chromecast/media/cma/backend/mixer_input.cc - chromium/src - Git at Google

 // Copyright 2018 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.

 #include "chromecast/media/cma/backend/mixer_input.h"

 #include <stdint.h>

 #include <algorithm>
 #include <cmath>
 #include <utility>

 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/logging.h"
 #include "chromecast/media/cma/backend/audio_fader.h"
 #include "chromecast/media/cma/backend/audio_output_redirector_input.h"
 #include "chromecast/media/cma/backend/filter_group.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_timestamp_helper.h"
 #include "media/base/multi_channel_resampler.h"

 namespace chromecast {
 namespace media {

 namespace {

 const int64_t kMicrosecondsPerSecond = 1000 * 1000;
 const int kDefaultSlewTimeMs = 15;

 int RoundUpMultiple(int value, int multiple) {
   return multiple * ((value + (multiple - 1)) / multiple);
 }

 }  // namespace

 MixerInput::MixerInput(Source* source, FilterGroup* filter_group)
     : source_(source),
       num_channels_(source->num_channels()),
       input_samples_per_second_(source->input_samples_per_second()),
       output_samples_per_second_(filter_group->input_samples_per_second()),
       primary_(source->primary()),
       device_id_(source->device_id()),
       content_type_(source->content_type()),
       stream_volume_multiplier_(1.0f),
       type_volume_multiplier_(1.0f),
       mute_volume_multiplier_(1.0f),
       slew_volume_(kDefaultSlewTimeMs),
       volume_applied_(false),
       previous_ended_in_silence_(false),
       first_buffer_(true),
       resampler_buffered_frames_(0.0) {
   DCHECK(source_);
   DCHECK_GT(num_channels_, 0);
   DCHECK_GT(input_samples_per_second_, 0);

   MediaPipelineBackend::AudioDecoder::RenderingDelay initial_rendering_delay =
       filter_group->GetRenderingDelayToOutput();

   int source_read_size = filter_group->input_frames_per_write();
   if (output_samples_per_second_ > 0 &&
       output_samples_per_second_ != input_samples_per_second_) {
     // Round up to nearest multiple of SincResampler::kKernelSize. The read size
     // must be > kKernelSize, so we round up to at least 2 * kKernelSize.
     source_read_size = std::max(source_->desired_read_size(),
                                 ::media::SincResampler::kKernelSize + 1);
     source_read_size =
         RoundUpMultiple(source_read_size, ::media::SincResampler::kKernelSize);
     double resample_ratio = static_cast<double>(input_samples_per_second_) /
                             output_samples_per_second_;
     resampler_ = std::make_unique<::media::MultiChannelResampler>(
         num_channels_, resample_ratio, source_read_size,
         base::BindRepeating(&MixerInput::ResamplerReadCallback,
                             base::Unretained(this)));
     resampler_->PrimeWithSilence();

     double resampler_queued_frames = resampler_->BufferedFrames();
     initial_rendering_delay.delay_microseconds +=
         static_cast<int64_t>(resampler_queued_frames * kMicrosecondsPerSecond /
                              input_samples_per_second_);
   }

   if (output_samples_per_second_ != 0) {
     // If output_samples_per_second_ is 0, this stream will be unusable.
     // OnError() will be called shortly.
     slew_volume_.SetSampleRate(output_samples_per_second_);
   }
   source_->InitializeAudioPlayback(source_read_size, initial_rendering_delay);

   SetFilterGroup(filter_group);
 }

 MixerInput::~MixerInput() {
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   SetFilterGroup(nullptr);
   source_->FinalizeAudioPlayback();
 }

 void MixerInput::SetFilterGroup(FilterGroup* filter_group) {
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   DCHECK(!filter_group || !filter_group_);

   if (filter_group == filter_group_) {
     return;
   }
   if (filter_group_) {
     filter_group_->RemoveInput(this);
   }
   if (filter_group) {
     filter_group->AddInput(this);
   }
   filter_group_ = filter_group;
 }

 void MixerInput::AddAudioOutputRedirector(
     AudioOutputRedirectorInput* redirector) {
   LOG(INFO) << "Add redirector to " << device_id_ << "(" << source_ << ")";
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   DCHECK(redirector);
   audio_output_redirectors_.insert(
       std::upper_bound(
           audio_output_redirectors_.begin(), audio_output_redirectors_.end(),
           redirector,
           [](AudioOutputRedirectorInput* a, AudioOutputRedirectorInput* b) {
             return (a->Order() < b->Order());
           }),
       redirector);
 }

