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d0679bd7e2
Ensures that newly added ChannelMixer is utilized when number of channels is larger than two in the output mixer. Decided to land with henrik.lundin as TBR since he has reviewed all other changes in WebRTC related to channel mixing for multi-channel cases. All this CL does is to ensure that the new channel mixing scheme can be used in Chrome. The old scheme is still used for mono and stereo combinations. TBR: henrik.lundin Bug: webrtc:10783 Change-Id: I11c02f1b4ef60e847095efbcd5e5f5faf27a5cdd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140290 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28517}
92 lines
3.4 KiB
C++
92 lines
3.4 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_mixer/audio_frame_manipulator.h"
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#include "audio/utility/audio_frame_operations.h"
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#include "audio/utility/channel_mixer.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
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if (audio_frame.muted()) {
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return 0;
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}
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uint32_t energy = 0;
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const int16_t* frame_data = audio_frame.data();
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for (size_t position = 0;
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position < audio_frame.samples_per_channel_ * audio_frame.num_channels_;
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position++) {
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// TODO(aleloi): This can overflow. Convert to floats.
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energy += frame_data[position] * frame_data[position];
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}
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return energy;
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}
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void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
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RTC_DCHECK(audio_frame);
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RTC_DCHECK_GE(start_gain, 0.0f);
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RTC_DCHECK_GE(target_gain, 0.0f);
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if (start_gain == target_gain || audio_frame->muted()) {
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return;
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}
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size_t samples = audio_frame->samples_per_channel_;
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RTC_DCHECK_LT(0, samples);
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float increment = (target_gain - start_gain) / samples;
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float gain = start_gain;
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int16_t* frame_data = audio_frame->mutable_data();
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for (size_t i = 0; i < samples; ++i) {
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// If the audio is interleaved of several channels, we want to
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// apply the same gain change to the ith sample of every channel.
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for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) {
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frame_data[audio_frame->num_channels_ * i + ch] *= gain;
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}
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gain += increment;
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}
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}
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void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
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RTC_DCHECK_GE(target_number_of_channels, 1);
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// TODO(bugs.webrtc.org/10783): take channel layout into account as well.
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if (frame->num_channels() == target_number_of_channels) {
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return;
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}
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// Use legacy components for the most simple cases (mono <-> stereo) to ensure
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// that native WebRTC clients are not affected when support for multi-channel
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// audio is added to Chrome.
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// TODO(bugs.webrtc.org/10783): utilize channel mixer for mono/stereo as well.
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if (target_number_of_channels < 3 && frame->num_channels() < 3) {
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if (frame->num_channels() > target_number_of_channels) {
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AudioFrameOperations::DownmixChannels(target_number_of_channels, frame);
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} else {
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AudioFrameOperations::UpmixChannels(target_number_of_channels, frame);
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}
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} else {
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// Use generic channel mixer when the number of channels for input our
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// output is larger than two. E.g. stereo -> 5.1 channel up-mixing.
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// TODO(bugs.webrtc.org/10783): ensure that actual channel layouts are used
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// instead of guessing based on number of channels.
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const ChannelLayout output_layout(
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GuessChannelLayout(target_number_of_channels));
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ChannelMixer mixer(GuessChannelLayout(frame->num_channels()),
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output_layout);
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mixer.Transform(frame);
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RTC_DCHECK_EQ(frame->channel_layout(), output_layout);
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}
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RTC_DCHECK_EQ(frame->num_channels(), target_number_of_channels)
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<< "Wrong number of channels, " << frame->num_channels() << " vs "
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<< target_number_of_channels;
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}
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} // namespace webrtc
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