6aac93bd9c
This is a step to solve https://code.google.com/p/webrtc/issues/detail?id=1906 In particular, we add an API in VoE and ACM to call Opus's API of setting maximum bandwidth. TEST = added a test in voe_cmd_test and listened to the result BUG= R=henrika@google.com, henrika@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6869 4adac7df-926f-26a2-2b94-8c16560cd09d
67 lines
1.8 KiB
C++
67 lines
1.8 KiB
C++
/*
|
|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_
|
|
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_
|
|
|
|
#include "webrtc/common_audio/resampler/include/resampler.h"
|
|
#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
|
|
|
|
struct WebRtcOpusEncInst;
|
|
struct WebRtcOpusDecInst;
|
|
|
|
namespace webrtc {
|
|
|
|
namespace acm2 {
|
|
|
|
class ACMOpus : public ACMGenericCodec {
|
|
public:
|
|
explicit ACMOpus(int16_t codec_id);
|
|
~ACMOpus();
|
|
|
|
ACMGenericCodec* CreateInstance(void);
|
|
|
|
int16_t InternalEncode(uint8_t* bitstream,
|
|
int16_t* bitstream_len_byte) OVERRIDE
|
|
EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
|
|
|
|
int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
|
|
|
|
virtual int SetFEC(bool enable_fec) OVERRIDE;
|
|
|
|
virtual int SetPacketLossRate(int loss_rate) OVERRIDE;
|
|
|
|
virtual int SetOpusMaxBandwidth(int max_bandwidth) OVERRIDE;
|
|
|
|
protected:
|
|
void DestructEncoderSafe();
|
|
|
|
int16_t InternalCreateEncoder();
|
|
|
|
void InternalDestructEncoderInst(void* ptr_inst);
|
|
|
|
int16_t SetBitRateSafe(const int32_t rate) OVERRIDE
|
|
EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
|
|
|
|
WebRtcOpusEncInst* encoder_inst_ptr_;
|
|
uint16_t sample_freq_;
|
|
int32_t bitrate_;
|
|
int channels_;
|
|
|
|
bool fec_enabled_;
|
|
int packet_loss_rate_;
|
|
};
|
|
|
|
} // namespace acm2
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_
|