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

Issue 1421013006: Delete a chain of methods in ViE, VoE and ACM (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 1 month 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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758 "%s failed: No send codec is registered.", caller_name); 758 "%s failed: No send codec is registered.", caller_name);
759 return false; 759 return false;
760 } 760 }
761 return true; 761 return true;
762 } 762 }
763 763
764 int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) { 764 int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
765 return receiver_.RemoveCodec(payload_type); 765 return receiver_.RemoveCodec(payload_type);
766 } 766 }
767 767
768 int AudioCodingModuleImpl::SetInitialPlayoutDelay(int delay_ms) {
769 {
770 CriticalSectionScoped lock(acm_crit_sect_.get());
771 // Initialize receiver, if it is not initialized. Otherwise, initial delay
772 // is reset upon initialization of the receiver.
773 if (!receiver_initialized_)
774 InitializeReceiverSafe();
775 }
776 return receiver_.SetInitialDelay(delay_ms);
777 }
778
779 int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) { 768 int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
780 return receiver_.EnableNack(max_nack_list_size); 769 return receiver_.EnableNack(max_nack_list_size);
781 } 770 }
782 771
783 void AudioCodingModuleImpl::DisableNack() { 772 void AudioCodingModuleImpl::DisableNack() {
784 receiver_.DisableNack(); 773 receiver_.DisableNack();
785 } 774 }
786 775
787 std::vector<uint16_t> AudioCodingModuleImpl::GetNackList( 776 std::vector<uint16_t> AudioCodingModuleImpl::GetNackList(
788 int64_t round_trip_time_ms) const { 777 int64_t round_trip_time_ms) const {
789 return receiver_.GetNackList(round_trip_time_ms); 778 return receiver_.GetNackList(round_trip_time_ms);
790 } 779 }
791 780
792 int AudioCodingModuleImpl::LeastRequiredDelayMs() const { 781 int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
793 return receiver_.LeastRequiredDelayMs(); 782 return receiver_.LeastRequiredDelayMs();
794 } 783 }
795 784
796 void AudioCodingModuleImpl::GetDecodingCallStatistics( 785 void AudioCodingModuleImpl::GetDecodingCallStatistics(
797 AudioDecodingCallStats* call_stats) const { 786 AudioDecodingCallStats* call_stats) const {
798 receiver_.GetDecodingCallStatistics(call_stats); 787 receiver_.GetDecodingCallStatistics(call_stats);
799 } 788 }
800 789
801 } // namespace acm2 790 } // namespace acm2
802 } // namespace webrtc 791 } // namespace webrtc
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