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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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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323 const PacketTime& packet_time) { | 323 const PacketTime& packet_time) { |
324 // TODO(solenberg): Tests call this function on a network thread, libjingle | 324 // TODO(solenberg): Tests call this function on a network thread, libjingle |
325 // calls on the worker thread. We should move towards always using a network | 325 // calls on the worker thread. We should move towards always using a network |
326 // thread. Then this check can be enabled. | 326 // thread. Then this check can be enabled. |
327 // RTC_DCHECK(!thread_checker_.CalledOnValidThread()); | 327 // RTC_DCHECK(!thread_checker_.CalledOnValidThread()); |
328 RTPHeader header; | 328 RTPHeader header; |
329 if (!rtp_header_parser_->Parse(packet, length, &header)) { | 329 if (!rtp_header_parser_->Parse(packet, length, &header)) { |
330 return false; | 330 return false; |
331 } | 331 } |
332 | 332 |
333 // Only forward if the parsed header has one of the headers necessary for | |
334 // bandwidth estimation. RTP timestamps has different rates for audio and | |
335 // video and shouldn't be mixed. | |
336 if (config_.rtp.transport_cc && | |
337 header.extension.hasTransportSequenceNumber) { | |
338 int64_t arrival_time_ms = rtc::TimeMillis(); | |
339 if (packet_time.timestamp >= 0) | |
340 arrival_time_ms = (packet_time.timestamp + 500) / 1000; | |
341 size_t payload_size = length - header.headerLength; | |
342 remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size, | |
343 header); | |
344 } | |
345 | |
346 return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time); | 333 return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time); |
347 } | 334 } |
348 | 335 |
349 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const { | 336 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const { |
350 RTC_DCHECK_RUN_ON(&worker_thread_checker_); | 337 RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
351 return config_; | 338 return config_; |
352 } | 339 } |
353 | 340 |
354 VoiceEngine* AudioReceiveStream::voice_engine() const { | 341 VoiceEngine* AudioReceiveStream::voice_engine() const { |
355 auto* voice_engine = audio_state()->voice_engine(); | 342 auto* voice_engine = audio_state()->voice_engine(); |
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366 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) { | 353 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) { |
367 ScopedVoEInterface<VoEBase> base(voice_engine()); | 354 ScopedVoEInterface<VoEBase> base(voice_engine()); |
368 if (playout) { | 355 if (playout) { |
369 return base->StartPlayout(config_.voe_channel_id); | 356 return base->StartPlayout(config_.voe_channel_id); |
370 } else { | 357 } else { |
371 return base->StopPlayout(config_.voe_channel_id); | 358 return base->StopPlayout(config_.voe_channel_id); |
372 } | 359 } |
373 } | 360 } |
374 } // namespace internal | 361 } // namespace internal |
375 } // namespace webrtc | 362 } // namespace webrtc |
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