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Enables usage of ChannelMixer in WebRTC's output mixer.

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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}
This commit is contained in:
henrika 2019-07-09 15:37:45 +02:00 committed by Commit Bot
parent b249c54209
commit d0679bd7e2
3 changed files with 30 additions and 8 deletions
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2
audio/channel_send.cc
View File

@ -1118,7 +1118,7 @@ void ChannelSend::ProcessAndEncodeAudio(
void ChannelSend::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
RTC_DCHECK_GT(audio_input->samples_per_channel_, 0);
RTC_DCHECK_LE(audio_input->num_channels_, 2);
RTC_DCHECK_LE(audio_input->num_channels_, 8);
// Measure time between when the audio frame is added to the task queue and
// when the task is actually executed. Goal is to keep track of unwanted

1
modules/audio_mixer/DEPS
View File

@ -1,5 +1,6 @@
include_rules = [
"+audio/utility/audio_frame_operations.h",
"+audio/utility/channel_mixer.h",
"+call",
"+common_audio",
"+modules/audio_coding",

35
modules/audio_mixer/audio_frame_manipulator.cc
View File

@ -11,6 +11,7 @@
#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 {
@ -56,16 +57,36 @@ void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
RTC_DCHECK_GE(target_number_of_channels, 1);
if (frame->num_channels_ == target_number_of_channels) {
// TODO(bugs.webrtc.org/10783): take channel layout into account as well.
if (frame->num_channels() == target_number_of_channels) {
return;
}
if (frame->num_channels_ > target_number_of_channels) {
AudioFrameOperations::DownmixChannels(target_number_of_channels, frame);
} else if (frame->num_channels_ < target_number_of_channels) {
AudioFrameOperations::UpmixChannels(target_number_of_channels, frame);
// 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));
ChannelMixer mixer(GuessChannelLayout(frame->num_channels()),
output_layout);
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 "
RTC_DCHECK_EQ(frame->num_channels(), target_number_of_channels)
<< "Wrong number of channels, " << frame->num_channels() << " vs "
<< target_number_of_channels;
}
} // namespace webrtc