audio_transport_impl.cc | Explore in Territory

/*
 *  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.
 */

#include "audio/audio_transport_impl.h"

#include <algorithm>
#include <memory>
#include <utility>

#include "audio/remix_resample.h"
#include "audio/utility/audio_frame_operations.h"
#include "call/audio_sender.h"
#include "modules/async_audio_processing/async_audio_processing.h"
#include "modules/audio_processing/include/audio_frame_proxies.h"
#include "rtc_base/checks.h"
#include "rtc_base/trace_event.h"

namespace webrtc {

namespace {

// We want to process at the lowest sample rate and channel count possible
// without losing information. Choose the lowest native rate at least equal to
// the minimum of input and codec rates, choose lowest channel count, and
// configure the audio frame.
void InitializeCaptureFrame(int input_sample_rate,
                            int send_sample_rate_hz,
                            size_t input_num_channels,
                            size_t send_num_channels,
                            AudioFrame* audio_frame) { … }

void ProcessCaptureFrame(uint32_t delay_ms,
                         bool key_pressed,
                         bool swap_stereo_channels,
                         AudioProcessing* audio_processing,
                         AudioFrame* audio_frame) { … }

// Resample audio in `frame` to given sample rate preserving the
// channel count and place the result in `destination`.
int Resample(const AudioFrame& frame,
             const int destination_sample_rate,
             PushResampler<int16_t>* resampler,
             InterleavedView<int16_t> destination) { … }
}  // namespace

AudioTransportImpl::AudioTransportImpl(
    AudioMixer* mixer,
    AudioProcessing* audio_processing,
    AsyncAudioProcessing::Factory* async_audio_processing_factory)
    : … { … }

AudioTransportImpl::~AudioTransportImpl() { … }

int32_t AudioTransportImpl::RecordedDataIsAvailable(
    const void* audio_data,
    size_t number_of_frames,
    size_t bytes_per_sample,
    size_t number_of_channels,
    uint32_t sample_rate,
    uint32_t audio_delay_milliseconds,
    int32_t clock_drift,
    uint32_t volume,
    bool key_pressed,
    uint32_t& new_mic_volume) { … }

// Not used in Chromium. Process captured audio and distribute to all sending
// streams, and try to do this at the lowest possible sample rate.
int32_t AudioTransportImpl::RecordedDataIsAvailable(
    const void* audio_data,
    size_t number_of_frames,
    size_t bytes_per_sample,
    size_t number_of_channels,
    uint32_t sample_rate,
    uint32_t audio_delay_milliseconds,
    int32_t /*clock_drift*/,
    uint32_t /*volume*/,
    bool key_pressed,
    uint32_t& /*new_mic_volume*/,
    absl::optional<int64_t>
        estimated_capture_time_ns) { … }

void AudioTransportImpl::SendProcessedData(
    std::unique_ptr<AudioFrame> audio_frame) { … }

// Mix all received streams, feed the result to the AudioProcessing module, then
// resample the result to the requested output rate.
int32_t AudioTransportImpl::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) { … }

// Used by Chromium - same as NeedMorePlayData() but because Chrome has its
// own APM instance, does not call audio_processing_->ProcessReverseStream().
void AudioTransportImpl::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) { … }

void AudioTransportImpl::UpdateAudioSenders(std::vector<AudioSender*> senders,
                                            int send_sample_rate_hz,
                                            size_t send_num_channels) { … }

void AudioTransportImpl::SetStereoChannelSwapping(bool enable) { … }

}  // namespace webrtc
chromium/third_party/webrtc/audio/audio_transport_impl.cc