 void MixerInput::RemoveAudioOutputRedirector(
     AudioOutputRedirectorInput* redirector) {
   LOG(INFO) << "Remove redirector from " << device_id_ << "(" << source_ << ")";
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   DCHECK(redirector);
   audio_output_redirectors_.erase(
       std::remove(audio_output_redirectors_.begin(),
                   audio_output_redirectors_.end(), redirector),
       audio_output_redirectors_.end());
 }

 int MixerInput::FillAudioData(int num_frames,
                               RenderingDelay rendering_delay,
                               ::media::AudioBus* dest) {
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   DCHECK(dest);
   DCHECK_EQ(num_channels_, dest->channels());
   DCHECK_GE(dest->frames(), num_frames);

   volume_applied_ = false;

   RenderingDelay redirected_delay = rendering_delay;
   if (!audio_output_redirectors_.empty()) {
     redirected_delay.delay_microseconds +=
         audio_output_redirectors_[0]->GetDelayMicroseconds();
   }
   int filled = FillBuffer(num_frames, redirected_delay, dest);

   bool redirected = false;
   for (auto* redirector : audio_output_redirectors_) {
     redirector->Redirect(dest, filled, rendering_delay, redirected);
     redirected = true;
   }

   if (first_buffer_ && redirected) {
     // If the first buffer is redirected, don't provide any data to the mixer
     // (we want to avoid a 'blip' of sound from the first buffer if it is being
     // redirected).
     filled = 0;
   } else if (previous_ended_in_silence_) {
     if (redirected) {
       // Previous buffer ended in silence, and the current buffer was redirected
       // by the output chain, so maintain silence.
       filled = 0;
     } else {
       // Smoothly fade in from previous silence.
       AudioFader::FadeInHelper(dest, filled, 0, filled, filled);
     }
   } else if (redirected) {
     // Smoothly fade out to silence, since output is now being redirected.
     AudioFader::FadeOutHelper(dest, filled, 0, filled, filled);
   }
   previous_ended_in_silence_ = redirected;
   first_buffer_ = false;

   return filled;
 }

 int MixerInput::FillBuffer(int num_frames,
                            RenderingDelay rendering_delay,
                            ::media::AudioBus* dest) {
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   DCHECK(dest);
   DCHECK_EQ(num_channels_, dest->channels());
   DCHECK_GE(dest->frames(), num_frames);

   if (resampler_) {
     mixer_rendering_delay_ = rendering_delay;
     // resampler_->BufferedFrames() gives incorrect values in the read callback,
     // so track the number of buffered frames ourselves.
     resampler_buffered_frames_ = resampler_->BufferedFrames();
     resampler_->Resample(num_frames, dest);
     return num_frames;
   } else {
     return source_->FillAudioPlaybackFrames(num_frames, rendering_delay, dest);
   }
 }

 void MixerInput::ResamplerReadCallback(int frame_delay,
                                        ::media::AudioBus* output) {
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);

   RenderingDelay delay = mixer_rendering_delay_;
   int64_t resampler_delay =
       std::round(resampler_buffered_frames_ * kMicrosecondsPerSecond /
                  input_samples_per_second_);
   delay.delay_microseconds += resampler_delay;

   const int needed_frames = output->frames();
   int filled = source_->FillAudioPlaybackFrames(needed_frames, delay, output);
   if (filled < needed_frames) {
     output->ZeroFramesPartial(filled, needed_frames - filled);
   }
   resampler_buffered_frames_ += output->frames();
 }

 void MixerInput::SignalError(Source::MixerError error) {
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   if (filter_group_) {
     filter_group_->RemoveInput(this);
     filter_group_ = nullptr;
   }
   source_->OnAudioPlaybackError(error);
 }

 void MixerInput::VolumeScaleAccumulate(const float* src,
                                        int frames,
                                        float* dest) {
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   slew_volume_.ProcessFMAC(volume_applied_ /* repeat_transition */, src, frames,
                            1, dest);
   volume_applied_ = true;
 }

