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/*
* Copyright (c) 2012 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/audio_frame_manipulator.h"
#include "audio/utility/audio_frame_operations.h"
#include "audio/utility/channel_mixer.h"
#include "rtc_base/checks.h"
namespace webrtc {
uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
if (audio_frame.muted()) {
return 0;
}
uint32_t energy = 0;
const int16_t* frame_data = audio_frame.data();
for (size_t position = 0;
position < audio_frame.samples_per_channel_ * audio_frame.num_channels_;
position++) {
// TODO(aleloi): This can overflow. Convert to floats.
energy += frame_data[position] * frame_data[position];
}
return energy;
}
void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
RTC_DCHECK(audio_frame);
RTC_DCHECK_GE(start_gain, 0.0f);
RTC_DCHECK_GE(target_gain, 0.0f);
if (start_gain == target_gain || audio_frame->muted()) {
return;
}
size_t samples = audio_frame->samples_per_channel_;
RTC_DCHECK_LT(0, samples);
float increment = (target_gain - start_gain) / samples;
float gain = start_gain;
int16_t* frame_data = audio_frame->mutable_data();
for (size_t i = 0; i < samples; ++i) {
// If the audio is interleaved of several channels, we want to
// apply the same gain change to the ith sample of every channel.
for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) {
frame_data[audio_frame->num_channels_ * i + ch] *= gain;
}
gain += increment;
}
}
void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
RTC_DCHECK_GE(target_number_of_channels, 1);
// TODO(bugs.webrtc.org/10783): take channel layout into account as well.
if (frame->num_channels() == target_number_of_channels) {
return;
}
// Use legacy components for the most simple cases (mono <-> stereo) to ensure
// that native WebRTC clients are not affected when support for multi-channel
// audio is added to Chrome.
// TODO(bugs.webrtc.org/10783): utilize channel mixer for mono/stereo as well.
if (target_number_of_channels < 3 && frame->num_channels() < 3) {
if (frame->num_channels() > target_number_of_channels) {
AudioFrameOperations::DownmixChannels(target_number_of_channels, frame);
} else {
AudioFrameOperations::UpmixChannels(target_number_of_channels, frame);
}
} else {
// Use generic channel mixer when the number of channels for input our
// output is larger than two. E.g. stereo -> 5.1 channel up-mixing.
// TODO(bugs.webrtc.org/10783): ensure that actual channel layouts are used
// instead of guessing based on number of channels.
const ChannelLayout output_layout(
GuessChannelLayout(target_number_of_channels));
const ChannelLayout input_layout(GuessChannelLayout(frame->num_channels()));
ChannelMixer mixer(input_layout, frame->num_channels(), output_layout,
target_number_of_channels);
mixer.Transform(frame);
RTC_DCHECK_EQ(frame->channel_layout(), output_layout);
}
RTC_DCHECK_EQ(frame->num_channels(), target_number_of_channels)
<< "Wrong number of channels, " << frame->num_channels() << " vs "
<< target_number_of_channels;
}
} // namespace webrtc
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