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Side by Side Diff: webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc

Issue 1534193008: Misc. small cleanups (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Unnecessary parens Created 4 years, 11 months ago
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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
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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 }
(...skipping 18 matching lines...) Expand all
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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