256 lines
10 KiB
C++
256 lines
10 KiB
C++
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/*
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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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#include "audio/audio_transport_impl.h"
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#include <algorithm>
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#include <memory>
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#include <utility>
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#include "audio/remix_resample.h"
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#include "audio/utility/audio_frame_operations.h"
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#include "call/audio_sender.h"
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#include "modules/audio_processing/include/audio_frame_proxies.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace {
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// We want to process at the lowest sample rate and channel count possible
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// without losing information. Choose the lowest native rate at least equal to
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// the minimum of input and codec rates, choose lowest channel count, and
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// configure the audio frame.
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void InitializeCaptureFrame(int input_sample_rate,
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int send_sample_rate_hz,
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size_t input_num_channels,
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size_t send_num_channels,
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AudioFrame* audio_frame) {
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RTC_DCHECK(audio_frame);
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int min_processing_rate_hz = std::min(input_sample_rate, send_sample_rate_hz);
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for (int native_rate_hz : AudioProcessing::kNativeSampleRatesHz) {
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audio_frame->sample_rate_hz_ = native_rate_hz;
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if (audio_frame->sample_rate_hz_ >= min_processing_rate_hz) {
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break;
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}
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}
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audio_frame->num_channels_ = std::min(input_num_channels, send_num_channels);
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}
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void ProcessCaptureFrame(uint32_t delay_ms,
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bool key_pressed,
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bool swap_stereo_channels,
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AudioProcessing* audio_processing,
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AudioFrame* audio_frame) {
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RTC_DCHECK(audio_frame);
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if (audio_processing) {
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audio_processing->set_stream_delay_ms(delay_ms);
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audio_processing->set_stream_key_pressed(key_pressed);
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int error = ProcessAudioFrame(audio_processing, audio_frame);
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RTC_DCHECK_EQ(0, error) << "ProcessStream() error: " << error;
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}
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if (swap_stereo_channels) {
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AudioFrameOperations::SwapStereoChannels(audio_frame);
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}
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}
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// Resample audio in |frame| to given sample rate preserving the
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// channel count and place the result in |destination|.
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int Resample(const AudioFrame& frame,
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const int destination_sample_rate,
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PushResampler<int16_t>* resampler,
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int16_t* destination) {
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const int number_of_channels = static_cast<int>(frame.num_channels_);
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const int target_number_of_samples_per_channel =
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destination_sample_rate / 100;
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resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate,
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number_of_channels);
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// TODO(yujo): make resampler take an AudioFrame, and add special case
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// handling of muted frames.
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return resampler->Resample(
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frame.data(), frame.samples_per_channel_ * number_of_channels,
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destination, number_of_channels * target_number_of_samples_per_channel);
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}
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} // namespace
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AudioTransportImpl::AudioTransportImpl(AudioMixer* mixer,
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AudioProcessing* audio_processing)
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: audio_processing_(audio_processing), mixer_(mixer) {
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RTC_DCHECK(mixer);
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}
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AudioTransportImpl::~AudioTransportImpl() {}
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// Not used in Chromium. Process captured audio and distribute to all sending
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// streams, and try to do this at the lowest possible sample rate.
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int32_t AudioTransportImpl::RecordedDataIsAvailable(
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const void* audio_data,
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const size_t number_of_frames,
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const size_t bytes_per_sample,
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const size_t number_of_channels,
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const uint32_t sample_rate,
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const uint32_t audio_delay_milliseconds,
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const int32_t /*clock_drift*/,
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const uint32_t /*volume*/,
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const bool key_pressed,
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uint32_t& /*new_mic_volume*/) { // NOLINT: to avoid changing APIs
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RTC_DCHECK(audio_data);
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RTC_DCHECK_GE(number_of_channels, 1);
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RTC_DCHECK_LE(number_of_channels, 2);
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RTC_DCHECK_EQ(2 * number_of_channels, bytes_per_sample);
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RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);
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// 100 = 1 second / data duration (10 ms).
