modules/audio_mixer/frame_combiner.cc - src - Git at Google

 /*
  *  Copyright (c) 2017 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 "modules/audio_mixer/frame_combiner.h"

 #include <algorithm>
 #include <array>
 #include <cstdint>
 #include <iterator>
 #include <memory>
 #include <string>
 #include <utility>
 #include <vector>

 #include "api/array_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/rtp_packet_info.h"
 #include "api/rtp_packet_infos.h"
 #include "common_audio/include/audio_util.h"
 #include "modules/audio_mixer/audio_frame_manipulator.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
 #include "modules/audio_processing/include/audio_frame_view.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "system_wrappers/include/metrics.h"

 namespace webrtc {
 namespace {

 void SetAudioFrameFields(rtc::ArrayView<const AudioFrame* const> mix_list,
                          size_t number_of_channels,
                          int sample_rate,
                          size_t /* number_of_streams */,
                          AudioFrame* audio_frame_for_mixing) {
   const size_t samples_per_channel =
       SampleRateToDefaultChannelSize(sample_rate);

   // TODO(minyue): Issue bugs.webrtc.org/3390.
   // Audio frame timestamp. The 'timestamp_' field is set to dummy
   // value '0', because it is only supported in the one channel case and
   // is then updated in the helper functions.
   audio_frame_for_mixing->UpdateFrame(
       0, nullptr, samples_per_channel, sample_rate, AudioFrame::kUndefined,
       AudioFrame::kVadUnknown, number_of_channels);

   if (mix_list.empty()) {
     audio_frame_for_mixing->elapsed_time_ms_ = -1;
   } else {
     audio_frame_for_mixing->timestamp_ = mix_list[0]->timestamp_;
     audio_frame_for_mixing->elapsed_time_ms_ = mix_list[0]->elapsed_time_ms_;
     audio_frame_for_mixing->ntp_time_ms_ = mix_list[0]->ntp_time_ms_;
     std::vector<RtpPacketInfo> packet_infos;
     for (const auto& frame : mix_list) {
       audio_frame_for_mixing->timestamp_ =
           std::min(audio_frame_for_mixing->timestamp_, frame->timestamp_);
       audio_frame_for_mixing->ntp_time_ms_ =
           std::min(audio_frame_for_mixing->ntp_time_ms_, frame->ntp_time_ms_);
       audio_frame_for_mixing->elapsed_time_ms_ = std::max(
           audio_frame_for_mixing->elapsed_time_ms_, frame->elapsed_time_ms_);
       packet_infos.insert(packet_infos.end(), frame->packet_infos_.begin(),
                           frame->packet_infos_.end());
     }
     audio_frame_for_mixing->packet_infos_ =
         RtpPacketInfos(std::move(packet_infos));
   }
 }

 void MixFewFramesWithNoLimiter(rtc::ArrayView<const AudioFrame* const> mix_list,
                                AudioFrame* audio_frame_for_mixing) {
   if (mix_list.empty()) {
     audio_frame_for_mixing->Mute();
     return;
   }
   RTC_DCHECK_LE(mix_list.size(), 1);
   InterleavedView<int16_t> dst = audio_frame_for_mixing->mutable_data(
       mix_list[0]->samples_per_channel_, mix_list[0]->num_channels_);
   CopySamples(dst, mix_list[0]->data_view());
 }

 void MixToFloatFrame(rtc::ArrayView<const AudioFrame* const> mix_list,
                      DeinterleavedView<float>& mixing_buffer) {
   const size_t number_of_channels = NumChannels(mixing_buffer);
   // Clear the mixing buffer.
   rtc::ArrayView<float> raw_data = mixing_buffer.data();
   ClearSamples(raw_data);

   // Convert to FloatS16 and mix.
   for (size_t i = 0; i < mix_list.size(); ++i) {
     InterleavedView<const int16_t> frame_data = mix_list[i]->data_view();
     RTC_CHECK(!frame_data.empty());
     for (size_t j = 0; j < number_of_channels; ++j) {
       MonoView<float> channel = mixing_buffer[j];
       for (size_t k = 0; k < SamplesPerChannel(channel); ++k) {
         channel[k] += frame_data[number_of_channels * k + j];
       }
     }
   }
 }

 void RunLimiter(DeinterleavedView<float> deinterleaved, Limiter* limiter) {
   limiter->SetSamplesPerChannel(deinterleaved.samples_per_channel());
   limiter->Process(deinterleaved);
 }

