), track->GetChannel());
         MixBuffers(mNumChannels, flags, gains, *pFloat, mTemp, result);
      }
 
      downstream.Release();
      mFloatBuffers.Advance(result);
      mFloatBuffers.Rotate();
   }
 
   if (backwards)
      mTime = std::clamp(mTime, mT1, oldTime);
   else
      mTime = std::clamp(mTime, oldTime, mT1);
 
   const auto dstStride = (mInterleaved ? mNumChannels : 1);
   for (size_t c = 0; c < mNumChannels; ++c)
      CopySamples((constSamplePtr)mTemp[c].data(), floatSample,
         (mInterleaved
            ? mBuffer[0].ptr() + (c * SAMPLE_SIZE(mFormat))
            : mBuffer[c].ptr()
         ),
         mFormat, maxOut,
         mHighQuality ? gHighQualityDither : gLowQualityDither,
         1, dstStride);
 
   // MB: this doesn't take warping into account, replaced with code based on mSamplePos
   //mT += (maxOut / mRate);
 
   assert(maxOut <= maxToProcess);
   return maxOut;
}
 
constSamplePtr Mixer::GetBuffer()
{
   return mBuffer[0].ptr();
}
 
constSamplePtr Mixer::GetBuffer(int channel)
{
   return mBuffer[channel].ptr();
}
 
double Mixer::MixGetCurrentTime()
{
   return mTimesAndSpeed->mTime;
}
 
#if 0
// Was used before 3.1.0 whenever looping play restarted
// No longer used
void Mixer::Restart()
{
   mTime = mT0;
 
   for(size_t i=0; i<mNumInputTracks; i++)
      mSamplePos[i] = mInputTrack[i].GetTrack()->TimeToLongSamples(mT0);
 
   for(size_t i=0; i<mNumInputTracks; i++) {
      mQueueStart[i] = 0;
      mQueueLen[i] = 0;
   }
 
   // Bug 1887:  libsoxr 0.1.3, first used in Audacity 2.3.0, crashes with
   // constant rate resampling if you try to reuse the resampler after it has
   // flushed.  Should that be considered a bug in sox?  This works around it:
   MakeResamplers();
}
#endif
 
void Mixer::Reposition(double t, bool bSkipping)
{
   auto &[mT0, mT1, _, mTime] = *mTimesAndSpeed;
   mTime = t;
   const bool backwards = (mT1 < mT0);
   if (backwards)
      mTime = std::clamp(mTime, mT1, mT0);
   else
      mTime = std::clamp(mTime, mT0, mT1);
 
   for (auto &source : mSources)
      source.Reposition(mTime, bSkipping);
}
 
void Mixer::SetTimesAndSpeed(double t0, double t1, double speed, bool bSkipping)
{
   wxASSERT(std::isfinite(speed));
   auto &[mT0, mT1, mSpeed, _] = *mTimesAndSpeed;
   mT0 = t0;
   mT1 = t1;
   mSpeed = fabs(speed);
   Reposition(t0, bSkipping);
}
 
void Mixer::SetSpeedForKeyboardScrubbing(double speed, double startTime)
{
   wxASSERT(std::isfinite(speed));
   auto &[mT0, mT1, mSpeed, _] = *mTimesAndSpeed;
 
   // Check if the direction has changed
   if ((speed > 0.0 && mT1 < mT0) || (speed < 0.0 && mT1 > mT0)) {
      // It's safe to use 0 and std::numeric_limits<double>::max(),
      // because Mixer::MixVariableRates() doesn't sample past the start
      // or end of the audio in a track.
      if (speed > 0.0 && mT1 < mT0) {
         mT0 = 0;
         mT1 = std::numeric_limits<double>::max();
      }
      else {
         mT0 = std::numeric_limits<double>::max();
         mT1 = 0;
      }
 
      Reposition(startTime, true);
   }
 
   mSpeed = fabs(speed);
}