7dba7860c7
This CL is to allow to set Opus target application at the creation of an encoder. According to Opus spec, there are three applications: OPUS_APPLICATION_VOIP OPUS_APPLICATION_AUDIO OPUS_APPLICATION_RESTRICTED_LOWDELAY BUG= R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37479004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8103 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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explicit ACMOpus(int16_t codec_id);
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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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