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Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
96 lines
3.7 KiB
C++
96 lines
3.7 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef AUDIO_AUDIO_TRANSPORT_IMPL_H_
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#define AUDIO_AUDIO_TRANSPORT_IMPL_H_
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#include <vector>
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#include "api/audio/audio_mixer.h"
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#include "audio/audio_level.h"
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#include "common_audio/resampler/include/push_resampler.h"
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#include "modules/audio_device/include/audio_device.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/typing_detection.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/scoped_ref_ptr.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class AudioSendStream;
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class AudioTransportImpl : public AudioTransport {
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public:
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AudioTransportImpl(AudioMixer* mixer, AudioProcessing* audio_processing);
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~AudioTransportImpl() override;
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int32_t RecordedDataIsAvailable(const void* audioSamples,
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const size_t nSamples,
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const size_t nBytesPerSample,
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const size_t nChannels,
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const uint32_t samplesPerSec,
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const uint32_t totalDelayMS,
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const int32_t clockDrift,
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const uint32_t currentMicLevel,
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const bool keyPressed,
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uint32_t& newMicLevel) override;
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int32_t NeedMorePlayData(const size_t nSamples,
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const size_t nBytesPerSample,
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const size_t nChannels,
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const uint32_t samplesPerSec,
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void* audioSamples,
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size_t& nSamplesOut,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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void PullRenderData(int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames,
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void* audio_data,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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void UpdateSendingStreams(std::vector<AudioSendStream*> streams,
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int send_sample_rate_hz,
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size_t send_num_channels);
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void SetStereoChannelSwapping(bool enable);
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bool typing_noise_detected() const;
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const voe::AudioLevel& audio_level() const { return audio_level_; }
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private:
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// Shared.
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AudioProcessing* audio_processing_ = nullptr;
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// Capture side.
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rtc::CriticalSection capture_lock_;
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std::vector<AudioSendStream*> sending_streams_ RTC_GUARDED_BY(capture_lock_);
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int send_sample_rate_hz_ RTC_GUARDED_BY(capture_lock_) = 8000;
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size_t send_num_channels_ RTC_GUARDED_BY(capture_lock_) = 1;
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bool typing_noise_detected_ RTC_GUARDED_BY(capture_lock_) = false;
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bool swap_stereo_channels_ RTC_GUARDED_BY(capture_lock_) = false;
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PushResampler<int16_t> capture_resampler_;
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voe::AudioLevel audio_level_;
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TypingDetection typing_detection_;
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// Render side.
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rtc::scoped_refptr<AudioMixer> mixer_;
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AudioFrame mixed_frame_;
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// Converts mixed audio to the audio device output rate.
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PushResampler<int16_t> render_resampler_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportImpl);
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};
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} // namespace webrtc
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#endif // AUDIO_AUDIO_TRANSPORT_IMPL_H_
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