mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
b2b6166dc4
Bug: chromium:335805780 Change-Id: Ida671d317c07983cc51faa1a498642747dbb810c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349322 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42199}
93 lines
3.5 KiB
C++
93 lines
3.5 KiB
C++
/*
|
|
* 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
|