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| 1 /* | 1 /* | 
| 2  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2  *  Copyright (c) 2014 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 | 
| (...skipping 65 matching lines...) Expand 10 before | Expand all | Expand 10 after  Loading... | 
| 76 | 76 | 
| 77 size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const { | 77 size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const { | 
| 78   return num_10ms_frames_per_packet_; | 78   return num_10ms_frames_per_packet_; | 
| 79 } | 79 } | 
| 80 | 80 | 
| 81 size_t AudioEncoderPcm::Max10MsFramesInAPacket() const { | 81 size_t AudioEncoderPcm::Max10MsFramesInAPacket() const { | 
| 82   return num_10ms_frames_per_packet_; | 82   return num_10ms_frames_per_packet_; | 
| 83 } | 83 } | 
| 84 | 84 | 
| 85 int AudioEncoderPcm::GetTargetBitrate() const { | 85 int AudioEncoderPcm::GetTargetBitrate() const { | 
| 86   return 8 * BytesPerSample() * SampleRateHz() * NumChannels(); | 86   return static_cast<int>( | 
|  | 87       8 * BytesPerSample() * SampleRateHz() * NumChannels()); | 
| 87 } | 88 } | 
| 88 | 89 | 
| 89 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal( | 90 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal( | 
| 90     uint32_t rtp_timestamp, | 91     uint32_t rtp_timestamp, | 
| 91     rtc::ArrayView<const int16_t> audio, | 92     rtc::ArrayView<const int16_t> audio, | 
| 92     size_t max_encoded_bytes, | 93     size_t max_encoded_bytes, | 
| 93     uint8_t* encoded) { | 94     uint8_t* encoded) { | 
| 94   if (speech_buffer_.empty()) { | 95   if (speech_buffer_.empty()) { | 
| 95     first_timestamp_in_buffer_ = rtp_timestamp; | 96     first_timestamp_in_buffer_ = rtp_timestamp; | 
| 96   } | 97   } | 
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| 115 | 116 | 
| 116 AudioEncoderPcmA::AudioEncoderPcmA(const CodecInst& codec_inst) | 117 AudioEncoderPcmA::AudioEncoderPcmA(const CodecInst& codec_inst) | 
| 117     : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(codec_inst)) {} | 118     : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(codec_inst)) {} | 
| 118 | 119 | 
| 119 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, | 120 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, | 
| 120                                     size_t input_len, | 121                                     size_t input_len, | 
| 121                                     uint8_t* encoded) { | 122                                     uint8_t* encoded) { | 
| 122   return WebRtcG711_EncodeA(audio, input_len, encoded); | 123   return WebRtcG711_EncodeA(audio, input_len, encoded); | 
| 123 } | 124 } | 
| 124 | 125 | 
| 125 int AudioEncoderPcmA::BytesPerSample() const { | 126 size_t AudioEncoderPcmA::BytesPerSample() const { | 
| 126   return 1; | 127   return 1; | 
| 127 } | 128 } | 
| 128 | 129 | 
| 129 AudioEncoderPcmU::AudioEncoderPcmU(const CodecInst& codec_inst) | 130 AudioEncoderPcmU::AudioEncoderPcmU(const CodecInst& codec_inst) | 
| 130     : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(codec_inst)) {} | 131     : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(codec_inst)) {} | 
| 131 | 132 | 
| 132 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, | 133 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, | 
| 133                                     size_t input_len, | 134                                     size_t input_len, | 
| 134                                     uint8_t* encoded) { | 135                                     uint8_t* encoded) { | 
| 135   return WebRtcG711_EncodeU(audio, input_len, encoded); | 136   return WebRtcG711_EncodeU(audio, input_len, encoded); | 
| 136 } | 137 } | 
| 137 | 138 | 
| 138 int AudioEncoderPcmU::BytesPerSample() const { | 139 size_t AudioEncoderPcmU::BytesPerSample() const { | 
| 139   return 1; | 140   return 1; | 
| 140 } | 141 } | 
| 141 | 142 | 
| 142 }  // namespace webrtc | 143 }  // namespace webrtc | 
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