OLD | NEW |
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" | 19 #include "webrtc/base/checks.h" |
20 #include "webrtc/base/format_macros.h" | 20 #include "webrtc/base/format_macros.h" |
21 #include "webrtc/base/logging.h" | 21 #include "webrtc/base/logging.h" |
22 #include "webrtc/base/safe_conversions.h" | 22 #include "webrtc/base/safe_conversions.h" |
23 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" | 23 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" |
24 #include "webrtc/common_types.h" | 24 #include "webrtc/common_types.h" |
25 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" | 25 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" |
26 #include "webrtc/modules/audio_coding/acm2/call_statistics.h" | 26 #include "webrtc/modules/audio_coding/acm2/call_statistics.h" |
27 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" | 27 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" |
28 #include "webrtc/system_wrappers/include/clock.h" | 28 #include "webrtc/system_wrappers/include/clock.h" |
29 #include "webrtc/system_wrappers/include/trace.h" | 29 #include "webrtc/system_wrappers/include/trace.h" |
| 30 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" |
30 | 31 |
31 namespace webrtc { | 32 namespace webrtc { |
32 | 33 |
33 namespace acm2 { | 34 namespace acm2 { |
34 | 35 |
35 AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config) | 36 AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config) |
36 : last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]), | 37 : last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]), |
37 neteq_(NetEq::Create(config.neteq_config, config.decoder_factory)), | 38 neteq_(NetEq::Create(config.neteq_config, config.decoder_factory)), |
38 clock_(config.clock), | 39 clock_(config.clock), |
39 resampled_last_output_frame_(true) { | 40 resampled_last_output_frame_(true) { |
(...skipping 152 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
192 return NetEqDecoder::kDecoderArbitrary; // External decoder. | 193 return NetEqDecoder::kDecoderArbitrary; // External decoder. |
193 const rtc::Optional<RentACodec::CodecId> cid = | 194 const rtc::Optional<RentACodec::CodecId> cid = |
194 RentACodec::CodecIdFromIndex(acm_codec_id); | 195 RentACodec::CodecIdFromIndex(acm_codec_id); |
195 RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id; | 196 RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id; |
196 const rtc::Optional<NetEqDecoder> ned = | 197 const rtc::Optional<NetEqDecoder> ned = |
197 RentACodec::NetEqDecoderFromCodecId(*cid, channels); | 198 RentACodec::NetEqDecoderFromCodecId(*cid, channels); |
198 RTC_DCHECK(ned) << "Invalid codec ID: " << static_cast<int>(*cid); | 199 RTC_DCHECK(ned) << "Invalid codec ID: " << static_cast<int>(*cid); |
199 return *ned; | 200 return *ned; |
200 }(); | 201 }(); |
201 const rtc::Optional<SdpAudioFormat> new_format = | 202 const rtc::Optional<SdpAudioFormat> new_format = |
202 RentACodec::NetEqDecoderToSdpAudioFormat(neteq_decoder); | 203 NetEqDecoderToSdpAudioFormat(neteq_decoder); |
203 | 204 |
204 rtc::CritScope lock(&crit_sect_); | 205 rtc::CritScope lock(&crit_sect_); |
205 | 206 |
206 const auto old_format = neteq_->GetDecoderFormat(payload_type); | 207 const auto old_format = neteq_->GetDecoderFormat(payload_type); |
207 if (old_format && new_format && *old_format == *new_format) { | 208 if (old_format && new_format && *old_format == *new_format) { |
208 // Re-registering the same codec. Do nothing and return. | 209 // Re-registering the same codec. Do nothing and return. |
209 return 0; | 210 return 0; |
210 } | 211 } |
211 | 212 |
212 if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK && | 213 if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK && |
(...skipping 174 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
387 | 388 |
388 void AcmReceiver::GetDecodingCallStatistics( | 389 void AcmReceiver::GetDecodingCallStatistics( |
389 AudioDecodingCallStats* stats) const { | 390 AudioDecodingCallStats* stats) const { |
390 rtc::CritScope lock(&crit_sect_); | 391 rtc::CritScope lock(&crit_sect_); |
391 *stats = call_stats_.GetDecodingStatistics(); | 392 *stats = call_stats_.GetDecodingStatistics(); |
392 } | 393 } |
393 | 394 |
394 } // namespace acm2 | 395 } // namespace acm2 |
395 | 396 |
396 } // namespace webrtc | 397 } // namespace webrtc |
OLD | NEW |