eb524d997b
Have Channel hold a pointer rather than reference, and shorten the name. TESTED=trybots R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2256004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4820 4adac7df-926f-26a2-2b94-8c16560cd09d
120 lines
3.6 KiB
C++
120 lines
3.6 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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#include "webrtc/common_types.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/modules/utility/source/coder.h"
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namespace webrtc {
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AudioCoder::AudioCoder(uint32_t instanceID)
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: _acm(AudioCodingModule::Create(instanceID)),
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_receiveCodec(),
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_encodeTimestamp(0),
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_encodedData(NULL),
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_encodedLengthInBytes(0),
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_decodeTimestamp(0)
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{
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_acm->InitializeSender();
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_acm->InitializeReceiver();
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_acm->RegisterTransportCallback(this);
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}
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AudioCoder::~AudioCoder()
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{
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}
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int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst,
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ACMAMRPackingFormat amrFormat)
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{
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if(_acm->RegisterSendCodec((CodecInst&)codecInst) == -1)
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{
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return -1;
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}
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return 0;
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}
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int32_t AudioCoder::SetDecodeCodec(const CodecInst& codecInst,
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ACMAMRPackingFormat amrFormat)
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{
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if(_acm->RegisterReceiveCodec((CodecInst&)codecInst) == -1)
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{
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return -1;
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}
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memcpy(&_receiveCodec,&codecInst,sizeof(CodecInst));
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return 0;
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}
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int32_t AudioCoder::Decode(AudioFrame& decodedAudio,
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uint32_t sampFreqHz,
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const int8_t* incomingPayload,
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int32_t payloadLength)
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{
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if (payloadLength > 0)
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{
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const uint8_t payloadType = _receiveCodec.pltype;
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_decodeTimestamp += _receiveCodec.pacsize;
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if(_acm->IncomingPayload((const uint8_t*) incomingPayload,
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payloadLength,
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payloadType,
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_decodeTimestamp) == -1)
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{
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return -1;
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}
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}
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return _acm->PlayoutData10Ms((uint16_t)sampFreqHz, &decodedAudio);
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}
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int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio,
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uint16_t& sampFreqHz)
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{
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return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio);
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}
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int32_t AudioCoder::Encode(const AudioFrame& audio,
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int8_t* encodedData,
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uint32_t& encodedLengthInBytes)
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{
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// Fake a timestamp in case audio doesn't contain a correct timestamp.
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// Make a local copy of the audio frame since audio is const
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AudioFrame audioFrame;
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audioFrame.CopyFrom(audio);
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audioFrame.timestamp_ = _encodeTimestamp;
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_encodeTimestamp += audioFrame.samples_per_channel_;
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// For any codec with a frame size that is longer than 10 ms the encoded
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// length in bytes should be zero until a a full frame has been encoded.
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_encodedLengthInBytes = 0;
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if(_acm->Add10MsData((AudioFrame&)audioFrame) == -1)
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{
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return -1;
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}
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_encodedData = encodedData;
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if(_acm->Process() == -1)
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{
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return -1;
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}
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encodedLengthInBytes = _encodedLengthInBytes;
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return 0;
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}
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int32_t AudioCoder::SendData(
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FrameType /* frameType */,
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uint8_t /* payloadType */,
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uint32_t /* timeStamp */,
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const uint8_t* payloadData,
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uint16_t payloadSize,
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const RTPFragmentationHeader* /* fragmentation*/)
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{
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memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize);
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_encodedLengthInBytes = payloadSize;
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return 0;
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}
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} // namespace webrtc
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