 // Both interleaves and rounds.
 void InterleaveToAudioFrame(DeinterleavedView<float> deinterleaved,
                             AudioFrame* audio_frame_for_mixing) {
   InterleavedView<int16_t> mixing_data = audio_frame_for_mixing->mutable_data(
       deinterleaved.samples_per_channel(), deinterleaved.num_channels());
   // Put data in the result frame.
   for (size_t i = 0; i < mixing_data.num_channels(); ++i) {
     auto channel = deinterleaved[i];
     for (size_t j = 0; j < mixing_data.samples_per_channel(); ++j) {
       mixing_data[mixing_data.num_channels() * j + i] =
           FloatS16ToS16(channel[j]);
     }
   }
 }
 }  // namespace

 constexpr size_t FrameCombiner::kMaximumNumberOfChannels;
 constexpr size_t FrameCombiner::kMaximumChannelSize;

 FrameCombiner::FrameCombiner(bool use_limiter)
     : data_dumper_(new ApmDataDumper(0)),
       limiter_(data_dumper_.get(), kMaximumChannelSize, "AudioMixer"),
       use_limiter_(use_limiter) {
   static_assert(kMaximumChannelSize * kMaximumNumberOfChannels <=
                     AudioFrame::kMaxDataSizeSamples,
                 "");
 }

 FrameCombiner::~FrameCombiner() = default;

 void FrameCombiner::Combine(rtc::ArrayView<AudioFrame* const> mix_list,
                             size_t number_of_channels,
                             int sample_rate,
                             size_t number_of_streams,
                             AudioFrame* audio_frame_for_mixing) {
   RTC_DCHECK(audio_frame_for_mixing);
   RTC_DCHECK_GT(sample_rate, 0);

   // Note: `mix_list` is allowed to be empty.
   // See FrameCombiner.CombiningZeroFramesShouldProduceSilence.

   // Make sure to cap `number_of_channels` to the kMaximumNumberOfChannels
   // limits since processing from hereon out will be bound by them.
   number_of_channels = std::min(number_of_channels, kMaximumNumberOfChannels);

   SetAudioFrameFields(mix_list, number_of_channels, sample_rate,
                       number_of_streams, audio_frame_for_mixing);

   size_t samples_per_channel = SampleRateToDefaultChannelSize(sample_rate);

 #if RTC_DCHECK_IS_ON
   for (const auto* frame : mix_list) {
     RTC_DCHECK_EQ(samples_per_channel, frame->samples_per_channel_);
     RTC_DCHECK_EQ(sample_rate, frame->sample_rate_hz_);
   }
 #endif

   // The 'num_channels_' field of frames in 'mix_list' could be
   // different from 'number_of_channels'.
   for (auto* frame : mix_list) {
     RemixFrame(number_of_channels, frame);
   }

   if (number_of_streams <= 1) {
     MixFewFramesWithNoLimiter(mix_list, audio_frame_for_mixing);
     return;
   }

   // Make sure that the size of the view based on the desired
   // `samples_per_channel` and `number_of_channels` doesn't exceed the size of
   // the `mixing_buffer_` buffer.
   RTC_DCHECK_LE(samples_per_channel, kMaximumChannelSize);
   // Since the above check is a DCHECK only, clamp down on `samples_per_channel`
   // to make sure we don't exceed the buffer size in non-dcheck builds.
   // See also FrameCombinerDeathTest.DebugBuildCrashesWithHighRate.
   samples_per_channel = std::min(samples_per_channel, kMaximumChannelSize);
   DeinterleavedView<float> deinterleaved(
       mixing_buffer_.data(), samples_per_channel, number_of_channels);
   MixToFloatFrame(mix_list, deinterleaved);

   if (use_limiter_) {
     RunLimiter(deinterleaved, &limiter_);
   }

   InterleaveToAudioFrame(deinterleaved, audio_frame_for_mixing);
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2017 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 "modules/audio_mixer/frame_combiner.h"

