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/* -*- 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 */
#ifndef OpusTrackEncoder_h_
#define OpusTrackEncoder_h_
#include <stdint.h>
#include <speex/speex_resampler.h>
#include "TimeUnits.h"
#include "TrackEncoder.h"
struct OpusEncoder;
namespace mozilla {
// Opus meta data structure
class OpusMetadata : public TrackMetadataBase {
// The ID Header of OggOpus. refer to
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 {
explicit OpusTrackEncoder(TrackRate aTrackRate);
virtual ~OpusTrackEncoder();
already_AddRefed<TrackMetadataBase> GetMetadata() override;
nsresult GetEncodedTrack(nsTArray<RefPtr<EncodedFrame>>& aData) override;
* The encoder lookahead at 48k rate.
int GetLookahead() const;
* The number of frames, in the input rate mTrackRate, needed to fill an
* encoded opus packet. A frame is a sample per channel.
int NumInputFramesPerPacket() const override;
nsresult Init(int aChannels) override;
* The number of frames, in the output rate (see GetOutputSampleRate), needed
* to fill an encoded opus packet. A frame is a sample per channel.
int NumOutputFramesPerPacket() const;
* True if the input needs to be resampled to be fed to the underlying opus
* encoder.
bool NeedsResampler() const;
* Get the samplerate of the data to be fed to the Opus encoder. This might be
* different from the input samplerate if resampling occurs.
const TrackRate mOutputSampleRate;
* 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 (in rate mOutputSampleRate), 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;
* Number of mLookahead samples that has been written. When non-zero and equal
* to mLookahead, encoding is complete.
int mLookaheadWritten;
* 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;
* Number of audio frames encoded, in kOpusSamplingRate.
uint64_t mNumOutputFrames;
} // namespace mozilla