
This effectively reverts r8211. The problem with r8210 was that the change in constructor signature was not done for other codec selections that then default one. That is, some code that was hidden under #ifdef did not get updated. This is now fixed. BUG=4228 COAUTHOR=kwiberg@webrtc.org TBR=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37879004 Cr-Commit-Position: refs/heads/master@{#8215} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8215 4adac7df-926f-26a2-2b94-8c16560cd09d
72 lines
1.9 KiB
C++
72 lines
1.9 KiB
C++
/*
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* Copyright (c) 2012 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_MAIN_ACM2_ACM_OPUS_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_
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#include "webrtc/common_audio/resampler/include/resampler.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
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struct WebRtcOpusEncInst;
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struct WebRtcOpusDecInst;
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namespace webrtc {
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namespace acm2 {
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class ACMOpus : public ACMGenericCodec {
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public:
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ACMOpus(int16_t codec_id, bool enable_red);
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~ACMOpus();
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ACMGenericCodec* CreateInstance(void);
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int16_t InternalEncode(uint8_t* bitstream,
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int16_t* bitstream_len_byte) override
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EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
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int16_t InitEncoderSafe(WebRtcACMCodecParams* codec_params,
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bool force_initialization) override
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EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
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int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
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int SetFEC(bool enable_fec) override;
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int SetOpusApplication(OpusApplicationMode mode) override;
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int SetPacketLossRate(int loss_rate) override;
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int SetOpusMaxPlaybackRate(int frequency_hz) override;
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protected:
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void DestructEncoderSafe();
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int16_t InternalCreateEncoder();
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int16_t SetBitRateSafe(const int32_t rate) override
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EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
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WebRtcOpusEncInst* encoder_inst_ptr_;
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uint16_t sample_freq_;
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int32_t bitrate_;
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int channels_;
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int packet_loss_rate_;
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OpusApplicationMode application_;
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};
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} // namespace acm2
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_
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