	#include <algorithm>
	#include <array>
	#include <cstdint>
	#include <iterator>
	#include <memory>
	#include <string>
	#include <utility>
	#include <vector>

	#include "api/array_view.h"
	#include "api/audio/audio_processing.h"
	#include "api/rtp_packet_info.h"
	#include "api/rtp_packet_infos.h"
	#include "common_audio/include/audio_util.h"
	#include "modules/audio_mixer/audio_frame_manipulator.h"
	#include "modules/audio_mixer/audio_mixer_impl.h"
	#include "modules/audio_processing/include/audio_frame_view.h"
	#include "modules/audio_processing/logging/apm_data_dumper.h"
	#include "rtc_base/arraysize.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/numerics/safe_conversions.h"
	#include "system_wrappers/include/metrics.h"

	namespace webrtc {
	namespace {

	void SetAudioFrameFields(rtc::ArrayView<const AudioFrame* const> mix_list,
	size_t number_of_channels,
	int sample_rate,
	size_t /* number_of_streams */,
	AudioFrame* audio_frame_for_mixing) {
	const size_t samples_per_channel =
	SampleRateToDefaultChannelSize(sample_rate);

	// TODO(minyue): Issue bugs.webrtc.org/3390.
	// Audio frame timestamp. The 'timestamp_' field is set to dummy
	// value '0', because it is only supported in the one channel case and
	// is then updated in the helper functions.
	audio_frame_for_mixing->UpdateFrame(
	0, nullptr, samples_per_channel, sample_rate, AudioFrame::kUndefined,
	AudioFrame::kVadUnknown, number_of_channels);

	if (mix_list.empty()) {
	audio_frame_for_mixing->elapsed_time_ms_ = -1;
	} else {
	audio_frame_for_mixing->timestamp_ = mix_list[0]->timestamp_;
	audio_frame_for_mixing->elapsed_time_ms_ = mix_list[0]->elapsed_time_ms_;
	audio_frame_for_mixing->ntp_time_ms_ = mix_list[0]->ntp_time_ms_;
	std::vector<RtpPacketInfo> packet_infos;
	for (const auto& frame : mix_list) {
	audio_frame_for_mixing->timestamp_ =
	std::min(audio_frame_for_mixing->timestamp_, frame->timestamp_);
	audio_frame_for_mixing->ntp_time_ms_ =
	std::min(audio_frame_for_mixing->ntp_time_ms_, frame->ntp_time_ms_);
	audio_frame_for_mixing->elapsed_time_ms_ = std::max(
	audio_frame_for_mixing->elapsed_time_ms_, frame->elapsed_time_ms_);
	packet_infos.insert(packet_infos.end(), frame->packet_infos_.begin(),
	frame->packet_infos_.end());
	}
	audio_frame_for_mixing->packet_infos_ =
	RtpPacketInfos(std::move(packet_infos));
	}
	}

	void MixFewFramesWithNoLimiter(rtc::ArrayView<const AudioFrame* const> mix_list,
	AudioFrame* audio_frame_for_mixing) {
	if (mix_list.empty()) {
	audio_frame_for_mixing->Mute();
	return;
	}
	RTC_DCHECK_LE(mix_list.size(), 1);
	InterleavedView<int16_t> dst = audio_frame_for_mixing->mutable_data(
	mix_list[0]->samples_per_channel_, mix_list[0]->num_channels_);
	CopySamples(dst, mix_list[0]->data_view());
	}

	void MixToFloatFrame(rtc::ArrayView<const AudioFrame* const> mix_list,
	DeinterleavedView<float>& mixing_buffer) {
	const size_t number_of_channels = NumChannels(mixing_buffer);
	// Clear the mixing buffer.
	rtc::ArrayView<float> raw_data = mixing_buffer.data();
	ClearSamples(raw_data);

	// Convert to FloatS16 and mix.
	for (size_t i = 0; i < mix_list.size(); ++i) {
	InterleavedView<const int16_t> frame_data = mix_list[i]->data_view();
	RTC_CHECK(!frame_data.empty());
	for (size_t j = 0; j < number_of_channels; ++j) {
	MonoView<float> channel = mixing_buffer[j];
	for (size_t k = 0; k < SamplesPerChannel(channel); ++k) {
	channel[k] += frame_data[number_of_channels * k + j];
	}
	}
	}
	}

	void RunLimiter(DeinterleavedView<float> deinterleaved, Limiter* limiter) {
	limiter->SetSamplesPerChannel(deinterleaved.samples_per_channel());
	limiter->Process(deinterleaved);
	}

