audio/audio_transport_impl.cc - src - Git at Google

 /*
  *  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_send_stream.h"
 #include "rtc_base/checks.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) {
   RTC_DCHECK(audio_frame);
   int min_processing_rate_hz = std::min(input_sample_rate, send_sample_rate_hz);
   for (int native_rate_hz : AudioProcessing::kNativeSampleRatesHz) {
     audio_frame->sample_rate_hz_ = native_rate_hz;
     if (audio_frame->sample_rate_hz_ >= min_processing_rate_hz) {
       break;
     }
   }
   audio_frame->num_channels_ = std::min(input_num_channels, send_num_channels);
 }

 void ProcessCaptureFrame(uint32_t delay_ms,
                          bool key_pressed,
                          bool swap_stereo_channels,
                          AudioProcessing* audio_processing,
                          AudioFrame* audio_frame) {
   RTC_DCHECK(audio_processing);
   RTC_DCHECK(audio_frame);
   audio_processing->set_stream_delay_ms(delay_ms);
   audio_processing->set_stream_key_pressed(key_pressed);
   int error = audio_processing->ProcessStream(audio_frame);
   RTC_DCHECK_EQ(0, error) << "ProcessStream() error: " << error;
   if (swap_stereo_channels) {
     AudioFrameOperations::SwapStereoChannels(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,
              int16_t* destination) {
   const int number_of_channels = static_cast<int>(frame.num_channels_);
   const int target_number_of_samples_per_channel =
       destination_sample_rate / 100;
   resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate,
                                 number_of_channels);

   // TODO(yujo): make resampler take an AudioFrame, and add special case
   // handling of muted frames.
   return resampler->Resample(
       frame.data(), frame.samples_per_channel_ * number_of_channels,
       destination, number_of_channels * target_number_of_samples_per_channel);
 }
 }  // namespace

 AudioTransportImpl::AudioTransportImpl(AudioMixer* mixer,
                                        AudioProcessing* audio_processing)
     : audio_processing_(audio_processing), mixer_(mixer) {
   RTC_DCHECK(mixer);
   RTC_DCHECK(audio_processing);
 }

 AudioTransportImpl::~AudioTransportImpl() {}

 // 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,
     const size_t number_of_frames,
     const size_t bytes_per_sample,
     const size_t number_of_channels,
     const uint32_t sample_rate,
     const uint32_t audio_delay_milliseconds,
     const int32_t /*clock_drift*/,
     const uint32_t /*volume*/,
     const bool key_pressed,
     uint32_t& /*new_mic_volume*/) {  // NOLINT: to avoid changing APIs
   RTC_DCHECK(audio_data);
   RTC_DCHECK_GE(number_of_channels, 1);
   RTC_DCHECK_LE(number_of_channels, 2);
   RTC_DCHECK_EQ(2 * number_of_channels, bytes_per_sample);
   RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);
   // 100 = 1 second / data duration (10 ms).
   RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);
   RTC_DCHECK_LE(bytes_per_sample * number_of_frames * number_of_channels,
                 AudioFrame::kMaxDataSizeBytes);

   int send_sample_rate_hz = 0;
   size_t send_num_channels = 0;
   bool swap_stereo_channels = false;
   {
     rtc::CritScope lock(&capture_lock_);
     send_sample_rate_hz = send_sample_rate_hz_;
     send_num_channels = send_num_channels_;
     swap_stereo_channels = swap_stereo_channels_;
   }

   std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
   InitializeCaptureFrame(sample_rate, send_sample_rate_hz, number_of_channels,
                          send_num_channels, audio_frame.get());
   voe::RemixAndResample(static_cast<const int16_t*>(audio_data),
                         number_of_frames, number_of_channels, sample_rate,
                         &capture_resampler_, audio_frame.get());
   ProcessCaptureFrame(audio_delay_milliseconds, key_pressed,
                       swap_stereo_channels, audio_processing_,
                       audio_frame.get());

   // Typing detection (utilizes the APM/VAD decision). We let the VAD determine
   // if we're using this feature or not.
   // TODO(solenberg): is_enabled() takes a lock. Work around that.
   bool typing_detected = false;
   if (audio_processing_->voice_detection()->is_enabled()) {
     if (audio_frame->vad_activity_ != AudioFrame::kVadUnknown) {
       bool vad_active = audio_frame->vad_activity_ == AudioFrame::kVadActive;
       typing_detected = typing_detection_.Process(key_pressed, vad_active);
     }
   }

   // Measure audio level of speech after all processing.
   double sample_duration = static_cast<double>(number_of_frames) / sample_rate;
   audio_level_.ComputeLevel(*audio_frame.get(), sample_duration);

   // Copy frame and push to each sending stream. The copy is required since an
   // encoding task will be posted internally to each stream.
   {
     rtc::CritScope lock(&capture_lock_);
     typing_noise_detected_ = typing_detected;

     RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
     if (!sending_streams_.empty()) {
       auto it = sending_streams_.begin();
       while (++it != sending_streams_.end()) {
         std::unique_ptr<AudioFrame> audio_frame_copy(new AudioFrame());
         audio_frame_copy->CopyFrom(*audio_frame.get());
         (*it)->SendAudioData(std::move(audio_frame_copy));
       }
       // Send the original frame to the first stream w/o copying.
       (*sending_streams_.begin())->SendAudioData(std::move(audio_frame));
     }
   }

   return 0;
 }

 // 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) {
   RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
   RTC_DCHECK_GE(nChannels, 1);
   RTC_DCHECK_LE(nChannels, 2);
   RTC_DCHECK_GE(
       samplesPerSec,
       static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));

   // 100 = 1 second / data duration (10 ms).
   RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
   RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
                 AudioFrame::kMaxDataSizeBytes);

   mixer_->Mix(nChannels, &mixed_frame_);
   *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
   *ntp_time_ms = mixed_frame_.ntp_time_ms_;

   const auto error = audio_processing_->ProcessReverseStream(&mixed_frame_);
   RTC_DCHECK_EQ(error, AudioProcessing::kNoError);

   nSamplesOut = Resample(mixed_frame_, samplesPerSec, &render_resampler_,
                          static_cast<int16_t*>(audioSamples));
   RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples);
   return 0;
 }

 // 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) {
   RTC_DCHECK_EQ(bits_per_sample, 16);
   RTC_DCHECK_GE(number_of_channels, 1);
   RTC_DCHECK_LE(number_of_channels, 2);
   RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);

   // 100 = 1 second / data duration (10 ms).
   RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);

   // 8 = bits per byte.
   RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
                 AudioFrame::kMaxDataSizeBytes);
   mixer_->Mix(number_of_channels, &mixed_frame_);
   *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
   *ntp_time_ms = mixed_frame_.ntp_time_ms_;

   auto output_samples = Resample(mixed_frame_, sample_rate, &render_resampler_,
                                  static_cast<int16_t*>(audio_data));
   RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
 }

 void AudioTransportImpl::UpdateSendingStreams(
     std::vector<AudioSendStream*> streams,
     int send_sample_rate_hz,
     size_t send_num_channels) {
   rtc::CritScope lock(&capture_lock_);
   sending_streams_ = std::move(streams);
   send_sample_rate_hz_ = send_sample_rate_hz;
   send_num_channels_ = send_num_channels;
 }

 void AudioTransportImpl::SetStereoChannelSwapping(bool enable) {
   rtc::CritScope lock(&capture_lock_);
   swap_stereo_channels_ = enable;
 }

 bool AudioTransportImpl::typing_noise_detected() const {
   rtc::CritScope lock(&capture_lock_);
   return typing_noise_detected_;
 }
 }  // namespace webrtc
	/*
	* 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_send_stream.h"
	#include "rtc_base/checks.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) {
	RTC_DCHECK(audio_frame);
	int min_processing_rate_hz = std::min(input_sample_rate, send_sample_rate_hz);
	for (int native_rate_hz : AudioProcessing::kNativeSampleRatesHz) {
	audio_frame->sample_rate_hz_ = native_rate_hz;
	if (audio_frame->sample_rate_hz_ >= min_processing_rate_hz) {
	break;
	}
	}
	audio_frame->num_channels_ = std::min(input_num_channels, send_num_channels);
	}

	void ProcessCaptureFrame(uint32_t delay_ms,
	bool key_pressed,
	bool swap_stereo_channels,
	AudioProcessing* audio_processing,
	AudioFrame* audio_frame) {
	RTC_DCHECK(audio_processing);
	RTC_DCHECK(audio_frame);
	audio_processing->set_stream_delay_ms(delay_ms);
	audio_processing->set_stream_key_pressed(key_pressed);
	int error = audio_processing->ProcessStream(audio_frame);
	RTC_DCHECK_EQ(0, error) << "ProcessStream() error: " << error;
	if (swap_stereo_channels) {
	AudioFrameOperations::SwapStereoChannels(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,
	int16_t* destination) {
	const int number_of_channels = static_cast<int>(frame.num_channels_);
	const int target_number_of_samples_per_channel =
	destination_sample_rate / 100;
	resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate,
	number_of_channels);

	// TODO(yujo): make resampler take an AudioFrame, and add special case
	// handling of muted frames.
	return resampler->Resample(
	frame.data(), frame.samples_per_channel_ * number_of_channels,
	destination, number_of_channels * target_number_of_samples_per_channel);
	}
	} // namespace

	AudioTransportImpl::AudioTransportImpl(AudioMixer* mixer,
	AudioProcessing* audio_processing)
	: audio_processing_(audio_processing), mixer_(mixer) {
	RTC_DCHECK(mixer);
	RTC_DCHECK(audio_processing);
	}

