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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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| 191 first_timestamp_in_buffer_ = rtp_timestamp; | 191 first_timestamp_in_buffer_ = rtp_timestamp; |
| 192 input_buffer_.insert(input_buffer_.end(), audio, | 192 input_buffer_.insert(input_buffer_.end(), audio, |
| 193 audio + samples_per_10ms_frame_); | 193 audio + samples_per_10ms_frame_); |
| 194 if (input_buffer_.size() < (static_cast<size_t>(num_10ms_frames_per_packet_) * | 194 if (input_buffer_.size() < (static_cast<size_t>(num_10ms_frames_per_packet_) * |
| 195 samples_per_10ms_frame_)) { | 195 samples_per_10ms_frame_)) { |
| 196 return EncodedInfo(); | 196 return EncodedInfo(); |
| 197 } | 197 } |
| 198 CHECK_EQ(input_buffer_.size(), | 198 CHECK_EQ(input_buffer_.size(), |
| 199 static_cast<size_t>(num_10ms_frames_per_packet_) * | 199 static_cast<size_t>(num_10ms_frames_per_packet_) * |
| 200 samples_per_10ms_frame_); | 200 samples_per_10ms_frame_); |
| 201 int16_t status = WebRtcOpus_Encode( | 201 int status = WebRtcOpus_Encode( |
| 202 inst_, &input_buffer_[0], | 202 inst_, &input_buffer_[0], |
| 203 rtc::CheckedDivExact(CastInt16(input_buffer_.size()), | 203 rtc::CheckedDivExact(CastInt16(input_buffer_.size()), |
| 204 static_cast<int16_t>(num_channels_)), | 204 static_cast<int16_t>(num_channels_)), |
| 205 ClampInt16(max_encoded_bytes), encoded); | 205 ClampInt16(max_encoded_bytes), encoded); |
| 206 CHECK_GE(status, 0); // Fails only if fed invalid data. | 206 CHECK_GE(status, 0); // Fails only if fed invalid data. |
| 207 input_buffer_.clear(); | 207 input_buffer_.clear(); |
| 208 EncodedInfo info; | 208 EncodedInfo info; |
| 209 info.encoded_bytes = static_cast<size_t>(status); | 209 info.encoded_bytes = static_cast<size_t>(status); |
| 210 info.encoded_timestamp = first_timestamp_in_buffer_; | 210 info.encoded_timestamp = first_timestamp_in_buffer_; |
| 211 info.payload_type = payload_type_; | 211 info.payload_type = payload_type_; |
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| 256 return Reconstruct(conf); | 256 return Reconstruct(conf); |
| 257 } | 257 } |
| 258 | 258 |
| 259 bool AudioEncoderMutableOpus::SetMaxPlaybackRate(int frequency_hz) { | 259 bool AudioEncoderMutableOpus::SetMaxPlaybackRate(int frequency_hz) { |
| 260 auto conf = config(); | 260 auto conf = config(); |
| 261 conf.max_playback_rate_hz = frequency_hz; | 261 conf.max_playback_rate_hz = frequency_hz; |
| 262 return Reconstruct(conf); | 262 return Reconstruct(conf); |
| 263 } | 263 } |
| 264 | 264 |
| 265 } // namespace webrtc | 265 } // namespace webrtc |
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