Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 174 insertions, 0 deletions

diff --git a/dom/media/AudioPacketizer.h b/dom/media/AudioPacketizer.h
new file mode 100644
index 0000000000..8df04c0c5c
--- /dev/null
+++ b/dom/media/AudioPacketizer.h
@@ -0,0 +1,174 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* 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/. */
+
+#ifndef AudioPacketizer_h_
+#define AudioPacketizer_h_
+
+#include <mozilla/PodOperations.h>
+#include <mozilla/Assertions.h>
+#include <mozilla/UniquePtr.h>
+#include <AudioSampleFormat.h>
+
+// Enable this to warn when `Output` has been called but not enough data was
+// buffered.
+// #define LOG_PACKETIZER_UNDERRUN
+
+namespace mozilla {
+/**
+ * This class takes arbitrary input data, and returns packets of a specific
+ * size. In the process, it can convert audio samples from 16bit integers to
+ * float (or vice-versa).
+ *
+ * Input and output, as well as length units in the public interface are
+ * interleaved frames.
+ *
+ * Allocations of output buffer can be performed by this class.  Buffers can
+ * simply be delete-d.  This is because packets are intended to be sent off to
+ * non-gecko code using normal pointers/length pairs
+ *
+ * Alternatively, consumers can pass in a buffer in which the output is copied.
+ * The buffer needs to be large enough to store a packet worth of audio.
+ *
+ * The implementation uses a circular buffer using absolute virtual indices.
+ */
+template <typename InputType, typename OutputType>
+class AudioPacketizer {
+ public:
+  AudioPacketizer(uint32_t aPacketSize, uint32_t aChannels)
+      : mPacketSize(aPacketSize),
+        mChannels(aChannels),
+        mReadIndex(0),
+        mWriteIndex(0),
+        // Start off with a single packet
+        mStorage(new InputType[aPacketSize * aChannels]),
+        mLength(aPacketSize * aChannels) {
+    MOZ_ASSERT(aPacketSize > 0 && aChannels > 0,
+               "The packet size and the number of channel should be strictly "
+               "positive");
+  }
+
+  void Input(const InputType* aFrames, uint32_t aFrameCount) {
+    uint32_t inputSamples = aFrameCount * mChannels;
+    // Need to grow the storage. This should rarely happen, if at all, once the
+    // array has the right size.
+    if (inputSamples > EmptySlots()) {
+      // Calls to Input and Output are roughtly interleaved
+      // (Input,Output,Input,Output, etc.), or balanced
+      // (Input,Input,Input,Output,Output,Output), so we update the buffer to
+      // the exact right size in order to not waste space.
+      uint32_t newLength = AvailableSamples() + inputSamples;
+      uint32_t toCopy = AvailableSamples();
+      UniquePtr<InputType[]> oldStorage = std::move(mStorage);
+      mStorage = mozilla::MakeUnique<InputType[]>(newLength);
+      // Copy the old data at the beginning of the new storage.
+      if (WriteIndex() >= ReadIndex()) {
+        PodCopy(mStorage.get(), oldStorage.get() + ReadIndex(),
+                AvailableSamples());
+      } else {
+        uint32_t firstPartLength = mLength - ReadIndex();
+        uint32_t secondPartLength = AvailableSamples() - firstPartLength;
+        PodCopy(mStorage.get(), oldStorage.get() + ReadIndex(),
+                firstPartLength);
+        PodCopy(mStorage.get() + firstPartLength, oldStorage.get(),
+                secondPartLength);
+      }
+      mWriteIndex = toCopy;
+      mReadIndex = 0;
+      mLength = newLength;
+    }
+
+    if (WriteIndex() + inputSamples <= mLength) {
+      PodCopy(mStorage.get() + WriteIndex(), aFrames, aFrameCount * mChannels);
+    } else {
+      uint32_t firstPartLength = mLength - WriteIndex();
+      uint32_t secondPartLength = inputSamples - firstPartLength;
+      PodCopy(mStorage.get() + WriteIndex(), aFrames, firstPartLength);
+      PodCopy(mStorage.get(), aFrames + firstPartLength, secondPartLength);
+    }
+
+    mWriteIndex += inputSamples;
+  }
+
+  OutputType* Output() {
+    uint32_t samplesNeeded = mPacketSize * mChannels;
+    OutputType* out = new OutputType[samplesNeeded];
+
+    Output(out);
+
+    return out;
+  }
+
+  void Output(OutputType* aOutputBuffer) {
+    uint32_t samplesNeeded = mPacketSize * mChannels;
+
+    // Under-run. Pad the end of the buffer with silence.
+    if (AvailableSamples() < samplesNeeded) {
+#ifdef LOG_PACKETIZER_UNDERRUN
+      char buf[256];
+      snprintf(buf, 256,
+               "AudioPacketizer %p underrun: available: %u, needed: %u\n", this,
+               AvailableSamples(), samplesNeeded);
+      NS_WARNING(buf);
+#endif
+      uint32_t zeros = samplesNeeded - AvailableSamples();
+      PodZero(aOutputBuffer + AvailableSamples(), zeros);
+      samplesNeeded -= zeros;
+    }
+    if (ReadIndex() + samplesNeeded <= mLength) {
+      ConvertAudioSamples<InputType, OutputType>(mStorage.get() + ReadIndex(),
+                                                 aOutputBuffer, samplesNeeded);
+    } else {
+      uint32_t firstPartLength = mLength - ReadIndex();
+      uint32_t secondPartLength = samplesNeeded - firstPartLength;
+      ConvertAudioSamples<InputType, OutputType>(
+          mStorage.get() + ReadIndex(), aOutputBuffer, firstPartLength);
+      ConvertAudioSamples<InputType, OutputType>(
+          mStorage.get(), aOutputBuffer + firstPartLength, secondPartLength);
+    }
+    mReadIndex += samplesNeeded;
+  }
+
+  void Clear() {
+    mReadIndex = 0;
+    mWriteIndex = 0;
+  }
+
+  uint32_t PacketsAvailable() const {
+    return AvailableSamples() / mChannels / mPacketSize;
+  }
+
+  uint32_t FramesAvailable() const { return AvailableSamples() / mChannels; }
+
+  bool Empty() const { return mWriteIndex == mReadIndex; }
+
+  bool Full() const { return mWriteIndex - mReadIndex == mLength; }
+
+  // Size of one packet of audio, in frames
+  const uint32_t mPacketSize;
+  // Number of channels of the stream flowing through this packetizer
+  const uint32_t mChannels;
+
+ private:
+  uint32_t ReadIndex() const { return mReadIndex % mLength; }
+
+  uint32_t WriteIndex() const { return mWriteIndex % mLength; }
+
+  uint32_t AvailableSamples() const { return mWriteIndex - mReadIndex; }
+
+  uint32_t EmptySlots() const { return mLength - AvailableSamples(); }
+
+  // Two virtual index into the buffer: the read position and the write
+  // position.
+  uint64_t mReadIndex;
+  uint64_t mWriteIndex;
+  // Storage for the samples
+  mozilla::UniquePtr<InputType[]> mStorage;
+  // Length of the buffer, in samples
+  uint32_t mLength;
+};
+
+}  // namespace mozilla
+
+#endif  // AudioPacketizer_h_