1/* 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11#include "webrtc/common_types.h" 12#include "webrtc/modules/include/module_common_types.h" 13#include "webrtc/modules/utility/source/coder.h" 14 15namespace webrtc { 16AudioCoder::AudioCoder(uint32_t instanceID) 17 : _acm(AudioCodingModule::Create(instanceID)), 18 _receiveCodec(), 19 _encodeTimestamp(0), 20 _encodedData(NULL), 21 _encodedLengthInBytes(0), 22 _decodeTimestamp(0) 23{ 24 _acm->InitializeReceiver(); 25 _acm->RegisterTransportCallback(this); 26} 27 28AudioCoder::~AudioCoder() 29{ 30} 31 32int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst) 33{ 34 if(_acm->RegisterSendCodec((CodecInst&)codecInst) == -1) 35 { 36 return -1; 37 } 38 return 0; 39} 40 41int32_t AudioCoder::SetDecodeCodec(const CodecInst& codecInst) 42{ 43 if(_acm->RegisterReceiveCodec((CodecInst&)codecInst) == -1) 44 { 45 return -1; 46 } 47 memcpy(&_receiveCodec,&codecInst,sizeof(CodecInst)); 48 return 0; 49} 50 51int32_t AudioCoder::Decode(AudioFrame& decodedAudio, 52 uint32_t sampFreqHz, 53 const int8_t* incomingPayload, 54 size_t payloadLength) 55{ 56 if (payloadLength > 0) 57 { 58 const uint8_t payloadType = _receiveCodec.pltype; 59 _decodeTimestamp += _receiveCodec.pacsize; 60 if(_acm->IncomingPayload((const uint8_t*) incomingPayload, 61 payloadLength, 62 payloadType, 63 _decodeTimestamp) == -1) 64 { 65 return -1; 66 } 67 } 68 return _acm->PlayoutData10Ms((uint16_t)sampFreqHz, &decodedAudio); 69} 70 71int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio, 72 uint16_t& sampFreqHz) 73{ 74 return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio); 75} 76 77int32_t AudioCoder::Encode(const AudioFrame& audio, 78 int8_t* encodedData, 79 size_t& encodedLengthInBytes) 80{ 81 // Fake a timestamp in case audio doesn't contain a correct timestamp. 82 // Make a local copy of the audio frame since audio is const 83 AudioFrame audioFrame; 84 audioFrame.CopyFrom(audio); 85 audioFrame.timestamp_ = _encodeTimestamp; 86 _encodeTimestamp += static_cast<uint32_t>(audioFrame.samples_per_channel_); 87 88 // For any codec with a frame size that is longer than 10 ms the encoded 89 // length in bytes should be zero until a a full frame has been encoded. 90 _encodedLengthInBytes = 0; 91 if(_acm->Add10MsData((AudioFrame&)audioFrame) == -1) 92 { 93 return -1; 94 } 95 _encodedData = encodedData; 96 encodedLengthInBytes = _encodedLengthInBytes; 97 return 0; 98} 99 100int32_t AudioCoder::SendData( 101 FrameType /* frameType */, 102 uint8_t /* payloadType */, 103 uint32_t /* timeStamp */, 104 const uint8_t* payloadData, 105 size_t payloadSize, 106 const RTPFragmentationHeader* /* fragmentation*/) 107{ 108 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize); 109 _encodedLengthInBytes = payloadSize; 110 return 0; 111} 112} // namespace webrtc 113