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
15 namespace webrtc {
AudioCoder(uint32_t instanceID)16 AudioCoder::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
~AudioCoder()29 AudioCoder::~AudioCoder()
30 {
31 }
32
SetEncodeCodec(const CodecInst & codecInst,ACMAMRPackingFormat amrFormat)33 int32_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
SetDecodeCodec(const CodecInst & codecInst,ACMAMRPackingFormat amrFormat)43 int32_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
Decode(AudioFrame & decodedAudio,uint32_t sampFreqHz,const int8_t * incomingPayload,int32_t payloadLength)54 int32_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
PlayoutData(AudioFrame & decodedAudio,uint16_t & sampFreqHz)74 int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio,
75 uint16_t& sampFreqHz)
76 {
77 return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio);
78 }
79
Encode(const AudioFrame & audio,int8_t * encodedData,uint32_t & encodedLengthInBytes)80 int32_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
SendData(FrameType,uint8_t,uint32_t,const uint8_t * payloadData,uint16_t payloadSize,const RTPFragmentationHeader *)107 int32_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