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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/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/api/audio_codecs/audio_decoder.h" | 18 #include "webrtc/api/audio_codecs/audio_decoder.h" |
19 #include "webrtc/base/checks.h" | |
20 #include "webrtc/base/format_macros.h" | |
21 #include "webrtc/base/logging.h" | |
22 #include "webrtc/base/safe_conversions.h" | |
23 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" | 19 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" |
24 #include "webrtc/common_types.h" | 20 #include "webrtc/common_types.h" |
25 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" | 21 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" |
26 #include "webrtc/modules/audio_coding/acm2/call_statistics.h" | 22 #include "webrtc/modules/audio_coding/acm2/call_statistics.h" |
| 23 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" |
27 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" | 24 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" |
| 25 #include "webrtc/rtc_base/checks.h" |
| 26 #include "webrtc/rtc_base/format_macros.h" |
| 27 #include "webrtc/rtc_base/logging.h" |
| 28 #include "webrtc/rtc_base/safe_conversions.h" |
28 #include "webrtc/system_wrappers/include/clock.h" | 29 #include "webrtc/system_wrappers/include/clock.h" |
29 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" | |
30 | 30 |
31 namespace webrtc { | 31 namespace webrtc { |
32 | 32 |
33 namespace acm2 { | 33 namespace acm2 { |
34 | 34 |
35 AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config) | 35 AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config) |
36 : last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]), | 36 : last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]), |
37 neteq_(NetEq::Create(config.neteq_config, config.decoder_factory)), | 37 neteq_(NetEq::Create(config.neteq_config, config.decoder_factory)), |
38 clock_(config.clock), | 38 clock_(config.clock), |
39 resampled_last_output_frame_(true) { | 39 resampled_last_output_frame_(true) { |
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393 | 393 |
394 void AcmReceiver::GetDecodingCallStatistics( | 394 void AcmReceiver::GetDecodingCallStatistics( |
395 AudioDecodingCallStats* stats) const { | 395 AudioDecodingCallStats* stats) const { |
396 rtc::CritScope lock(&crit_sect_); | 396 rtc::CritScope lock(&crit_sect_); |
397 *stats = call_stats_.GetDecodingStatistics(); | 397 *stats = call_stats_.GetDecodingStatistics(); |
398 } | 398 } |
399 | 399 |
400 } // namespace acm2 | 400 } // namespace acm2 |
401 | 401 |
402 } // namespace webrtc | 402 } // namespace webrtc |
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