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/*
* 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 <memory>
#include <vector>
#include "api/audio/audio_mixer.h"
#include "api/scoped_refptr.h"
#include "common_audio/resampler/include/push_resampler.h"
#include "modules/async_audio_processing/async_audio_processing.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class AudioSender;
class AudioTransportImpl : public AudioTransport {
public:
AudioTransportImpl(
AudioMixer* mixer,
AudioProcessing* audio_processing,
AsyncAudioProcessing::Factory* async_audio_processing_factory);
AudioTransportImpl() = delete;
AudioTransportImpl(const AudioTransportImpl&) = delete;
AudioTransportImpl& operator=(const AudioTransportImpl&) = delete;
~AudioTransportImpl() override;
// TODO(bugs.webrtc.org/13620) Deprecate this function
int32_t RecordedDataIsAvailable(const void* audioSamples,
size_t nSamples,
size_t nBytesPerSample,
size_t nChannels,
uint32_t samplesPerSec,
uint32_t totalDelayMS,
int32_t clockDrift,
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel) override;
int32_t RecordedDataIsAvailable(
const void* audioSamples,
size_t nSamples,
size_t nBytesPerSample,
size_t nChannels,
uint32_t samplesPerSec,
uint32_t totalDelayMS,
int32_t clockDrift,
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel,
absl::optional<int64_t> estimated_capture_time_ns) override;
int32_t NeedMorePlayData(size_t nSamples,
size_t nBytesPerSample,
size_t nChannels,
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 UpdateAudioSenders(std::vector<AudioSender*> senders,
int send_sample_rate_hz,
size_t send_num_channels);
void SetStereoChannelSwapping(bool enable);
private:
void SendProcessedData(std::unique_ptr<AudioFrame> audio_frame);
// Shared.
AudioProcessing* audio_processing_ = nullptr;
// Capture side.
// Thread-safe.
const std::unique_ptr<AsyncAudioProcessing> async_audio_processing_;
mutable Mutex capture_lock_;
std::vector<AudioSender*> audio_senders_ 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 swap_stereo_channels_ RTC_GUARDED_BY(capture_lock_) = false;
PushResampler<int16_t> capture_resampler_;
// Render side.
rtc::scoped_refptr<AudioMixer> mixer_;
AudioFrame mixed_frame_;
// Converts mixed audio to the audio device output rate.
PushResampler<int16_t> render_resampler_;
};
} // namespace webrtc
#endif // AUDIO_AUDIO_TRANSPORT_IMPL_H_
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