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Side by Side Diff: webrtc/modules/audio_coding/acm2/acm_receiver.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/acm_receiver.h" 11 #include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
12 12
13 #include <stdlib.h> // malloc 13 #include <stdlib.h> // malloc
14 14
15 #include <algorithm> // sort 15 #include <algorithm> // sort
16 #include <vector> 16 #include <vector>
17 17
18 #include "webrtc/base/checks.h" 18 #include "webrtc/base/checks.h"
19 #include "webrtc/base/format_macros.h" 19 #include "webrtc/base/format_macros.h"
20 #include "webrtc/base/logging.h" 20 #include "webrtc/base/logging.h"
21 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar y.h" 21 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar y.h"
22 #include "webrtc/common_types.h" 22 #include "webrtc/common_types.h"
23 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" 23 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
24 #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" 24 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
25 #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" 25 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
26 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" 26 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
27 #include "webrtc/system_wrappers/include/clock.h" 27 #include "webrtc/system_wrappers/include/clock.h"
28 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" 28 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
29 #include "webrtc/system_wrappers/include/tick_util.h" 29 #include "webrtc/system_wrappers/include/tick_util.h"
30 #include "webrtc/system_wrappers/include/trace.h" 30 #include "webrtc/system_wrappers/include/trace.h"
31 31
32 namespace webrtc { 32 namespace webrtc {
33 33
34 namespace acm2 { 34 namespace acm2 {
35 35
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531 531
532 void AcmReceiver::GetDecodingCallStatistics( 532 void AcmReceiver::GetDecodingCallStatistics(
533 AudioDecodingCallStats* stats) const { 533 AudioDecodingCallStats* stats) const {
534 CriticalSectionScoped lock(crit_sect_.get()); 534 CriticalSectionScoped lock(crit_sect_.get());
535 *stats = call_stats_.GetDecodingStatistics(); 535 *stats = call_stats_.GetDecodingStatistics();
536 } 536 }
537 537
538 } // namespace acm2 538 } // namespace acm2
539 539
540 } // namespace webrtc 540 } // namespace webrtc
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