	// Both interleaves and rounds.
	void InterleaveToAudioFrame(DeinterleavedView<float> deinterleaved,
	AudioFrame* audio_frame_for_mixing) {
	InterleavedView<int16_t> mixing_data = audio_frame_for_mixing->mutable_data(
	deinterleaved.samples_per_channel(), deinterleaved.num_channels());
	// Put data in the result frame.
	for (size_t i = 0; i < mixing_data.num_channels(); ++i) {
	auto channel = deinterleaved[i];
	for (size_t j = 0; j < mixing_data.samples_per_channel(); ++j) {
	mixing_data[mixing_data.num_channels() * j + i] =
	FloatS16ToS16(channel[j]);
	}
	}
	}
	} // namespace

	constexpr size_t FrameCombiner::kMaximumNumberOfChannels;
	constexpr size_t FrameCombiner::kMaximumChannelSize;

	FrameCombiner::FrameCombiner(bool use_limiter)
	: data_dumper_(new ApmDataDumper(0)),
	limiter_(data_dumper_.get(), kMaximumChannelSize, "AudioMixer"),
	use_limiter_(use_limiter) {
	static_assert(kMaximumChannelSize * kMaximumNumberOfChannels <=
	AudioFrame::kMaxDataSizeSamples,
	"");
	}

	FrameCombiner::~FrameCombiner() = default;

	void FrameCombiner::Combine(rtc::ArrayView<AudioFrame* const> mix_list,
	size_t number_of_channels,
	int sample_rate,
	size_t number_of_streams,
	AudioFrame* audio_frame_for_mixing) {
	RTC_DCHECK(audio_frame_for_mixing);
	RTC_DCHECK_GT(sample_rate, 0);

	// Note: `mix_list` is allowed to be empty.
	// See FrameCombiner.CombiningZeroFramesShouldProduceSilence.

	// Make sure to cap `number_of_channels` to the kMaximumNumberOfChannels
	// limits since processing from hereon out will be bound by them.
	number_of_channels = std::min(number_of_channels, kMaximumNumberOfChannels);

	SetAudioFrameFields(mix_list, number_of_channels, sample_rate,
	number_of_streams, audio_frame_for_mixing);

	size_t samples_per_channel = SampleRateToDefaultChannelSize(sample_rate);

	#if RTC_DCHECK_IS_ON
	for (const auto* frame : mix_list) {
	RTC_DCHECK_EQ(samples_per_channel, frame->samples_per_channel_);
	RTC_DCHECK_EQ(sample_rate, frame->sample_rate_hz_);
	}
	#endif

	// The 'num_channels_' field of frames in 'mix_list' could be
	// different from 'number_of_channels'.
	for (auto* frame : mix_list) {
	RemixFrame(number_of_channels, frame);
	}

	if (number_of_streams <= 1) {
	MixFewFramesWithNoLimiter(mix_list, audio_frame_for_mixing);
	return;
	}

	// Make sure that the size of the view based on the desired
	// `samples_per_channel` and `number_of_channels` doesn't exceed the size of
	// the `mixing_buffer_` buffer.
	RTC_DCHECK_LE(samples_per_channel, kMaximumChannelSize);
	// Since the above check is a DCHECK only, clamp down on `samples_per_channel`
	// to make sure we don't exceed the buffer size in non-dcheck builds.
	// See also FrameCombinerDeathTest.DebugBuildCrashesWithHighRate.
	samples_per_channel = std::min(samples_per_channel, kMaximumChannelSize);
	DeinterleavedView<float> deinterleaved(
	mixing_buffer_.data(), samples_per_channel, number_of_channels);
	MixToFloatFrame(mix_list, deinterleaved);

	if (use_limiter_) {
	RunLimiter(deinterleaved, &limiter_);
	}

	InterleaveToAudioFrame(deinterleaved, audio_frame_for_mixing);
	}

	} // namespace webrtc