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

Issue 1631443002: AudioCodingModuleImpl: Initialize encoder_stack_ to nullptr (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 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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100 } 100 }
101 } 101 }
102 102
103 AudioCodingModuleImpl::AudioCodingModuleImpl( 103 AudioCodingModuleImpl::AudioCodingModuleImpl(
104 const AudioCodingModule::Config& config) 104 const AudioCodingModule::Config& config)
105 : id_(config.id), 105 : id_(config.id),
106 expected_codec_ts_(0xD87F3F9F), 106 expected_codec_ts_(0xD87F3F9F),
107 expected_in_ts_(0xD87F3F9F), 107 expected_in_ts_(0xD87F3F9F),
108 receiver_(config), 108 receiver_(config),
109 bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"), 109 bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
110 encoder_stack_(nullptr),
110 previous_pltype_(255), 111 previous_pltype_(255),
111 receiver_initialized_(false), 112 receiver_initialized_(false),
112 first_10ms_data_(false), 113 first_10ms_data_(false),
113 first_frame_(true), 114 first_frame_(true),
114 packetization_callback_(NULL), 115 packetization_callback_(NULL),
115 vad_callback_(NULL) { 116 vad_callback_(NULL) {
116 if (InitializeReceiverSafe() < 0) { 117 if (InitializeReceiverSafe() < 0) {
117 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, 118 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
118 "Cannot initialize receiver"); 119 "Cannot initialize receiver");
119 } 120 }
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813 return receiver_.LeastRequiredDelayMs(); 814 return receiver_.LeastRequiredDelayMs();
814 } 815 }
815 816
816 void AudioCodingModuleImpl::GetDecodingCallStatistics( 817 void AudioCodingModuleImpl::GetDecodingCallStatistics(
817 AudioDecodingCallStats* call_stats) const { 818 AudioDecodingCallStats* call_stats) const {
818 receiver_.GetDecodingCallStatistics(call_stats); 819 receiver_.GetDecodingCallStatistics(call_stats);
819 } 820 }
820 821
821 } // namespace acm2 822 } // namespace acm2
822 } // namespace webrtc 823 } // namespace webrtc
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