 void MixerInput::SetVolumeMultiplier(float multiplier) {
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   stream_volume_multiplier_ = std::max(0.0f, multiplier);
   float target_volume = TargetVolume();
   LOG(INFO) << device_id_ << "(" << source_
             << "): stream volume = " << stream_volume_multiplier_
             << ", effective multiplier = " << target_volume;
   slew_volume_.SetMaxSlewTimeMs(kDefaultSlewTimeMs);
   slew_volume_.SetVolume(target_volume);
 }

 void MixerInput::SetContentTypeVolume(float volume, int fade_ms) {
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   DCHECK(content_type_ != AudioContentType::kOther);

   type_volume_multiplier_ = std::max(0.0f, std::min(volume, 1.0f));
   float target_volume = TargetVolume();
   LOG(INFO) << device_id_ << "(" << source_
             << "): type volume = " << type_volume_multiplier_
             << ", effective multiplier = " << target_volume;
   if (fade_ms < 0) {
     fade_ms = kDefaultSlewTimeMs;
   } else {
     LOG(INFO) << "Fade over " << fade_ms << " ms";
   }
   slew_volume_.SetMaxSlewTimeMs(fade_ms);
   slew_volume_.SetVolume(target_volume);
 }

 void MixerInput::SetMuted(bool muted) {
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   DCHECK(content_type_ != AudioContentType::kOther);

   mute_volume_multiplier_ = muted ? 0.0f : 1.0f;
   float target_volume = TargetVolume();
   LOG(INFO) << device_id_ << "(" << source_
             << "): mute volume = " << mute_volume_multiplier_
             << ", effective multiplier = " << target_volume;
   slew_volume_.SetMaxSlewTimeMs(kDefaultSlewTimeMs);
   slew_volume_.SetVolume(target_volume);
 }

 float MixerInput::TargetVolume() {
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   float volume = stream_volume_multiplier_ * type_volume_multiplier_ *
                  mute_volume_multiplier_;
   return std::max(0.0f, std::min(volume, 1.0f));
 }

 float MixerInput::InstantaneousVolume() {
   DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
   return slew_volume_.LastBufferMaxMultiplier();
 }

 }  // namespace media
 }  // namespace chromecast
	// Copyright 2018 The Chromium Authors. All rights reserved.
	// Use of this source code is governed by a BSD-style license that can be
	// found in the LICENSE file.

	#include "chromecast/media/cma/backend/mixer_input.h"

	#include <stdint.h>

	#include <algorithm>
	#include <cmath>
	#include <utility>

	#include "base/bind.h"
	#include "base/bind_helpers.h"
	#include "base/logging.h"
	#include "chromecast/media/cma/backend/audio_fader.h"
	#include "chromecast/media/cma/backend/audio_output_redirector_input.h"
	#include "chromecast/media/cma/backend/filter_group.h"
	#include "media/base/audio_bus.h"
	#include "media/base/audio_timestamp_helper.h"
	#include "media/base/multi_channel_resampler.h"

	namespace chromecast {
	namespace media {

	namespace {

	const int64_t kMicrosecondsPerSecond = 1000 * 1000;
	const int kDefaultSlewTimeMs = 15;

	int RoundUpMultiple(int value, int multiple) {
	return multiple * ((value + (multiple - 1)) / multiple);
	}

	} // namespace

	MixerInput::MixerInput(Source* source, FilterGroup* filter_group)
	: source_(source),
	num_channels_(source->num_channels()),
	input_samples_per_second_(source->input_samples_per_second()),
	output_samples_per_second_(filter_group->input_samples_per_second()),
	primary_(source->primary()),
	device_id_(source->device_id()),
	content_type_(source->content_type()),
	stream_volume_multiplier_(1.0f),
	type_volume_multiplier_(1.0f),
	mute_volume_multiplier_(1.0f),
	slew_volume_(kDefaultSlewTimeMs),
	volume_applied_(false),
	previous_ended_in_silence_(false),
	first_buffer_(true),
	resampler_buffered_frames_(0.0) {
	DCHECK(source_);
	DCHECK_GT(num_channels_, 0);
	DCHECK_GT(input_samples_per_second_, 0);

