import FIREFOX_52_6_0esr_RELEASE from mozilla-esr52 hg repo

dom/media/encoder/OpusTrackEncoder.h

@@ -0,0 +1,91 @@
+/* -*- 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 OpusTrackEncoder_h_
+#define OpusTrackEncoder_h_
+
+#include <stdint.h>
+#include <speex/speex_resampler.h>
+#include "TrackEncoder.h"
+
+struct OpusEncoder;
+
+namespace mozilla {
+
+// Opus meta data structure
+class OpusMetadata : public TrackMetadataBase
+{
+public:
+  // The ID Header of OggOpus. refer to http://wiki.xiph.org/OggOpus.
+  nsTArray<uint8_t> mIdHeader;
+  // The Comment Header of OggOpus.
+  nsTArray<uint8_t> mCommentHeader;
+  int32_t mChannels;
+  float mSamplingFrequency;
+  MetadataKind GetKind() const override { return METADATA_OPUS; }
+};
+
+class OpusTrackEncoder : public AudioTrackEncoder
+{
+public:
+  OpusTrackEncoder();
+  virtual ~OpusTrackEncoder();
+
+  already_AddRefed<TrackMetadataBase> GetMetadata() override;
+
+  nsresult GetEncodedTrack(EncodedFrameContainer& aData) override;
+
+protected:
+  int GetPacketDuration() override;
+
+  nsresult Init(int aChannels, int aSamplingRate) override;
+
+  /**
+   * Get the samplerate of the data to be fed to the Opus encoder. This might be
+   * different from the input samplerate if resampling occurs.
+   */
+  int GetOutputSampleRate();
+
+private:
+  /**
+   * The Opus encoder from libopus.
+   */
+  OpusEncoder* mEncoder;
+
+  /**
+   * A local segment queue which takes the raw data out from mRawSegment in the
+   * call of GetEncodedTrack(). Opus encoder only accepts GetPacketDuration()
+   * samples from mSourceSegment every encoding cycle, thus it needs to be
+   * global in order to store the leftover segments taken from mRawSegment.
+   */
+  AudioSegment mSourceSegment;
+
+  /**
+   * Total samples of delay added by codec, can be queried by the encoder. From
+   * the perspective of decoding, real data begins this many samples late, so
+   * the encoder needs to append this many null samples to the end of stream,
+   * in order to align the time of input and output.
+   */
+  int mLookahead;
+
+  /**
+   * If the input sample rate does not divide 48kHz evenly, the input data are
+   * resampled.
+   */
+  SpeexResamplerState* mResampler;
+
+  /**
+   * Store the resampled frames that don't fit into an Opus packet duration.
+   * They will be prepended to the resampled frames next encoding cycle.
+   */
+  nsTArray<AudioDataValue> mResampledLeftover;
+
+  // TimeStamp in microseconds.
+  uint64_t mOutputTimeStamp;
+};
+
+} // namespace mozilla
+
+#endif