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// Copyright (c) 2012 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 "media/base/audio_converter.h"
#include <algorithm>
#include "base/bind.h"
#include "base/bind_helpers.h"
#include "media/base/audio_pull_fifo.h"
#include "media/base/channel_mixer.h"
#include "media/base/multi_channel_resampler.h"
#include "media/base/vector_math.h"
namespace media {
AudioConverter::AudioConverter(const AudioParameters& input_params,
const AudioParameters& output_params,
bool disable_fifo)
: downmix_early_(false),
resampler_frame_delay_(0),
input_channel_count_(input_params.channels()) {
CHECK(input_params.IsValid());
CHECK(output_params.IsValid());
// Handle different input and output channel layouts.
if (input_params.channel_layout() != output_params.channel_layout()) {
DVLOG(1) << "Remixing channel layout from " << input_params.channel_layout()
<< " to " << output_params.channel_layout() << "; from "
<< input_params.channels() << " channels to "
<< output_params.channels() << " channels.";
channel_mixer_.reset(new ChannelMixer(
input_params.channel_layout(), output_params.channel_layout()));
// Pare off data as early as we can for efficiency.
downmix_early_ = input_params.channels() > output_params.channels();
if (downmix_early_) {
DVLOG(1) << "Remixing channel layout prior to resampling.";
// |unmixed_audio_| will be allocated on the fly.
} else {
// Instead, if we're not downmixing early we need a temporary AudioBus
// which matches the input channel count but uses the output frame size
// since we'll mix into the AudioBus from the output stream.
unmixed_audio_ = AudioBus::Create(
input_params.channels(), output_params.frames_per_buffer());
}
}
// Only resample if necessary since it's expensive.
if (input_params.sample_rate() != output_params.sample_rate()) {
DVLOG(1) << "Resampling from " << input_params.sample_rate() << " to "
<< output_params.sample_rate();
double io_sample_rate_ratio = input_params.sample_rate() /
static_cast<double>(output_params.sample_rate());
resampler_.reset(new MultiChannelResampler(
downmix_early_ ? output_params.channels() :
input_params.channels(),
io_sample_rate_ratio, base::Bind(
&AudioConverter::ProvideInput, base::Unretained(this))));
}
input_frame_duration_ = base::TimeDelta::FromMicroseconds(
base::Time::kMicrosecondsPerSecond /
static_cast<double>(input_params.sample_rate()));
output_frame_duration_ = base::TimeDelta::FromMicroseconds(
base::Time::kMicrosecondsPerSecond /
static_cast<double>(output_params.sample_rate()));
if (disable_fifo)
return;
// Since the resampler / output device may want a different buffer size than
// the caller asked for, we need to use a FIFO to ensure that both sides
// read in chunk sizes they're configured for.
if (resampler_.get() ||
input_params.frames_per_buffer() != output_params.frames_per_buffer()) {
DVLOG(1) << "Rebuffering from " << input_params.frames_per_buffer()
<< " to " << output_params.frames_per_buffer();
audio_fifo_.reset(new AudioPullFifo(
downmix_early_ ? output_params.channels() :
input_params.channels(),
input_params.frames_per_buffer(), base::Bind(
&AudioConverter::SourceCallback,
base::Unretained(this))));
}
}
AudioConverter::~AudioConverter() {}
void AudioConverter::AddInput(InputCallback* input) {
transform_inputs_.push_back(input);
}
void AudioConverter::RemoveInput(InputCallback* input) {
DCHECK(std::find(transform_inputs_.begin(), transform_inputs_.end(), input) !=
transform_inputs_.end());
transform_inputs_.remove(input);
if (transform_inputs_.empty())
Reset();
}
void AudioConverter::Reset() {
if (audio_fifo_)
audio_fifo_->Clear();
if (resampler_)
resampler_->Flush();
}
void AudioConverter::Convert(AudioBus* dest) {
if (transform_inputs_.empty()) {
dest->Zero();
return;
}
bool needs_mixing = channel_mixer_ && !downmix_early_;
AudioBus* temp_dest = needs_mixing ? unmixed_audio_.get() : dest;
DCHECK(temp_dest);
if (!resampler_ && !audio_fifo_) {
SourceCallback(0, temp_dest);
} else {
if (resampler_)
resampler_->Resample(temp_dest, temp_dest->frames());
else
ProvideInput(0, temp_dest);
}
if (needs_mixing) {
DCHECK_EQ(temp_dest->frames(), dest->frames());
channel_mixer_->Transform(temp_dest, dest);
}
}
void AudioConverter::SourceCallback(int fifo_frame_delay, AudioBus* dest) {
bool needs_downmix = channel_mixer_ && downmix_early_;
if (!mixer_input_audio_bus_ ||
mixer_input_audio_bus_->frames() != dest->frames()) {
mixer_input_audio_bus_ =
AudioBus::Create(input_channel_count_, dest->frames());
}
if (needs_downmix &&
(!unmixed_audio_ || unmixed_audio_->frames() != dest->frames())) {
// If we're downmixing early we need a temporary AudioBus which matches
// the the input channel count and input frame size since we're passing
// |unmixed_audio_| directly to the |source_callback_|.
unmixed_audio_ = AudioBus::Create(input_channel_count_, dest->frames());
}
AudioBus* temp_dest = needs_downmix ? unmixed_audio_.get() : dest;
// Sanity check our inputs.
DCHECK_EQ(temp_dest->frames(), mixer_input_audio_bus_->frames());
DCHECK_EQ(temp_dest->channels(), mixer_input_audio_bus_->channels());
// Calculate the buffer delay for this callback.
base::TimeDelta buffer_delay;
if (resampler_) {
buffer_delay += base::TimeDelta::FromMicroseconds(
resampler_frame_delay_ * output_frame_duration_.InMicroseconds());
}
if (audio_fifo_) {
buffer_delay += base::TimeDelta::FromMicroseconds(
fifo_frame_delay * input_frame_duration_.InMicroseconds());
}
// Have each mixer render its data into an output buffer then mix the result.
for (InputCallbackSet::iterator it = transform_inputs_.begin();
it != transform_inputs_.end(); ++it) {
InputCallback* input = *it;
float volume = input->ProvideInput(
mixer_input_audio_bus_.get(), buffer_delay);
// Optimize the most common single input, full volume case.
if (it == transform_inputs_.begin()) {
if (volume == 1.0f) {
mixer_input_audio_bus_->CopyTo(temp_dest);
continue;
}
// Zero |temp_dest| otherwise, so we're mixing into a clean buffer.
temp_dest->Zero();
}
// Volume adjust and mix each mixer input into |temp_dest| after rendering.
if (volume > 0) {
for (int i = 0; i < mixer_input_audio_bus_->channels(); ++i) {
vector_math::FMAC(
mixer_input_audio_bus_->channel(i), volume,
mixer_input_audio_bus_->frames(), temp_dest->channel(i));
}
}
}
if (needs_downmix) {
DCHECK_EQ(temp_dest->frames(), dest->frames());
channel_mixer_->Transform(temp_dest, dest);
}
}
void AudioConverter::ProvideInput(int resampler_frame_delay, AudioBus* dest) {
resampler_frame_delay_ = resampler_frame_delay;
if (audio_fifo_)
audio_fifo_->Consume(dest, dest->frames());
else
SourceCallback(0, dest);
}
} // namespace media
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