2011-07-07 08:21:25 +00:00
|
|
|
/*
|
|
|
|
|
* Copyright (c) 2011 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_UTILITY_SOURCE_CODER_H_
|
|
|
|
|
#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
|
|
|
|
|
|
2015-02-26 14:34:55 +00:00
|
|
|
#include "webrtc/base/scoped_ptr.h"
|
2013-07-12 08:28:10 +00:00
|
|
|
#include "webrtc/common_types.h"
|
|
|
|
|
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
|
|
|
|
|
#include "webrtc/typedefs.h"
|
2011-07-07 08:21:25 +00:00
|
|
|
|
|
|
|
|
namespace webrtc {
|
|
|
|
|
class AudioFrame;
|
|
|
|
|
|
|
|
|
|
class AudioCoder : public AudioPacketizationCallback
|
|
|
|
|
{
|
|
|
|
|
public:
|
2013-04-09 13:32:55 +00:00
|
|
|
AudioCoder(uint32_t instanceID);
|
2011-07-07 08:21:25 +00:00
|
|
|
~AudioCoder();
|
|
|
|
|
|
2013-04-09 13:32:55 +00:00
|
|
|
int32_t SetEncodeCodec(
|
2011-07-07 08:21:25 +00:00
|
|
|
const CodecInst& codecInst,
|
2014-09-10 22:14:59 +00:00
|
|
|
ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient);
|
2011-07-07 08:21:25 +00:00
|
|
|
|
2013-04-09 13:32:55 +00:00
|
|
|
int32_t SetDecodeCodec(
|
2011-07-07 08:21:25 +00:00
|
|
|
const CodecInst& codecInst,
|
2014-09-10 22:14:59 +00:00
|
|
|
ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient);
|
2011-07-07 08:21:25 +00:00
|
|
|
|
2013-04-09 13:32:55 +00:00
|
|
|
int32_t Decode(AudioFrame& decodedAudio, uint32_t sampFreqHz,
|
Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
|
|
|
const int8_t* incomingPayload, size_t payloadLength);
|
2011-07-07 08:21:25 +00:00
|
|
|
|
2013-04-09 13:32:55 +00:00
|
|
|
int32_t PlayoutData(AudioFrame& decodedAudio, uint16_t& sampFreqHz);
|
2011-07-07 08:21:25 +00:00
|
|
|
|
2013-04-09 13:32:55 +00:00
|
|
|
int32_t Encode(const AudioFrame& audio, int8_t* encodedData,
|
Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
|
|
|
size_t& encodedLengthInBytes);
|
2011-07-07 08:21:25 +00:00
|
|
|
|
|
|
|
|
protected:
|
2015-03-04 12:58:35 +00:00
|
|
|
int32_t SendData(FrameType frameType,
|
|
|
|
|
uint8_t payloadType,
|
|
|
|
|
uint32_t timeStamp,
|
|
|
|
|
const uint8_t* payloadData,
|
|
|
|
|
size_t payloadSize,
|
|
|
|
|
const RTPFragmentationHeader* fragmentation) override;
|
2011-07-07 08:21:25 +00:00
|
|
|
|
|
|
|
|
private:
|
2015-02-26 14:34:55 +00:00
|
|
|
rtc::scoped_ptr<AudioCodingModule> _acm;
|
2011-07-07 08:21:25 +00:00
|
|
|
|
|
|
|
|
CodecInst _receiveCodec;
|
|
|
|
|
|
2013-04-09 13:32:55 +00:00
|
|
|
uint32_t _encodeTimestamp;
|
|
|
|
|
int8_t* _encodedData;
|
Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
|
|
|
size_t _encodedLengthInBytes;
|
2011-07-07 08:21:25 +00:00
|
|
|
|
2013-04-09 13:32:55 +00:00
|
|
|
uint32_t _decodeTimestamp;
|
2011-07-07 08:21:25 +00:00
|
|
|
};
|
2013-07-03 15:12:26 +00:00
|
|
|
} // namespace webrtc
|
2011-07-07 08:21:25 +00:00
|
|
|
|
|
|
|
|
#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
|