Files
platform-external-webrtc/webrtc/modules/audio_coding/main/acm2/acm_opus.h
tina.legrand@webrtc.org 65d61c3924 Opus send rate overflows if over 65 kbps
The member holding the send rate for Opus had too low resolution for rates above ~65 kbps.

I've added a test that checks if the average rate in a Opus test is in the right range. The test fails before my fix, and now passes.

BUG=3267
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 13:42:51 +00:00

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/*
* 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);
int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
virtual int SetFEC(bool enable_fec) OVERRIDE;
virtual int SetPacketLossRate(int loss_rate) OVERRIDE;
protected:
void DestructEncoderSafe();
int16_t InternalCreateEncoder();
void InternalDestructEncoderInst(void* ptr_inst);
int16_t SetBitRateSafe(const int32_t rate);
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_