OLD | NEW |
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 96 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
107 static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1); | 107 static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1); |
108 const size_t approx_encoded_bytes = | 108 const size_t approx_encoded_bytes = |
109 Num10msFramesPerPacket() * 10 * bytes_per_millisecond; | 109 Num10msFramesPerPacket() * 10 * bytes_per_millisecond; |
110 return 2 * approx_encoded_bytes; | 110 return 2 * approx_encoded_bytes; |
111 } | 111 } |
112 | 112 |
113 int AudioEncoderOpus::SampleRateHz() const { | 113 int AudioEncoderOpus::SampleRateHz() const { |
114 return kSampleRateHz; | 114 return kSampleRateHz; |
115 } | 115 } |
116 | 116 |
117 int AudioEncoderOpus::NumChannels() const { | 117 size_t AudioEncoderOpus::NumChannels() const { |
118 return config_.num_channels; | 118 return config_.num_channels; |
119 } | 119 } |
120 | 120 |
121 size_t AudioEncoderOpus::Num10MsFramesInNextPacket() const { | 121 size_t AudioEncoderOpus::Num10MsFramesInNextPacket() const { |
122 return Num10msFramesPerPacket(); | 122 return Num10msFramesPerPacket(); |
123 } | 123 } |
124 | 124 |
125 size_t AudioEncoderOpus::Max10MsFramesInAPacket() const { | 125 size_t AudioEncoderOpus::Max10MsFramesInAPacket() const { |
126 return Num10msFramesPerPacket(); | 126 return Num10msFramesPerPacket(); |
127 } | 127 } |
(...skipping 12 matching lines...) Expand all Loading... |
140 RTC_DCHECK_EQ(SamplesPer10msFrame(), audio.size()); | 140 RTC_DCHECK_EQ(SamplesPer10msFrame(), audio.size()); |
141 input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend()); | 141 input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend()); |
142 if (input_buffer_.size() < | 142 if (input_buffer_.size() < |
143 (Num10msFramesPerPacket() * SamplesPer10msFrame())) { | 143 (Num10msFramesPerPacket() * SamplesPer10msFrame())) { |
144 return EncodedInfo(); | 144 return EncodedInfo(); |
145 } | 145 } |
146 RTC_CHECK_EQ(input_buffer_.size(), | 146 RTC_CHECK_EQ(input_buffer_.size(), |
147 Num10msFramesPerPacket() * SamplesPer10msFrame()); | 147 Num10msFramesPerPacket() * SamplesPer10msFrame()); |
148 int status = WebRtcOpus_Encode( | 148 int status = WebRtcOpus_Encode( |
149 inst_, &input_buffer_[0], | 149 inst_, &input_buffer_[0], |
150 rtc::CheckedDivExact(input_buffer_.size(), | 150 rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels), |
151 static_cast<size_t>(config_.num_channels)), | |
152 rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded); | 151 rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded); |
153 RTC_CHECK_GE(status, 0); // Fails only if fed invalid data. | 152 RTC_CHECK_GE(status, 0); // Fails only if fed invalid data. |
154 input_buffer_.clear(); | 153 input_buffer_.clear(); |
155 EncodedInfo info; | 154 EncodedInfo info; |
156 info.encoded_bytes = static_cast<size_t>(status); | 155 info.encoded_bytes = static_cast<size_t>(status); |
157 info.encoded_timestamp = first_timestamp_in_buffer_; | 156 info.encoded_timestamp = first_timestamp_in_buffer_; |
158 info.payload_type = config_.payload_type; | 157 info.payload_type = config_.payload_type; |
159 info.send_even_if_empty = true; // Allows Opus to send empty packets. | 158 info.send_even_if_empty = true; // Allows Opus to send empty packets. |
160 info.speech = (status > 0); | 159 info.speech = (status > 0); |
161 return info; | 160 return info; |
(...skipping 86 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
248 RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_)); | 247 RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_)); |
249 } | 248 } |
250 RTC_CHECK_EQ(0, | 249 RTC_CHECK_EQ(0, |
251 WebRtcOpus_SetPacketLossRate( | 250 WebRtcOpus_SetPacketLossRate( |
252 inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5))); | 251 inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5))); |
253 config_ = config; | 252 config_ = config; |
254 return true; | 253 return true; |
255 } | 254 } |
256 | 255 |
257 } // namespace webrtc | 256 } // namespace webrtc |
OLD | NEW |