	AudioTransportImpl::~AudioTransportImpl() {}

	// 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,
	const size_t number_of_frames,
	const size_t bytes_per_sample,
	const size_t number_of_channels,
	const uint32_t sample_rate,
	const uint32_t audio_delay_milliseconds,
	const int32_t /clock_drift/,
	const uint32_t /volume/,
	const bool key_pressed,
	uint32_t& /new_mic_volume/) { // NOLINT: to avoid changing APIs
	RTC_DCHECK(audio_data);
	RTC_DCHECK_GE(number_of_channels, 1);
	RTC_DCHECK_LE(number_of_channels, 2);
	RTC_DCHECK_EQ(2 * number_of_channels, bytes_per_sample);
	RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);
	// 100 = 1 second / data duration (10 ms).
	RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);
	RTC_DCHECK_LE(bytes_per_sample * number_of_frames * number_of_channels,
	AudioFrame::kMaxDataSizeBytes);

	int send_sample_rate_hz = 0;
	size_t send_num_channels = 0;
	bool swap_stereo_channels = false;
	{
	rtc::CritScope lock(&capture_lock_);
	send_sample_rate_hz = send_sample_rate_hz_;
	send_num_channels = send_num_channels_;
	swap_stereo_channels = swap_stereo_channels_;
	}

	std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
	InitializeCaptureFrame(sample_rate, send_sample_rate_hz, number_of_channels,
	send_num_channels, audio_frame.get());
	voe::RemixAndResample(static_cast<const int16_t*>(audio_data),
	number_of_frames, number_of_channels, sample_rate,
	&capture_resampler_, audio_frame.get());
	ProcessCaptureFrame(audio_delay_milliseconds, key_pressed,
	swap_stereo_channels, audio_processing_,
	audio_frame.get());

	// Typing detection (utilizes the APM/VAD decision). We let the VAD determine
	// if we're using this feature or not.
	// TODO(solenberg): is_enabled() takes a lock. Work around that.
	bool typing_detected = false;
	if (audio_processing_->voice_detection()->is_enabled()) {
	if (audio_frame->vad_activity_ != AudioFrame::kVadUnknown) {
	bool vad_active = audio_frame->vad_activity_ == AudioFrame::kVadActive;
	typing_detected = typing_detection_.Process(key_pressed, vad_active);
	}
	}

	// Measure audio level of speech after all processing.
	double sample_duration = static_cast<double>(number_of_frames) / sample_rate;
	audio_level_.ComputeLevel(*audio_frame.get(), sample_duration);

	// Copy frame and push to each sending stream. The copy is required since an
	// encoding task will be posted internally to each stream.
	{
	rtc::CritScope lock(&capture_lock_);
	typing_noise_detected_ = typing_detected;

	RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
	if (!sending_streams_.empty()) {
	auto it = sending_streams_.begin();
	while (++it != sending_streams_.end()) {
	std::unique_ptr<AudioFrame> audio_frame_copy(new AudioFrame());
	audio_frame_copy->CopyFrom(*audio_frame.get());
	(*it)->SendAudioData(std::move(audio_frame_copy));
	}
	// Send the original frame to the first stream w/o copying.
	(*sending_streams_.begin())->SendAudioData(std::move(audio_frame));
	}
	}

	return 0;
	}

	// 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) {
	RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
	RTC_DCHECK_GE(nChannels, 1);
	RTC_DCHECK_LE(nChannels, 2);
	RTC_DCHECK_GE(
	samplesPerSec,
	static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));

	// 100 = 1 second / data duration (10 ms).
	RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
	RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
	AudioFrame::kMaxDataSizeBytes);

	mixer_->Mix(nChannels, &mixed_frame_);
	*elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
	*ntp_time_ms = mixed_frame_.ntp_time_ms_;

	const auto error = audio_processing_->ProcessReverseStream(&mixed_frame_);
	RTC_DCHECK_EQ(error, AudioProcessing::kNoError);

	nSamplesOut = Resample(mixed_frame_, samplesPerSec, &render_resampler_,
	static_cast<int16_t*>(audioSamples));
	RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples);
	return 0;
	}

	// 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) {
	RTC_DCHECK_EQ(bits_per_sample, 16);
	RTC_DCHECK_GE(number_of_channels, 1);
	RTC_DCHECK_LE(number_of_channels, 2);
	RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);

	// 100 = 1 second / data duration (10 ms).
	RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);

	// 8 = bits per byte.
	RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
	AudioFrame::kMaxDataSizeBytes);
	mixer_->Mix(number_of_channels, &mixed_frame_);
	*elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
	*ntp_time_ms = mixed_frame_.ntp_time_ms_;

	auto output_samples = Resample(mixed_frame_, sample_rate, &render_resampler_,
	static_cast<int16_t*>(audio_data));
	RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
	}

	void AudioTransportImpl::UpdateSendingStreams(
	std::vector<AudioSendStream*> streams,
	int send_sample_rate_hz,
	size_t send_num_channels) {
	rtc::CritScope lock(&capture_lock_);
	sending_streams_ = std::move(streams);
	send_sample_rate_hz_ = send_sample_rate_hz;
	send_num_channels_ = send_num_channels;
	}

	void AudioTransportImpl::SetStereoChannelSwapping(bool enable) {
	rtc::CritScope lock(&capture_lock_);
	swap_stereo_channels_ = enable;
	}

	bool AudioTransportImpl::typing_noise_detected() const {
	rtc::CritScope lock(&capture_lock_);
	return typing_noise_detected_;
	}
	} // namespace webrtc