	MediaPipelineBackend::AudioDecoder::RenderingDelay initial_rendering_delay =
	filter_group->GetRenderingDelayToOutput();

	int source_read_size = filter_group->input_frames_per_write();
	if (output_samples_per_second_ > 0 &&
	output_samples_per_second_ != input_samples_per_second_) {
	// Round up to nearest multiple of SincResampler::kKernelSize. The read size
	// must be > kKernelSize, so we round up to at least 2 * kKernelSize.
	source_read_size = std::max(source_->desired_read_size(),
	::media::SincResampler::kKernelSize + 1);
	source_read_size =
	RoundUpMultiple(source_read_size, ::media::SincResampler::kKernelSize);
	double resample_ratio = static_cast<double>(input_samples_per_second_) /
	output_samples_per_second_;
	resampler_ = std::make_unique<::media::MultiChannelResampler>(
	num_channels_, resample_ratio, source_read_size,
	base::BindRepeating(&MixerInput::ResamplerReadCallback,
	base::Unretained(this)));
	resampler_->PrimeWithSilence();

	double resampler_queued_frames = resampler_->BufferedFrames();
	initial_rendering_delay.delay_microseconds +=
	static_cast<int64_t>(resampler_queued_frames * kMicrosecondsPerSecond /
	input_samples_per_second_);
	}

	if (output_samples_per_second_ != 0) {
	// If output_samples_per_second_ is 0, this stream will be unusable.
	// OnError() will be called shortly.
	slew_volume_.SetSampleRate(output_samples_per_second_);
	}
	source_->InitializeAudioPlayback(source_read_size, initial_rendering_delay);

	SetFilterGroup(filter_group);
	}

	MixerInput::~MixerInput() {
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	SetFilterGroup(nullptr);
	source_->FinalizeAudioPlayback();
	}

	void MixerInput::SetFilterGroup(FilterGroup* filter_group) {
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	DCHECK(!filter_group \|\| !filter_group_);

	if (filter_group == filter_group_) {
	return;
	}
	if (filter_group_) {
	filter_group_->RemoveInput(this);
	}
	if (filter_group) {
	filter_group->AddInput(this);
	}
	filter_group_ = filter_group;
	}

	void MixerInput::AddAudioOutputRedirector(
	AudioOutputRedirectorInput* redirector) {
	LOG(INFO) << "Add redirector to " << device_id_ << "(" << source_ << ")";
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	DCHECK(redirector);
	audio_output_redirectors_.insert(
	std::upper_bound(
	audio_output_redirectors_.begin(), audio_output_redirectors_.end(),
	redirector,
	[](AudioOutputRedirectorInput* a, AudioOutputRedirectorInput* b) {
	return (a->Order() < b->Order());
	}),
	redirector);
	}

	void MixerInput::RemoveAudioOutputRedirector(
	AudioOutputRedirectorInput* redirector) {
	LOG(INFO) << "Remove redirector from " << device_id_ << "(" << source_ << ")";
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	DCHECK(redirector);
	audio_output_redirectors_.erase(
	std::remove(audio_output_redirectors_.begin(),
	audio_output_redirectors_.end(), redirector),
	audio_output_redirectors_.end());
	}

	int MixerInput::FillAudioData(int num_frames,
	RenderingDelay rendering_delay,
	::media::AudioBus* dest) {
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	DCHECK(dest);
	DCHECK_EQ(num_channels_, dest->channels());
	DCHECK_GE(dest->frames(), num_frames);

	volume_applied_ = false;

	RenderingDelay redirected_delay = rendering_delay;
	if (!audio_output_redirectors_.empty()) {
	redirected_delay.delay_microseconds +=
	audio_output_redirectors_[0]->GetDelayMicroseconds();
	}
	int filled = FillBuffer(num_frames, redirected_delay, dest);

	bool redirected = false;
	for (auto* redirector : audio_output_redirectors_) {
	redirector->Redirect(dest, filled, rendering_delay, redirected);
	redirected = true;
	}

	if (first_buffer_ && redirected) {
	// If the first buffer is redirected, don't provide any data to the mixer
	// (we want to avoid a 'blip' of sound from the first buffer if it is being
	// redirected).
	filled = 0;
	} else if (previous_ended_in_silence_) {
	if (redirected) {
	// Previous buffer ended in silence, and the current buffer was redirected
	// by the output chain, so maintain silence.
	filled = 0;
	} else {
	// Smoothly fade in from previous silence.
	AudioFader::FadeInHelper(dest, filled, 0, filled, filled);
	}
	} else if (redirected) {
	// Smoothly fade out to silence, since output is now being redirected.
	AudioFader::FadeOutHelper(dest, filled, 0, filled, filled);
	}
	previous_ended_in_silence_ = redirected;
	first_buffer_ = false;

