62238097c9
This is a safe cleanup change since top-level const applied to parameters in function declarations (that are not also definitions) are ignored by the compiler. Hence, such changes do not change the type of the declared functions and are simply no-ops. Bug: webrtc:13610 Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35802}
108 lines
3.8 KiB
C++
108 lines
3.8 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 <memory>
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#include <vector>
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#include "api/audio/audio_mixer.h"
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#include "api/scoped_refptr.h"
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#include "common_audio/resampler/include/push_resampler.h"
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#include "modules/async_audio_processing/async_audio_processing.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/synchronization/mutex.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class AudioSender;
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class AudioTransportImpl : public AudioTransport {
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public:
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AudioTransportImpl(
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AudioMixer* mixer,
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AudioProcessing* audio_processing,
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AsyncAudioProcessing::Factory* async_audio_processing_factory);
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AudioTransportImpl() = delete;
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AudioTransportImpl(const AudioTransportImpl&) = delete;
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AudioTransportImpl& operator=(const AudioTransportImpl&) = delete;
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~AudioTransportImpl() override;
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int32_t RecordedDataIsAvailable(const void* audioSamples,
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size_t nSamples,
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size_t nBytesPerSample,
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size_t nChannels,
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uint32_t samplesPerSec,
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uint32_t totalDelayMS,
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int32_t clockDrift,
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uint32_t currentMicLevel,
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bool keyPressed,
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uint32_t& newMicLevel) override;
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int32_t NeedMorePlayData(size_t nSamples,
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size_t nBytesPerSample,
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size_t nChannels,
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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 UpdateAudioSenders(std::vector<AudioSender*> senders,
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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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private:
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void SendProcessedData(std::unique_ptr<AudioFrame> audio_frame);
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// Shared.
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AudioProcessing* audio_processing_ = nullptr;
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// Capture side.
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// Thread-safe.
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const std::unique_ptr<AsyncAudioProcessing> async_audio_processing_;
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mutable Mutex capture_lock_;
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std::vector<AudioSender*> audio_senders_ 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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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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};
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} // namespace webrtc
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#endif // AUDIO_AUDIO_TRANSPORT_IMPL_H_
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