| /* |
| * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef AUDIO_AUDIO_TRANSPORT_IMPL_H_ |
| #define AUDIO_AUDIO_TRANSPORT_IMPL_H_ |
| |
| #include <vector> |
| |
| #include "api/audio/audio_mixer.h" |
| #include "api/scoped_refptr.h" |
| #include "common_audio/resampler/include/push_resampler.h" |
| #include "modules/audio_device/include/audio_device.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "modules/audio_processing/typing_detection.h" |
| #include "rtc_base/constructor_magic.h" |
| #include "rtc_base/critical_section.h" |
| #include "rtc_base/thread_annotations.h" |
| |
| namespace webrtc { |
| |
| class AudioSendStream; |
| |
| class AudioTransportImpl : public AudioTransport { |
| public: |
| AudioTransportImpl(AudioMixer* mixer, AudioProcessing* audio_processing); |
| ~AudioTransportImpl() override; |
| |
| int32_t RecordedDataIsAvailable(const void* audioSamples, |
| const size_t nSamples, |
| const size_t nBytesPerSample, |
| const size_t nChannels, |
| const uint32_t samplesPerSec, |
| const uint32_t totalDelayMS, |
| const int32_t clockDrift, |
| const uint32_t currentMicLevel, |
| const bool keyPressed, |
| uint32_t& newMicLevel) override; |
| |
| int32_t NeedMorePlayData(const size_t nSamples, |
| const size_t nBytesPerSample, |
| const size_t nChannels, |
| const uint32_t samplesPerSec, |
| void* audioSamples, |
| size_t& nSamplesOut, |
| int64_t* elapsed_time_ms, |
| int64_t* ntp_time_ms) override; |
| |
| void PullRenderData(int bits_per_sample, |
| int sample_rate, |
| size_t number_of_channels, |
| size_t number_of_frames, |
| void* audio_data, |
| int64_t* elapsed_time_ms, |
| int64_t* ntp_time_ms) override; |
| |
| void UpdateSendingStreams(std::vector<AudioSendStream*> streams, |
| int send_sample_rate_hz, |
| size_t send_num_channels); |
| void SetStereoChannelSwapping(bool enable); |
| bool typing_noise_detected() const; |
| |
| private: |
| // Shared. |
| AudioProcessing* audio_processing_ = nullptr; |
| |
| // Capture side. |
| rtc::CriticalSection capture_lock_; |
| std::vector<AudioSendStream*> sending_streams_ RTC_GUARDED_BY(capture_lock_); |
| int send_sample_rate_hz_ RTC_GUARDED_BY(capture_lock_) = 8000; |
| size_t send_num_channels_ RTC_GUARDED_BY(capture_lock_) = 1; |
| bool typing_noise_detected_ RTC_GUARDED_BY(capture_lock_) = false; |
| bool swap_stereo_channels_ RTC_GUARDED_BY(capture_lock_) = false; |
| PushResampler<int16_t> capture_resampler_; |
| TypingDetection typing_detection_; |
| |
| // Render side. |
| rtc::scoped_refptr<AudioMixer> mixer_; |
| AudioFrame mixed_frame_; |
| // Converts mixed audio to the audio device output rate. |
| PushResampler<int16_t> render_resampler_; |
| |
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportImpl); |
| }; |
| } // namespace webrtc |
| |
| #endif // AUDIO_AUDIO_TRANSPORT_IMPL_H_ |