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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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78 | 78 |
79 int32_t AudioCoder::Encode(const AudioFrame& audio, | 79 int32_t AudioCoder::Encode(const AudioFrame& audio, |
80 int8_t* encodedData, | 80 int8_t* encodedData, |
81 size_t& encodedLengthInBytes) | 81 size_t& encodedLengthInBytes) |
82 { | 82 { |
83 // Fake a timestamp in case audio doesn't contain a correct timestamp. | 83 // Fake a timestamp in case audio doesn't contain a correct timestamp. |
84 // Make a local copy of the audio frame since audio is const | 84 // Make a local copy of the audio frame since audio is const |
85 AudioFrame audioFrame; | 85 AudioFrame audioFrame; |
86 audioFrame.CopyFrom(audio); | 86 audioFrame.CopyFrom(audio); |
87 audioFrame.timestamp_ = _encodeTimestamp; | 87 audioFrame.timestamp_ = _encodeTimestamp; |
88 _encodeTimestamp += audioFrame.samples_per_channel_; | 88 _encodeTimestamp += static_cast<uint32_t>(audioFrame.samples_per_channel_); |
89 | 89 |
90 // For any codec with a frame size that is longer than 10 ms the encoded | 90 // For any codec with a frame size that is longer than 10 ms the encoded |
91 // length in bytes should be zero until a a full frame has been encoded. | 91 // length in bytes should be zero until a a full frame has been encoded. |
92 _encodedLengthInBytes = 0; | 92 _encodedLengthInBytes = 0; |
93 if(_acm->Add10MsData((AudioFrame&)audioFrame) == -1) | 93 if(_acm->Add10MsData((AudioFrame&)audioFrame) == -1) |
94 { | 94 { |
95 return -1; | 95 return -1; |
96 } | 96 } |
97 _encodedData = encodedData; | 97 _encodedData = encodedData; |
98 encodedLengthInBytes = _encodedLengthInBytes; | 98 encodedLengthInBytes = _encodedLengthInBytes; |
99 return 0; | 99 return 0; |
100 } | 100 } |
101 | 101 |
102 int32_t AudioCoder::SendData( | 102 int32_t AudioCoder::SendData( |
103 FrameType /* frameType */, | 103 FrameType /* frameType */, |
104 uint8_t /* payloadType */, | 104 uint8_t /* payloadType */, |
105 uint32_t /* timeStamp */, | 105 uint32_t /* timeStamp */, |
106 const uint8_t* payloadData, | 106 const uint8_t* payloadData, |
107 size_t payloadSize, | 107 size_t payloadSize, |
108 const RTPFragmentationHeader* /* fragmentation*/) | 108 const RTPFragmentationHeader* /* fragmentation*/) |
109 { | 109 { |
110 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize); | 110 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize); |
111 _encodedLengthInBytes = payloadSize; | 111 _encodedLengthInBytes = payloadSize; |
112 return 0; | 112 return 0; |
113 } | 113 } |
114 } // namespace webrtc | 114 } // namespace webrtc |
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