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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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199 return codec_manager_.RegisterEncoder(send_codec); | 199 return codec_manager_.RegisterEncoder(send_codec); |
200 } | 200 } |
201 | 201 |
202 void AudioCodingModuleImpl::RegisterExternalSendCodec( | 202 void AudioCodingModuleImpl::RegisterExternalSendCodec( |
203 AudioEncoder* external_speech_encoder) { | 203 AudioEncoder* external_speech_encoder) { |
204 CriticalSectionScoped lock(acm_crit_sect_.get()); | 204 CriticalSectionScoped lock(acm_crit_sect_.get()); |
205 codec_manager_.RegisterEncoder(external_speech_encoder); | 205 codec_manager_.RegisterEncoder(external_speech_encoder); |
206 } | 206 } |
207 | 207 |
208 // Get current send codec. | 208 // Get current send codec. |
209 int AudioCodingModuleImpl::SendCodec(CodecInst* current_codec) const { | 209 rtc::Maybe<CodecInst> AudioCodingModuleImpl::SendCodec() const { |
210 CriticalSectionScoped lock(acm_crit_sect_.get()); | 210 CriticalSectionScoped lock(acm_crit_sect_.get()); |
211 return codec_manager_.GetCodecInst(current_codec); | 211 return codec_manager_.GetCodecInst(); |
212 } | 212 } |
213 | 213 |
214 // Get current send frequency. | 214 // Get current send frequency. |
215 int AudioCodingModuleImpl::SendFrequency() const { | 215 int AudioCodingModuleImpl::SendFrequency() const { |
216 WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, | 216 WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, |
217 "SendFrequency()"); | 217 "SendFrequency()"); |
218 CriticalSectionScoped lock(acm_crit_sect_.get()); | 218 CriticalSectionScoped lock(acm_crit_sect_.get()); |
219 | 219 |
220 if (!codec_manager_.CurrentEncoder()) { | 220 if (!codec_manager_.CurrentEncoder()) { |
221 WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, | 221 WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, |
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782 return receiver_.LeastRequiredDelayMs(); | 782 return receiver_.LeastRequiredDelayMs(); |
783 } | 783 } |
784 | 784 |
785 void AudioCodingModuleImpl::GetDecodingCallStatistics( | 785 void AudioCodingModuleImpl::GetDecodingCallStatistics( |
786 AudioDecodingCallStats* call_stats) const { | 786 AudioDecodingCallStats* call_stats) const { |
787 receiver_.GetDecodingCallStatistics(call_stats); | 787 receiver_.GetDecodingCallStatistics(call_stats); |
788 } | 788 } |
789 | 789 |
790 } // namespace acm2 | 790 } // namespace acm2 |
791 } // namespace webrtc | 791 } // namespace webrtc |
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