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Changed folder structure in new mixer and fixed simple lint errors.

Browse Source 
The folder structure is now as was agreed on in the 'Slim and Modular
WebRTC' effort.  Also added some dependencies that were previously in
another part of the tree.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2238803002
Cr-Commit-Position: refs/heads/master@{#13742}
This commit is contained in:
aleloi
2016-08-15 03:01:31 -07:00
committed by Commit bot
parent 714dd4e532
commit 5bcc00e538
11 changed files with 130 additions and 41 deletions
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14
webrtc/modules/audio_mixer/BUILD.gn
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@ -45,15 +45,17 @@ source_set("audio_mixer") {
source_set("audio_conference_mixer") {
sources = [
"include/audio_mixer_defines.h",
"include/new_audio_conference_mixer.h",
"source/new_audio_conference_mixer_impl.cc",
"source/new_audio_conference_mixer_impl.h",
"audio_frame_manipulator.cc",
"audio_frame_manipulator.h",
"audio_mixer_defines.h",
"new_audio_conference_mixer.h",
"new_audio_conference_mixer_impl.cc",
"new_audio_conference_mixer_impl.h",
]
public = [
"include/audio_mixer_defines.h",
"include/new_audio_conference_mixer.h",
"audio_mixer_defines.h",
"new_audio_conference_mixer.h",
]
configs += [ "../..:common_config" ]

62
webrtc/modules/audio_mixer/audio_frame_manipulator.cc Normal file
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@ -0,0 +1,62 @@
/*
* 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 "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
namespace {
// Linear ramping over 80 samples.
// TODO(hellner): ramp using fix point?
const float kRampArray[] = {
0.0000f, 0.0127f, 0.0253f, 0.0380f, 0.0506f, 0.0633f, 0.0759f, 0.0886f,
0.1013f, 0.1139f, 0.1266f, 0.1392f, 0.1519f, 0.1646f, 0.1772f, 0.1899f,
0.2025f, 0.2152f, 0.2278f, 0.2405f, 0.2532f, 0.2658f, 0.2785f, 0.2911f,
0.3038f, 0.3165f, 0.3291f, 0.3418f, 0.3544f, 0.3671f, 0.3797f, 0.3924f,
0.4051f, 0.4177f, 0.4304f, 0.4430f, 0.4557f, 0.4684f, 0.4810f, 0.4937f,
0.5063f, 0.5190f, 0.5316f, 0.5443f, 0.5570f, 0.5696f, 0.5823f, 0.5949f,
0.6076f, 0.6203f, 0.6329f, 0.6456f, 0.6582f, 0.6709f, 0.6835f, 0.6962f,
0.7089f, 0.7215f, 0.7342f, 0.7468f, 0.7595f, 0.7722f, 0.7848f, 0.7975f,
0.8101f, 0.8228f, 0.8354f, 0.8481f, 0.8608f, 0.8734f, 0.8861f, 0.8987f,
0.9114f, 0.9241f, 0.9367f, 0.9494f, 0.9620f, 0.9747f, 0.9873f, 1.0000f};
const size_t kRampSize = sizeof(kRampArray) / sizeof(kRampArray[0]);
} // namespace
uint32_t NewMixerCalculateEnergy(const AudioFrame& audio_frame) {
uint32_t energy = 0;
for (size_t position = 0; position < audio_frame.samples_per_channel_;
position++) {
// TODO(andrew): this can easily overflow.
energy += audio_frame.data_[position] * audio_frame.data_[position];
}
return energy;
}
void NewMixerRampIn(AudioFrame* audio_frame) {
assert(kRampSize <= audio_frame->samples_per_channel_);
for (size_t i = 0; i < kRampSize; i++) {
audio_frame->data_[i] =
static_cast<int16_t>(kRampArray[i] * audio_frame->data_[i]);
}
}
void NewMixerRampOut(AudioFrame* audio_frame) {
assert(kRampSize <= audio_frame->samples_per_channel_);
for (size_t i = 0; i < kRampSize; i++) {
const size_t kRampPos = kRampSize - 1 - i;
audio_frame->data_[i] =
static_cast<int16_t>(kRampArray[kRampPos] * audio_frame->data_[i]);
}
memset(&audio_frame->data_[kRampSize], 0,
(audio_frame->samples_per_channel_ - kRampSize) *
sizeof(audio_frame->data_[0]));
}
} // namespace webrtc

28
webrtc/modules/audio_mixer/audio_frame_manipulator.h Normal file
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@ -0,0 +1,28 @@
/*
* Copyright (c) 2011 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 WEBRTC_MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
#define WEBRTC_MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioFrame;
// Updates the audioFrame's energy (based on its samples).
uint32_t NewMixerCalculateEnergy(const AudioFrame& audio_frame);
// Apply linear step function that ramps in/out the audio samples in audio_frame
void NewMixerRampIn(AudioFrame* audio_frame);
void NewMixerRampOut(AudioFrame* audio_frame);
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_

10
webrtc/modules/audio_mixer/audio_mixer.gypi
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@ -18,10 +18,12 @@
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
],
'sources': [
'include/new_audio_conference_mixer.h',
'include/audio_mixer_defines.h',
'source/new_audio_conference_mixer_impl.cc',
'source/new_audio_conference_mixer_impl.h',
'audio_frame_manipulator.cc',
'audio_frame_manipulator.h',
'new_audio_conference_mixer.h',
'audio_mixer_defines.h',
'new_audio_conference_mixer_impl.cc',
'new_audio_conference_mixer_impl.h',
],
},
{

