From 4cfb9301d1b3fb487e68ede6e8910ae52db6a256 Mon Sep 17 00:00:00 2001
From: Paul Licameli <paul.licameli@audacityteam.org>
Date: Wed, 15 May 2024 17:58:14 -0400
Subject: [PATCH] Apply channel gains on the producer side of the RingBuffer

---
 libraries/lib-audio-io/AudioIO.cpp | 60 +++++++++++++++++++-----------
 libraries/lib-audio-io/AudioIO.h   |  7 +---
 2 files changed, 40 insertions(+), 27 deletions(-)

diff --git a/libraries/lib-audio-io/AudioIO.cpp b/libraries/lib-audio-io/AudioIO.cpp
index c53013e64ab3..3e91c3f642b7 100644
--- a/libraries/lib-audio-io/AudioIO.cpp
+++ b/libraries/lib-audio-io/AudioIO.cpp
@@ -2086,12 +2086,19 @@ void AudioIO::TransformPlayBuffers(
    const auto numPlaybackSequences = mPlaybackSequences.size();
    // mPlaybackBuffers correspond many-to-one with mPlaybackSequences
    size_t iBuffer = 0;
+   const auto factor = decayFactor(mRate);
+   const bool silent =
+      mForceFadeOut.load(std::memory_order_relaxed) || IsPaused();
+   size_t iTrack = 0;
    for (const auto vt : mPlaybackSequences) {
+      Finally Do{ [&]{ ++iTrack; } };
       if (!vt)
          continue;
       const auto pGroup = vt->FindChannelGroup();
       if (!pGroup)
          continue;
+      // Check for asynchronous user changes in mute, solo, pause status
+      const bool drop = (silent || SequenceShouldBeSilent(*vt));
 
       // Loop over the blocks of unflushed data, at most two
       size_t discardable = 0;
@@ -2123,6 +2130,33 @@ void AudioIO::TransformPlayBuffers(
          auto discarded = ringBuffer.Unput(discardable);
          // assert(discarded == discardable);
       }
+      // Iterate the two blocks again, only after latency has been shifted out,
+      // which maybe changes the extent of those blocks in memory, and apply
+      // the per-track pan, mute, gain, and solo, with microfading
+      for (unsigned iBlock : {0, 1}) {
+         size_t len = 0;
+         for (size_t iChannel = 0; iChannel < mNumPlaybackChannels; ++iChannel) {
+            const float goal = (drop ? 0.0f : vt->GetChannelGain(iChannel));
+            auto &laggingChannelGain = mOldChannelGains[iTrack][iChannel];
+            // if no microfades, jump in volume.
+            float diff = (mbMicroFades ? goal - laggingChannelGain : 0.0f);
+            auto &ringBuffer = *mPlaybackBuffers[iBuffer + iChannel];
+            const auto pair = ringBuffer.GetUnflushed(iBlock);
+            // Playback RingBuffers have float format: see AllocateBuffers
+            auto pFloats = reinterpret_cast<float*>(pair.first);
+            // The lengths of corresponding unflushed blocks should be
+            // the same for all channels
+            if (len == 0)
+               len = pair.second;
+            else
+               assert(len == pair.second);
+            for (size_t ii = 0; ii < len; ++ii) {
+               pFloats[ii] *= (goal - diff);
+               diff *= factor;
+            }
+            laggingChannelGain = goal - diff;
+         }
+      }
       iBuffer += mNumPlaybackChannels;
    }
 }
@@ -2582,31 +2616,17 @@ void AudioIoCallback::AddToOutputChannel(unsigned int chan,
    float * outputMeterFloats,
    float * outputFloats,
    const float * tempBuf,
-   bool drop,
-   const unsigned long len,
-   const PlayableSequence &ps,
-   float &laggingChannelGain)
+   const unsigned long len)
 {
    const auto numPlaybackChannels = mNumPlaybackChannels;
-   const auto factor = decayFactor(mRate);
-
-   float gain = ps.GetChannelGain(chan);
-   if (drop || mForceFadeOut.load(std::memory_order_relaxed) || IsPaused())
-      gain = 0.0;
-
-   const float goal = gain;
-   // if no microfades, jump in volume.
-   float diff = (mbMicroFades ? goal - laggingChannelGain : 0.0f);
    for (size_t i = 0; i < len; ++i) {
-      const auto term = (goal - diff) * tempBuf[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;
-      diff *= factor;
    }
-   laggingChannelGain = goal - diff;
 };
 
 // Limit values to -1.0..+1.0
@@ -2684,9 +2704,6 @@ bool AudioIoCallback::FillOutputBuffers(
    for (unsigned tt = 0; tt < numPlaybackSequences; ++tt) {
       auto vt = mPlaybackSequences[tt].get();
 
-      // Check for asynchronous user changes in mute, solo, pause status
-      const bool drop = SequenceShouldBeSilent(*vt);
-
       decltype(framesPerBuffer) len = 0;
 
       for (size_t c = 0; c < mNumPlaybackChannels; ++c) {
@@ -2725,12 +2742,11 @@ bool AudioIoCallback::FillOutputBuffers(
       // the device. For example mono channels output to both left and right
       // output channels.
       if (len > 0) {
-         auto &gains = mOldChannelGains[tt];
          AddToOutputChannel(0, outputMeterFloats, outputFloats,
-            tempBufs[0], drop, len, *vt, gains[0]);
+            tempBufs[0], len);
 
          AddToOutputChannel(1, outputMeterFloats, outputFloats,
-            tempBufs[1], drop, len, *vt, gains[1]);
+            tempBufs[1], len);
 
          // Output volume emulation
          const auto factor = decayFactor(mRate);
diff --git a/libraries/lib-audio-io/AudioIO.h b/libraries/lib-audio-io/AudioIO.h
index 4f1af8198332..ef3727ada7ce 100644
--- a/libraries/lib-audio-io/AudioIO.h
+++ b/libraries/lib-audio-io/AudioIO.h
@@ -194,14 +194,11 @@ class AUDIO_IO_API AudioIoCallback /* not final */
    /*!
     @param[in,out] channelGain
     */
-   void AddToOutputChannel( unsigned int chan, // index into gains
+   void AddToOutputChannel(unsigned int chan,
       float * outputMeterFloats,
       float * outputFloats,
       const float * tempBuf,
-      bool drop,
-      unsigned long len,
-      const PlayableSequence &ps,
-      float &laggingChannelGain
+      unsigned long len
    );
    bool FillOutputBuffers(
       float *outputBuffer,