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5cccea6f0f
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-o -iname "*.h" \
-o -iname "*.c" \
-o -iname "*.cc" \
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convert MOZ_OVERRIDE override
convert MOZ_FINAL final
87 lines
2.4 KiB
C++
87 lines
2.4 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this file,
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* You can obtain one at http://mozilla.org/MPL/2.0/. */
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#ifndef OpusTrackEncoder_h_
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#define OpusTrackEncoder_h_
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#include <stdint.h>
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#include <speex/speex_resampler.h>
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#include "TrackEncoder.h"
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struct OpusEncoder;
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namespace mozilla {
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// Opus meta data structure
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class OpusMetadata : public TrackMetadataBase
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{
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public:
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// The ID Header of OggOpus. refer to http://wiki.xiph.org/OggOpus.
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nsTArray<uint8_t> mIdHeader;
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// The Comment Header of OggOpus.
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nsTArray<uint8_t> mCommentHeader;
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MetadataKind GetKind() const override { return METADATA_OPUS; }
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};
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class OpusTrackEncoder : public AudioTrackEncoder
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{
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public:
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OpusTrackEncoder();
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virtual ~OpusTrackEncoder();
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already_AddRefed<TrackMetadataBase> GetMetadata() override;
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nsresult GetEncodedTrack(EncodedFrameContainer& aData) override;
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protected:
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int GetPacketDuration() override;
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nsresult Init(int aChannels, int aSamplingRate) override;
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/**
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* Get the samplerate of the data to be fed to the Opus encoder. This might be
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* different from the input samplerate if resampling occurs.
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*/
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int GetOutputSampleRate();
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private:
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/**
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* The Opus encoder from libopus.
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*/
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OpusEncoder* mEncoder;
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/**
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* A local segment queue which takes the raw data out from mRawSegment in the
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* call of GetEncodedTrack(). Opus encoder only accepts GetPacketDuration()
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* samples from mSourceSegment every encoding cycle, thus it needs to be
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* global in order to store the leftover segments taken from mRawSegment.
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*/
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AudioSegment mSourceSegment;
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/**
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* Total samples of delay added by codec, can be queried by the encoder. From
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* the perspective of decoding, real data begins this many samples late, so
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* the encoder needs to append this many null samples to the end of stream,
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* in order to align the time of input and output.
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*/
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int mLookahead;
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/**
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* If the input sample rate does not divide 48kHz evenly, the input data are
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* resampled.
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*/
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SpeexResamplerState* mResampler;
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/**
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* Store the resampled frames that don't fit into an Opus packet duration.
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* They will be prepended to the resampled frames next encoding cycle.
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*/
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nsTArray<AudioDataValue> mResampledLeftover;
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};
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}
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#endif
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