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/interface/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->InitializeSender(); 25 _acm->InitializeReceiver(); 26 _acm->RegisterTransportCallback(this); 27} 28 29AudioCoder::~AudioCoder() 30{ 31} 32 33int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst, 34 ACMAMRPackingFormat amrFormat) 35{ 36 if(_acm->RegisterSendCodec((CodecInst&)codecInst) == -1) 37 { 38 return -1; 39 } 40 return 0; 41} 42 43int32_t AudioCoder::SetDecodeCodec(const CodecInst& codecInst, 44 ACMAMRPackingFormat amrFormat) 45{ 46 if(_acm->RegisterReceiveCodec((CodecInst&)codecInst) == -1) 47 { 48 return -1; 49 } 50 memcpy(&_receiveCodec,&codecInst,sizeof(CodecInst)); 51 return 0; 52} 53 54int32_t AudioCoder::Decode(AudioFrame& decodedAudio, 55 uint32_t sampFreqHz, 56 const int8_t* incomingPayload, 57 int32_t payloadLength) 58{ 59 if (payloadLength > 0) 60 { 61 const uint8_t payloadType = _receiveCodec.pltype; 62 _decodeTimestamp += _receiveCodec.pacsize; 63 if(_acm->IncomingPayload((const uint8_t*) incomingPayload, 64 payloadLength, 65 payloadType, 66 _decodeTimestamp) == -1) 67 { 68 return -1; 69 } 70 } 71 return _acm->PlayoutData10Ms((uint16_t)sampFreqHz, &decodedAudio); 72} 73 74int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio, 75 uint16_t& sampFreqHz) 76{ 77 return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio); 78} 79 80int32_t AudioCoder::Encode(const AudioFrame& audio, 81 int8_t* encodedData, 82 uint32_t& encodedLengthInBytes) 83{ 84 // Fake a timestamp in case audio doesn't contain a correct timestamp. 85 // Make a local copy of the audio frame since audio is const 86 AudioFrame audioFrame; 87 audioFrame.CopyFrom(audio); 88 audioFrame.timestamp_ = _encodeTimestamp; 89 _encodeTimestamp += audioFrame.samples_per_channel_; 90 91 // For any codec with a frame size that is longer than 10 ms the encoded 92 // length in bytes should be zero until a a full frame has been encoded. 93 _encodedLengthInBytes = 0; 94 if(_acm->Add10MsData((AudioFrame&)audioFrame) == -1) 95 { 96 return -1; 97 } 98 _encodedData = encodedData; 99 if(_acm->Process() == -1) 100 { 101 return -1; 102 } 103 encodedLengthInBytes = _encodedLengthInBytes; 104 return 0; 105} 106 107int32_t AudioCoder::SendData( 108 FrameType /* frameType */, 109 uint8_t /* payloadType */, 110 uint32_t /* timeStamp */, 111 const uint8_t* payloadData, 112 uint16_t payloadSize, 113 const RTPFragmentationHeader* /* fragmentation*/) 114{ 115 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize); 116 _encodedLengthInBytes = payloadSize; 117 return 0; 118} 119} // namespace webrtc 120