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

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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
11 #include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h" 11 #include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
12 12
13 namespace webrtc { 13 namespace webrtc {
14 14
15 namespace acm2 { 15 namespace acm2 {
16 16
17 InitialDelayManager::InitialDelayManager(int initial_delay_ms, 17 InitialDelayManager::InitialDelayManager(int initial_delay_ms,
18 int late_packet_threshold) 18 int late_packet_threshold)
19 : last_packet_type_(kUndefinedPacket), 19 : last_packet_type_(kUndefinedPacket),
20 last_receive_timestamp_(0), 20 last_receive_timestamp_(0),
21 timestamp_step_(0), 21 timestamp_step_(0),
(...skipping 211 matching lines...) Expand 10 before | Expand all | Expand 10 after
233 233
234 void InitialDelayManager::UpdatePlayoutTimestamp( 234 void InitialDelayManager::UpdatePlayoutTimestamp(
235 const RTPHeader& current_header, int sample_rate_hz) { 235 const RTPHeader& current_header, int sample_rate_hz) {
236 playout_timestamp_ = current_header.timestamp - static_cast<uint32_t>( 236 playout_timestamp_ = current_header.timestamp - static_cast<uint32_t>(
237 initial_delay_ms_ * sample_rate_hz / 1000); 237 initial_delay_ms_ * sample_rate_hz / 1000);
238 } 238 }
239 239
240 } // namespace acm2 240 } // namespace acm2
241 241
242 } // namespace webrtc 242 } // namespace webrtc
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