	return filled;
	}

	int MixerInput::FillBuffer(int num_frames,
	RenderingDelay rendering_delay,
	::media::AudioBus* dest) {
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	DCHECK(dest);
	DCHECK_EQ(num_channels_, dest->channels());
	DCHECK_GE(dest->frames(), num_frames);

	if (resampler_) {
	mixer_rendering_delay_ = rendering_delay;
	// resampler_->BufferedFrames() gives incorrect values in the read callback,
	// so track the number of buffered frames ourselves.
	resampler_buffered_frames_ = resampler_->BufferedFrames();
	resampler_->Resample(num_frames, dest);
	return num_frames;
	} else {
	return source_->FillAudioPlaybackFrames(num_frames, rendering_delay, dest);
	}
	}

	void MixerInput::ResamplerReadCallback(int frame_delay,
	::media::AudioBus* output) {
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);

	RenderingDelay delay = mixer_rendering_delay_;
	int64_t resampler_delay =
	std::round(resampler_buffered_frames_ * kMicrosecondsPerSecond /
	input_samples_per_second_);
	delay.delay_microseconds += resampler_delay;

	const int needed_frames = output->frames();
	int filled = source_->FillAudioPlaybackFrames(needed_frames, delay, output);
	if (filled < needed_frames) {
	output->ZeroFramesPartial(filled, needed_frames - filled);
	}
	resampler_buffered_frames_ += output->frames();
	}

	void MixerInput::SignalError(Source::MixerError error) {
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	if (filter_group_) {
	filter_group_->RemoveInput(this);
	filter_group_ = nullptr;
	}
	source_->OnAudioPlaybackError(error);
	}

	void MixerInput::VolumeScaleAccumulate(const float* src,
	int frames,
	float* dest) {
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	slew_volume_.ProcessFMAC(volume_applied_ /* repeat_transition */, src, frames,
	1, dest);
	volume_applied_ = true;
	}

	void MixerInput::SetVolumeMultiplier(float multiplier) {
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	stream_volume_multiplier_ = std::max(0.0f, multiplier);
	float target_volume = TargetVolume();
	LOG(INFO) << device_id_ << "(" << source_
	<< "): stream volume = " << stream_volume_multiplier_
	<< ", effective multiplier = " << target_volume;
	slew_volume_.SetMaxSlewTimeMs(kDefaultSlewTimeMs);
	slew_volume_.SetVolume(target_volume);
	}

	void MixerInput::SetContentTypeVolume(float volume, int fade_ms) {
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	DCHECK(content_type_ != AudioContentType::kOther);

	type_volume_multiplier_ = std::max(0.0f, std::min(volume, 1.0f));
	float target_volume = TargetVolume();
	LOG(INFO) << device_id_ << "(" << source_
	<< "): type volume = " << type_volume_multiplier_
	<< ", effective multiplier = " << target_volume;
	if (fade_ms < 0) {
	fade_ms = kDefaultSlewTimeMs;
	} else {
	LOG(INFO) << "Fade over " << fade_ms << " ms";
	}
	slew_volume_.SetMaxSlewTimeMs(fade_ms);
	slew_volume_.SetVolume(target_volume);
	}

	void MixerInput::SetMuted(bool muted) {
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	DCHECK(content_type_ != AudioContentType::kOther);

	mute_volume_multiplier_ = muted ? 0.0f : 1.0f;
	float target_volume = TargetVolume();
	LOG(INFO) << device_id_ << "(" << source_
	<< "): mute volume = " << mute_volume_multiplier_
	<< ", effective multiplier = " << target_volume;
	slew_volume_.SetMaxSlewTimeMs(kDefaultSlewTimeMs);
	slew_volume_.SetVolume(target_volume);
	}

	float MixerInput::TargetVolume() {
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	float volume = stream_volume_multiplier_ * type_volume_multiplier_ *
	mute_volume_multiplier_;
	return std::max(0.0f, std::min(volume, 1.0f));
	}

	float MixerInput::InstantaneousVolume() {
	DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
	return slew_volume_.LastBufferMaxMultiplier();
	}

	} // namespace media
	} // namespace chromecast