4
webrtc/modules/audio_mixer/audio_mixer.h
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@ -14,8 +14,8 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h"
#include "webrtc/modules/audio_mixer/include/audio_mixer_defines.h"
#include "webrtc/modules/audio_mixer/new_audio_conference_mixer.h"
#include "webrtc/modules/audio_mixer/audio_mixer_defines.h"
#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/level_indicator.h"
#include "webrtc/voice_engine/voice_engine_defines.h"

6
webrtc/modules/audio_mixer/include/audio_mixer_defines.h → webrtc/modules/audio_mixer/audio_mixer_defines.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_AUDIO_MIXER_DEFINES_H_
#define WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_AUDIO_MIXER_DEFINES_H_
#ifndef WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_DEFINES_H_
#define WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_DEFINES_H_
#include "webrtc/base/checks.h"
#include "webrtc/modules/include/module_common_types.h"
@ -54,4 +54,4 @@ class MixerAudioSource {
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_AUDIO_MIXER_DEFINES_H_
#endif // WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_DEFINES_H_

8
webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h → webrtc/modules/audio_mixer/new_audio_conference_mixer.h
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@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_NEW_AUDIO_CONFERENCE_MIXER_H_
#define WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_NEW_AUDIO_CONFERENCE_MIXER_H_
#ifndef WEBRTC_MODULES_AUDIO_MIXER_NEW_AUDIO_CONFERENCE_MIXER_H_
#define WEBRTC_MODULES_AUDIO_MIXER_NEW_AUDIO_CONFERENCE_MIXER_H_
#include "webrtc/modules/audio_mixer/include/audio_mixer_defines.h"
#include "webrtc/modules/audio_mixer/audio_mixer_defines.h"
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/include/module_common_types.h"
@ -64,4 +64,4 @@ class NewAudioConferenceMixer {
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_NEW_AUDIO_CONFERENCE_MIXER_H_
#endif // WEBRTC_MODULES_AUDIO_MIXER_NEW_AUDIO_CONFERENCE_MIXER_H_

12
webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.cc → webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.cc
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@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h"
#include "webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.h"
#include <algorithm>
#include <functional>
#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
#include "webrtc/modules/audio_mixer/include/audio_mixer_defines.h"
#include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
#include "webrtc/modules/audio_mixer/audio_mixer_defines.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
@ -32,7 +32,7 @@ class SourceFrame {
muted_(m),
was_mixed_before_(was_mixed_before) {
if (!muted_) {
energy_ = CalculateEnergy(*a);
energy_ = NewMixerCalculateEnergy(*a);
}
}
@ -410,7 +410,7 @@ AudioFrameList NewAudioConferenceMixerImpl::UpdateToMix(
if (maxAudioFrameCounter > 0) {
--maxAudioFrameCounter;
if (!p.was_mixed_before_) {
RampIn(*p.audio_frame_);
NewMixerRampIn(p.audio_frame_);
}
result.emplace_back(p.audio_frame_, false);
is_mixed = true;
@ -418,7 +418,7 @@ AudioFrameList NewAudioConferenceMixerImpl::UpdateToMix(
// Ramp out unmuted.
if (p.was_mixed_before_ && !is_mixed) {
RampOut(*p.audio_frame_);
NewMixerRampOut(p.audio_frame_);
result.emplace_back(p.audio_frame_, false);
}

8
webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h → webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
#define WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
#ifndef WEBRTC_MODULES_AUDIO_MIXER_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
#define WEBRTC_MODULES_AUDIO_MIXER_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
#include <list>
#include <map>
@ -18,7 +18,7 @@
#include "webrtc/base/thread_checker.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h"
#include "webrtc/modules/audio_mixer/new_audio_conference_mixer.h"
#include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
#include "webrtc/modules/include/module_common_types.h"
@ -154,4 +154,4 @@ class NewAudioConferenceMixerImpl : public NewAudioConferenceMixer {
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
#endif // WEBRTC_MODULES_AUDIO_MIXER_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_

5
webrtc/modules/audio_mixer/source/OWNERS
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@ -1,5 +0,0 @@
# These are for the common case of adding or renaming files. If you're doing
# structural changes, please get a review from a reviewer in this file.
per-file *.gyp=*
per-file *.gypi=*

14
webrtc/modules/audio_mixer/test/audio_mixer_unittest.cc
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@ -17,9 +17,9 @@
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
#include "webrtc/modules/audio_mixer/audio_mixer.h"
#include "webrtc/modules/audio_mixer/include/audio_mixer_defines.h"
#include "webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h"
#include "webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h"
#include "webrtc/modules/audio_mixer/audio_mixer_defines.h"
#include "webrtc/modules/audio_mixer/new_audio_conference_mixer.h"
#include "webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.h"
using testing::_;
using testing::Exactly;
@ -455,10 +455,10 @@ TEST_F(CompareWithOldMixerTest, ThreeParticipantsDifferentFrames) {
TEST_F(CompareWithOldMixerTest, ManyParticipantsDifferentFrames) {
Reset();
constexpr int num_participants = 20;
AudioFrame audio_frames[num_participants];
constexpr int kNumParticipants = 20;
AudioFrame audio_frames[kNumParticipants];
for (int i = 0; i < num_participants; ++i) {
for (int i = 0; i < kNumParticipants; ++i) {
ResetFrame(&audio_frames[i]);
audio_frames[i].id_ = 1;
audio_frames[i].data_[10] = 100 * (i % 5);
@ -468,7 +468,7 @@ TEST_F(CompareWithOldMixerTest, ManyParticipantsDifferentFrames) {
}
}
for (int i = 0; i < num_participants; ++i) {
for (int i = 0; i < kNumParticipants; ++i) {
if (i % 2 == 0) {
AddParticipant(&audio_frames[i],
MixerParticipant::AudioFrameInfo::kMuted);