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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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| 824 RTC_DCHECK_EQ(last_mode_, kModeExpand); | 824 RTC_DCHECK_EQ(last_mode_, kModeExpand); |
| 825 playout_timestamp_ += static_cast<uint32_t>(output_size_samples_); | 825 playout_timestamp_ += static_cast<uint32_t>(output_size_samples_); |
| 826 audio_frame->sample_rate_hz_ = fs_hz_; | 826 audio_frame->sample_rate_hz_ = fs_hz_; |
| 827 audio_frame->samples_per_channel_ = output_size_samples_; | 827 audio_frame->samples_per_channel_ = output_size_samples_; |
| 828 audio_frame->timestamp_ = | 828 audio_frame->timestamp_ = |
| 829 first_packet_ | 829 first_packet_ |
| 830 ? 0 | 830 ? 0 |
| 831 : timestamp_scaler_->ToExternal(playout_timestamp_) - | 831 : timestamp_scaler_->ToExternal(playout_timestamp_) - |
| 832 static_cast<uint32_t>(audio_frame->samples_per_channel_); | 832 static_cast<uint32_t>(audio_frame->samples_per_channel_); |
| 833 audio_frame->num_channels_ = sync_buffer_->Channels(); | 833 audio_frame->num_channels_ = sync_buffer_->Channels(); |
| 834 stats_.ExpandedNoiseSamples(output_size_samples_); |
| 834 *muted = true; | 835 *muted = true; |
| 835 return 0; | 836 return 0; |
| 836 } | 837 } |
| 837 | 838 |
| 838 int return_value = GetDecision(&operation, &packet_list, &dtmf_event, | 839 int return_value = GetDecision(&operation, &packet_list, &dtmf_event, |
| 839 &play_dtmf); | 840 &play_dtmf); |
| 840 if (return_value != 0) { | 841 if (return_value != 0) { |
| 841 last_mode_ = kModeError; | 842 last_mode_ = kModeError; |
| 842 return return_value; | 843 return return_value; |
| 843 } | 844 } |
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| 2163 } | 2164 } |
| 2164 } | 2165 } |
| 2165 | 2166 |
| 2166 void NetEqImpl::CreateDecisionLogic() { | 2167 void NetEqImpl::CreateDecisionLogic() { |
| 2167 decision_logic_.reset(DecisionLogic::Create( | 2168 decision_logic_.reset(DecisionLogic::Create( |
| 2168 fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(), | 2169 fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(), |
| 2169 *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(), | 2170 *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(), |
| 2170 tick_timer_.get())); | 2171 tick_timer_.get())); |
| 2171 } | 2172 } |
| 2172 } // namespace webrtc | 2173 } // namespace webrtc |
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