Remove AudioProcessing::ChannelLayout

This enum is no longer needed. Also moving the last piece of code from
common.h to audio_processing_impl.h, allowing to delete common.h.

Bug: chromium:1271981, b/217349489
Change-Id: If115336c36d6d7b5845a903e421c18aebfe434ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251242
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35946}
This commit is contained in:
Henrik Lundin 2022-02-08 09:15:12 +00:00 committed by WebRTC LUCI CQ
parent fd6c744a7c
commit 8a9aa55561
9 changed files with 4 additions and 66 deletions

View file

@ -146,7 +146,6 @@ rtc_library("audio_processing") {
"audio_processing_builder_impl.cc",
"audio_processing_impl.cc",
"audio_processing_impl.h",
"common.h",
"echo_control_mobile_impl.cc",
"echo_control_mobile_impl.h",
"gain_control_impl.cc",

View file

@ -24,7 +24,6 @@
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/common.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "modules/audio_processing/optionally_built_submodule_creators.h"

View file

@ -51,6 +51,10 @@ namespace webrtc {
class ApmDataDumper;
class AudioConverter;
constexpr int RuntimeSettingQueueSize() {
return 100;
}
class AudioProcessingImpl : public AudioProcessing {
public:
// Methods forcing APM to run in a single-threaded manner.

View file

@ -14,7 +14,6 @@
#include <memory>
#include "api/scoped_refptr.h"
#include "modules/audio_processing/common.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/optionally_built_submodule_creators.h"
#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"

View file

@ -27,7 +27,6 @@
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/audio_processing_impl.h"
#include "modules/audio_processing/common.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
#include "modules/audio_processing/test/protobuf_utils.h"

View file

@ -1,38 +0,0 @@
/*
* Copyright (c) 2014 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.
*/
#ifndef MODULES_AUDIO_PROCESSING_COMMON_H_
#define MODULES_AUDIO_PROCESSING_COMMON_H_
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/checks.h"
namespace webrtc {
constexpr int RuntimeSettingQueueSize() {
return 100;
}
static inline size_t ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
switch (layout) {
case AudioProcessing::kMono:
case AudioProcessing::kMonoAndKeyboard:
return 1;
case AudioProcessing::kStereo:
case AudioProcessing::kStereoAndKeyboard:
return 2;
}
RTC_DCHECK_NOTREACHED();
return 0;
}
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_COMMON_H_

View file

@ -385,17 +385,6 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
std::string ToString() const;
};
// TODO(mgraczyk): Remove once all methods that use ChannelLayout are gone.
enum ChannelLayout {
kMono,
// Left, right.
kStereo,
// Mono, keyboard, and mic.
kMonoAndKeyboard,
// Left, right, keyboard, and mic.
kStereoAndKeyboard
};
// Specifies the properties of a setting to be passed to AudioProcessing at
// runtime.
class RuntimeSetting {

View file

@ -139,15 +139,4 @@ void SetFrameSampleRate(Int16FrameData* frame, int sample_rate_hz) {
AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000;
}
AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) {
switch (num_channels) {
case 1:
return AudioProcessing::kMono;
case 2:
return AudioProcessing::kStereo;
default:
RTC_CHECK_NOTREACHED();
}
}
} // namespace webrtc

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@ -154,8 +154,6 @@ void SetContainerFormat(int sample_rate_hz,
cb->reset(new ChannelBuffer<T>(frame->samples_per_channel, num_channels));
}
AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels);
template <typename T>
float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) {
float mse = 0;