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Side by Side Diff: webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc

Issue 1976913002: Add muted_output parameter to ACM (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 months ago
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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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784 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, 784 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
785 "Delay must be in the range of 0-1000 milliseconds."); 785 "Delay must be in the range of 0-1000 milliseconds.");
786 return -1; 786 return -1;
787 } 787 }
788 return receiver_.SetMaximumDelay(time_ms); 788 return receiver_.SetMaximumDelay(time_ms);
789 } 789 }
790 790
791 // Get 10 milliseconds of raw audio data to play out. 791 // Get 10 milliseconds of raw audio data to play out.
792 // Automatic resample to the requested frequency. 792 // Automatic resample to the requested frequency.
793 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz, 793 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
794 AudioFrame* audio_frame) { 794 AudioFrame* audio_frame,
795 bool* muted) {
795 // GetAudio always returns 10 ms, at the requested sample rate. 796 // GetAudio always returns 10 ms, at the requested sample rate.
796 if (receiver_.GetAudio(desired_freq_hz, audio_frame) != 0) { 797 if (receiver_.GetAudio(desired_freq_hz, audio_frame, muted) != 0) {
797 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, 798 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
798 "PlayoutData failed, RecOut Failed"); 799 "PlayoutData failed, RecOut Failed");
799 return -1; 800 return -1;
800 } 801 }
801 audio_frame->id_ = id_; 802 audio_frame->id_ = id_;
802 return 0; 803 return 0;
803 } 804 }
804 805
806 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
807 AudioFrame* audio_frame) {
808 bool muted;
809 int ret = PlayoutData10Ms(desired_freq_hz, audio_frame, &muted);
810 RTC_DCHECK(!muted);
minyue-webrtc 2016/05/13 09:43:12 Is it possible to check enable_fast_accelerate of
hlundin-webrtc 2016/05/13 09:45:52 The config is not stored anywhere, so it is not st
811 return ret;
812 }
813
805 ///////////////////////////////////////// 814 /////////////////////////////////////////
806 // Statistics 815 // Statistics
807 // 816 //
808 817
809 // TODO(turajs) change the return value to void. Also change the corresponding 818 // TODO(turajs) change the return value to void. Also change the corresponding
810 // NetEq function. 819 // NetEq function.
811 int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) { 820 int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) {
812 receiver_.GetNetworkStatistics(statistics); 821 receiver_.GetNetworkStatistics(statistics);
813 return 0; 822 return 0;
814 } 823 }
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938 return receiver_.LeastRequiredDelayMs(); 947 return receiver_.LeastRequiredDelayMs();
939 } 948 }
940 949
941 void AudioCodingModuleImpl::GetDecodingCallStatistics( 950 void AudioCodingModuleImpl::GetDecodingCallStatistics(
942 AudioDecodingCallStats* call_stats) const { 951 AudioDecodingCallStats* call_stats) const {
943 receiver_.GetDecodingCallStatistics(call_stats); 952 receiver_.GetDecodingCallStatistics(call_stats);
944 } 953 }
945 954
946 } // namespace acm2 955 } // namespace acm2
947 } // namespace webrtc 956 } // namespace webrtc
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