diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/audio/audio_transport_impl.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/audio_transport_impl.h')
-rw-r--r-- | third_party/libwebrtc/audio/audio_transport_impl.h | 117 |
1 files changed, 117 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/audio_transport_impl.h b/third_party/libwebrtc/audio/audio_transport_impl.h new file mode 100644 index 0000000000..24b09d2140 --- /dev/null +++ b/third_party/libwebrtc/audio/audio_transport_impl.h @@ -0,0 +1,117 @@ +/* + * 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_ |