Adding upstream version 110.0.1.upstream/110.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 128 insertions, 0 deletions

diff --git a/dom/media/AudioCompactor.h b/dom/media/AudioCompactor.h
new file mode 100644
index 0000000000..b058ba3755
--- /dev/null
+++ b/dom/media/AudioCompactor.h
@@ -0,0 +1,128 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+#if !defined(AudioCompactor_h)
+#  define AudioCompactor_h
+
+#  include "MediaQueue.h"
+#  include "MediaData.h"
+#  include "VideoUtils.h"
+
+namespace mozilla {
+
+class AudioCompactor {
+ public:
+  explicit AudioCompactor(MediaQueue<AudioData>& aQueue) : mQueue(aQueue) {
+    // Determine padding size used by AlignedBuffer.
+    size_t paddedSize = AlignedAudioBuffer::AlignmentPaddingSize();
+    mSamplesPadding = paddedSize / sizeof(AudioDataValue);
+    if (mSamplesPadding * sizeof(AudioDataValue) < paddedSize) {
+      // Round up.
+      mSamplesPadding++;
+    }
+  }
+
+  // Push audio data into the underlying queue with minimal heap allocation
+  // slop.  This method is responsible for allocating AudioDataValue[] buffers.
+  // The caller must provide a functor to copy the data into the buffers.  The
+  // functor must provide the following signature:
+  //
+  //   uint32_t operator()(AudioDataValue *aBuffer, uint32_t aSamples);
+  //
+  // The functor must copy as many complete frames as possible to the provided
+  // buffer given its length (in AudioDataValue elements).  The number of frames
+  // copied must be returned.  This copy functor must support being called
+  // multiple times in order to copy the audio data fully.  The copy functor
+  // must copy full frames as partial frames will be ignored.
+  template <typename CopyFunc>
+  bool Push(int64_t aOffset, int64_t aTime, int32_t aSampleRate,
+            uint32_t aFrames, uint32_t aChannels, CopyFunc aCopyFunc) {
+    auto time = media::TimeUnit::FromMicroseconds(aTime);
+
+    // If we are losing more than a reasonable amount to padding, try to chunk
+    // the data.
+    size_t maxSlop = AudioDataSize(aFrames, aChannels) / MAX_SLOP_DIVISOR;
+
+    while (aFrames > 0) {
+      uint32_t samples = GetChunkSamples(aFrames, aChannels, maxSlop);
+      if (samples / aChannels > mSamplesPadding / aChannels + 1) {
+        samples -= mSamplesPadding;
+      }
+      AlignedAudioBuffer buffer(samples);
+      if (!buffer) {
+        return false;
+      }
+
+      // Copy audio data to buffer using caller-provided functor.
+      uint32_t framesCopied = aCopyFunc(buffer.get(), samples);
+
+      NS_ASSERTION(framesCopied <= aFrames, "functor copied too many frames");
+      buffer.SetLength(size_t(framesCopied) * aChannels);
+
+      auto duration = FramesToTimeUnit(framesCopied, aSampleRate);
+      if (!duration.IsValid()) {
+        return false;
+      }
+
+      RefPtr<AudioData> data = new AudioData(aOffset, time, std::move(buffer),
+                                             aChannels, aSampleRate);
+      MOZ_DIAGNOSTIC_ASSERT(duration == data->mDuration, "must be equal");
+      mQueue.Push(data);
+
+      // Remove the frames we just pushed into the queue and loop if there is
+      // more to be done.
+      time += duration;
+      aFrames -= framesCopied;
+
+      // NOTE: No need to update aOffset as its only an approximation anyway.
+    }
+
+    return true;
+  }
+
+  // Copy functor suitable for copying audio samples already in the
+  // AudioDataValue format/layout expected by AudioStream on this platform.
+  class NativeCopy {
+   public:
+    NativeCopy(const uint8_t* aSource, size_t aSourceBytes, uint32_t aChannels)
+        : mSource(aSource),
+          mSourceBytes(aSourceBytes),
+          mChannels(aChannels),
+          mNextByte(0) {}
+
+    uint32_t operator()(AudioDataValue* aBuffer, uint32_t aSamples);
+
+   private:
+    const uint8_t* const mSource;
+    const size_t mSourceBytes;
+    const uint32_t mChannels;
+    size_t mNextByte;
+  };
+
+  // Allow 12.5% slop before chunking kicks in.  Public so that the gtest can
+  // access it.
+  static const size_t MAX_SLOP_DIVISOR = 8;
+
+ private:
+  // Compute the number of AudioDataValue samples that will be fit the most
+  // frames while keeping heap allocation slop less than the given threshold.
+  static uint32_t GetChunkSamples(uint32_t aFrames, uint32_t aChannels,
+                                  size_t aMaxSlop);
+
+  static size_t BytesPerFrame(uint32_t aChannels) {
+    return sizeof(AudioDataValue) * aChannels;
+  }
+
+  static size_t AudioDataSize(uint32_t aFrames, uint32_t aChannels) {
+    return aFrames * BytesPerFrame(aChannels);
+  }
+
+  MediaQueue<AudioData>& mQueue;
+  size_t mSamplesPadding;
+};
+
+}  // namespace mozilla
+
+#endif  // AudioCompactor_h