Make an AudioEncoder subclass for Opus
BUG=3926 R=henrik.lundin@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23239004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7552 4adac7df-926f-26a2-2b94-8c16560cd09d
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/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
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#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
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#include <vector>
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#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
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#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
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namespace webrtc {
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class AudioEncoderOpus : public AudioEncoder {
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public:
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struct Config {
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Config();
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bool IsOk() const;
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int frame_size_ms;
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int num_channels;
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};
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explicit AudioEncoderOpus(const Config& config);
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virtual ~AudioEncoderOpus() OVERRIDE;
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virtual int sample_rate_hz() const OVERRIDE;
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virtual int num_channels() const OVERRIDE;
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virtual int num_10ms_frames_per_packet() const OVERRIDE;
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protected:
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virtual bool Encode(uint32_t timestamp,
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const int16_t* audio,
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size_t max_encoded_bytes,
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uint8_t* encoded,
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size_t* encoded_bytes,
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uint32_t* encoded_timestamp) OVERRIDE;
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private:
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const int num_10ms_frames_per_packet_;
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const int num_channels_;
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const int samples_per_10ms_frame_;
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std::vector<int16_t> input_buffer_;
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OpusEncInst* inst_;
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uint32_t first_timestamp_in_buffer_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
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