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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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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425 int8_t OutputEnergyLevel() const | 425 int8_t OutputEnergyLevel() const |
426 { | 426 { |
427 return _outputAudioLevel.Level(); | 427 return _outputAudioLevel.Level(); |
428 } | 428 } |
429 uint32_t Demultiplex(const AudioFrame& audioFrame); | 429 uint32_t Demultiplex(const AudioFrame& audioFrame); |
430 // Demultiplex the data to the channel's |_audioFrame|. The difference | 430 // Demultiplex the data to the channel's |_audioFrame|. The difference |
431 // between this method and the overloaded method above is that |audio_data| | 431 // between this method and the overloaded method above is that |audio_data| |
432 // does not go through transmit_mixer and APM. | 432 // does not go through transmit_mixer and APM. |
433 void Demultiplex(const int16_t* audio_data, | 433 void Demultiplex(const int16_t* audio_data, |
434 int sample_rate, | 434 int sample_rate, |
435 int number_of_frames, | 435 size_t number_of_frames, |
henrika_webrtc
2015/07/13 07:38:08
It is not clear to me why size_t is required here.
Peter Kasting
2015/07/13 22:12:28
Sure. A fairly complete argument/set of tradeoffs
henrika_webrtc
2015/07/14 07:54:09
Thanks!
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436 int number_of_channels); | 436 int number_of_channels); |
437 uint32_t PrepareEncodeAndSend(int mixingFrequency); | 437 uint32_t PrepareEncodeAndSend(int mixingFrequency); |
438 uint32_t EncodeAndSend(); | 438 uint32_t EncodeAndSend(); |
439 | 439 |
440 // Associate to a send channel. | 440 // Associate to a send channel. |
441 // Used for obtaining RTT for a receive-only channel. | 441 // Used for obtaining RTT for a receive-only channel. |
442 void set_associate_send_channel(const ChannelOwner& channel) { | 442 void set_associate_send_channel(const ChannelOwner& channel) { |
443 assert(_channelId != channel.channel()->ChannelId()); | 443 assert(_channelId != channel.channel()->ChannelId()); |
444 CriticalSectionScoped lock(assoc_send_channel_lock_.get()); | 444 CriticalSectionScoped lock(assoc_send_channel_lock_.get()); |
445 associate_send_channel_ = channel; | 445 associate_send_channel_ = channel; |
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578 rtc::scoped_ptr<NetworkPredictor> network_predictor_; | 578 rtc::scoped_ptr<NetworkPredictor> network_predictor_; |
579 // An associated send channel. | 579 // An associated send channel. |
580 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_; | 580 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_; |
581 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); | 581 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); |
582 }; | 582 }; |
583 | 583 |
584 } // namespace voe | 584 } // namespace voe |
585 } // namespace webrtc | 585 } // namespace webrtc |
586 | 586 |
587 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 587 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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