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RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);
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RTC_DCHECK_LE(bytes_per_sample * number_of_frames * number_of_channels,
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AudioFrame::kMaxDataSizeBytes);
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int send_sample_rate_hz = 0;
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size_t send_num_channels = 0;
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bool swap_stereo_channels = false;
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{
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MutexLock lock(&capture_lock_);
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send_sample_rate_hz = send_sample_rate_hz_;
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send_num_channels = send_num_channels_;
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swap_stereo_channels = swap_stereo_channels_;
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}
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std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
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InitializeCaptureFrame(sample_rate, send_sample_rate_hz, number_of_channels,
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send_num_channels, audio_frame.get());
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voe::RemixAndResample(static_cast<const int16_t*>(audio_data),
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number_of_frames, number_of_channels, sample_rate,
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&capture_resampler_, audio_frame.get());
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ProcessCaptureFrame(audio_delay_milliseconds, key_pressed,
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swap_stereo_channels, audio_processing_,
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audio_frame.get());
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// Typing detection (utilizes the APM/VAD decision). We let the VAD determine
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// if we're using this feature or not.
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// TODO(solenberg): GetConfig() takes a lock. Work around that.
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bool typing_detected = false;
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if (audio_processing_ &&
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audio_processing_->GetConfig().voice_detection.enabled) {
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if (audio_frame->vad_activity_ != AudioFrame::kVadUnknown) {
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bool vad_active = audio_frame->vad_activity_ == AudioFrame::kVadActive;
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typing_detected = typing_detection_.Process(key_pressed, vad_active);
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}
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}
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// Copy frame and push to each sending stream. The copy is required since an
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// encoding task will be posted internally to each stream.
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{
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MutexLock lock(&capture_lock_);
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typing_noise_detected_ = typing_detected;
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RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
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if (!audio_senders_.empty()) {
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auto it = audio_senders_.begin();
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while (++it != audio_senders_.end()) {
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std::unique_ptr<AudioFrame> audio_frame_copy(new AudioFrame());
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audio_frame_copy->CopyFrom(*audio_frame);
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(*it)->SendAudioData(std::move(audio_frame_copy));
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}
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// Send the original frame to the first stream w/o copying.
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(*audio_senders_.begin())->SendAudioData(std::move(audio_frame));
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}
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}
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return 0;
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}
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// Mix all received streams, feed the result to the AudioProcessing module, then
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// resample the result to the requested output rate.
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int32_t AudioTransportImpl::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) {
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RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
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RTC_DCHECK_GE(nChannels, 1);
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RTC_DCHECK_LE(nChannels, 2);
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RTC_DCHECK_GE(
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samplesPerSec,
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static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));
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// 100 = 1 second / data duration (10 ms).
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RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
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RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
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AudioFrame::kMaxDataSizeBytes);
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mixer_->Mix(nChannels, &mixed_frame_);
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*elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
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*ntp_time_ms = mixed_frame_.ntp_time_ms_;
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if (audio_processing_) {
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const auto error =
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ProcessReverseAudioFrame(audio_processing_, &mixed_frame_);
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RTC_DCHECK_EQ(error, AudioProcessing::kNoError);
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}
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nSamplesOut = Resample(mixed_frame_, samplesPerSec, &render_resampler_,
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static_cast<int16_t*>(audioSamples));
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RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples);
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return 0;
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}
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// Used by Chromium - same as NeedMorePlayData() but because Chrome has its
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// own APM instance, does not call audio_processing_->ProcessReverseStream().
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void AudioTransportImpl::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) {
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RTC_DCHECK_EQ(bits_per_sample, 16);
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RTC_DCHECK_GE(number_of_channels, 1);
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RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);
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// 100 = 1 second / data duration (10 ms).
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RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);
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// 8 = bits per byte.
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RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
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AudioFrame::kMaxDataSizeBytes);
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mixer_->Mix(number_of_channels, &mixed_frame_);
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*elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
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*ntp_time_ms = mixed_frame_.ntp_time_ms_;
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auto output_samples = Resample(mixed_frame_, sample_rate, &render_resampler_,
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static_cast<int16_t*>(audio_data));
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RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
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}
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void AudioTransportImpl::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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MutexLock lock(&capture_lock_);
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audio_senders_ = std::move(senders);
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send_sample_rate_hz_ = send_sample_rate_hz;
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send_num_channels_ = send_num_channels;
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}
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void AudioTransportImpl::SetStereoChannelSwapping(bool enable) {
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MutexLock lock(&capture_lock_);
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swap_stereo_channels_ = enable;
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}
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bool AudioTransportImpl::typing_noise_detected() const {
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MutexLock lock(&capture_lock_);
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return typing_noise_detected_;
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}
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} // namespace webrtc
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