zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Move functionality out from AudioFrame and into AudioFrameOperations.

Browse Source 
This CL is in preparation to move the AudioFrame into webrtc/api. The
AudioFrame is a POD type used for representing 10ms of audio. It
appears as a parameter and return value of interfaces being migrated
to webrtc/api, in particular AudioMixer.

Here, methods operator+=, operator>>=, Mute are
moved into a new target webrtc/audio/utility/audio_frame_operations,
and dependencies are changed to use
the new versions. The old AudioFrame methods are marked deprecated.

The audio frame utilities in webrtc/modules/utility:audio_frame_operations
are also moved to the new location.

TBR=kjellander@webrtc.org
BUG=webrtc:6548
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2424173003
Cr-Commit-Position: refs/heads/master@{#15413}
This commit is contained in:
aleloi
2016-12-05 01:46:09 -08:00
committed by Commit bot
parent bd9bdf6140
commit 6321b49a0d
29 changed files with 261 additions and 128 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
.gn
View File

@ -25,6 +25,7 @@ check_targets = [
"//webrtc/api:audio_mixer_api",
"//webrtc/api:transport_api",
"//webrtc/api:rtc_stats_api",
"//webrtc/audio/utility/*",
"//webrtc/modules/audio_coding:audio_decoder_factory_interface",
"//webrtc/modules/audio_coding:audio_format",
"//webrtc/modules/audio_coding:audio_format_conversion",

8
webrtc/audio/BUILD.gn
View File

@ -37,11 +37,6 @@ rtc_static_library("audio") {
"../modules/audio_processing",
"../system_wrappers",
"../voice_engine",
"utility", # Bogus dep, needed for landing
# codereview.webrtc.org/2424173003 without breaking
# internal projects. See
# bugs.webrtc.org/6548. TODO(aleloi): remove dependency
# when codereview.webrtc.org/2424173003 has landed.
]
}
if (rtc_include_tests) {
@ -51,6 +46,7 @@ if (rtc_include_tests) {
"audio_receive_stream_unittest.cc",
"audio_send_stream_unittest.cc",
"audio_state_unittest.cc",
"utility/audio_frame_operations_unittest.cc",
]
deps = [
":audio",
@ -59,6 +55,8 @@ if (rtc_include_tests) {
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer:audio_mixer_impl",
"../test:test_common",
"../test:test_support",
"utility:audio_frame_operations",
"//testing/gmock",
"//testing/gtest",
]

16
webrtc/audio/utility/BUILD.gn
View File

@ -8,4 +8,20 @@
import("../../build/webrtc.gni")
group("utility") {
public_deps = [
":audio_frame_operations",
]
}
rtc_static_library("audio_frame_operations") {
sources = [
"audio_frame_operations.cc",
"audio_frame_operations.h",
]
deps = [
"../..:webrtc_common",
"../../base:rtc_base_approved",
"../../modules/audio_coding:audio_format_conversion",
]
}

84
webrtc/modules/utility/source/audio_frame_operations.cc → webrtc/audio/utility/audio_frame_operations.cc
View File

@ -8,18 +8,65 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/audio/utility/audio_frame_operations.h"
#include <algorithm>
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
namespace {
namespace {
// 2.7ms @ 48kHz, 4ms @ 32kHz, 8ms @ 16kHz.
const size_t kMuteFadeFrames = 128;
const float kMuteFadeInc = 1.0f / kMuteFadeFrames;
} // namespace {
} // namespace
void AudioFrameOperations::Add(const AudioFrame& frame_to_add,
AudioFrame* result_frame) {
// Sanity check.
RTC_DCHECK(result_frame);
RTC_DCHECK_GT(result_frame->num_channels_, 0);
RTC_DCHECK_EQ(result_frame->num_channels_, frame_to_add.num_channels_);
bool no_previous_data = false;
if (result_frame->samples_per_channel_ != frame_to_add.samples_per_channel_) {
// Special case we have no data to start with.
RTC_DCHECK_EQ(result_frame->samples_per_channel_, 0);
result_frame->samples_per_channel_ = frame_to_add.samples_per_channel_;
no_previous_data = true;
}
if (result_frame->vad_activity_ == AudioFrame::kVadActive ||
frame_to_add.vad_activity_ == AudioFrame::kVadActive) {
result_frame->vad_activity_ = AudioFrame::kVadActive;
} else if (result_frame->vad_activity_ == AudioFrame::kVadUnknown ||
frame_to_add.vad_activity_ == AudioFrame::kVadUnknown) {
result_frame->vad_activity_ = AudioFrame::kVadUnknown;
}
if (result_frame->speech_type_ != frame_to_add.speech_type_)
result_frame->speech_type_ = AudioFrame::kUndefined;
if (no_previous_data) {
std::copy(frame_to_add.data_, frame_to_add.data_ +
frame_to_add.samples_per_channel_ *
result_frame->num_channels_,
result_frame->data_);
} else {
for (size_t i = 0;
i < result_frame->samples_per_channel_ * result_frame->num_channels_;
i++) {
const int32_t wrap_guard = static_cast<int32_t>(result_frame->data_[i]) +
static_cast<int32_t>(frame_to_add.data_[i]);
result_frame->data_[i] = rtc::saturated_cast<int16_t>(wrap_guard);
}
}
return;
}
void AudioFrameOperations::MonoToStereo(const int16_t* src_audio,
size_t samples_per_channel,
@ -68,7 +115,10 @@ int AudioFrameOperations::StereoToMono(AudioFrame* frame) {
}
void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
if (frame->num_channels_ != 2) return;
RTC_DCHECK(frame);
if (frame->num_channels_ != 2) {
return;
}
for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
int16_t temp_data = frame->data_[i];
@ -77,7 +127,8 @@ void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
}
}
void AudioFrameOperations::Mute(AudioFrame* frame, bool previous_frame_muted,
void AudioFrameOperations::Mute(AudioFrame* frame,
bool previous_frame_muted,
bool current_frame_muted) {
RTC_DCHECK(frame);
if (!previous_frame_muted && !current_frame_muted) {
@ -125,14 +176,30 @@ void AudioFrameOperations::Mute(AudioFrame* frame, bool previous_frame_muted,
}
}
void AudioFrameOperations::Mute(AudioFrame* frame) {
Mute(frame, true, true);
}
void AudioFrameOperations::ApplyHalfGain(AudioFrame* frame) {
RTC_DCHECK(frame);
RTC_DCHECK_GT(frame->num_channels_, 0);
if (frame->num_channels_ < 1) {
return;
}
for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
i++) {
frame->data_[i] = frame->data_[i] >> 1;
}
}
int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) {
if (frame.num_channels_ != 2) {
return -1;
}
for (size_t i = 0; i < frame.samples_per_channel_; i++) {
frame.data_[2 * i] =
static_cast<int16_t>(left * frame.data_[2 * i]);
frame.data_[2 * i] = static_cast<int16_t>(left * frame.data_[2 * i]);
frame.data_[2 * i + 1] =
static_cast<int16_t>(right * frame.data_[2 * i + 1]);
}
@ -156,5 +223,4 @@ int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame& frame) {
}
return 0;
}
} // namespace webrtc

83
webrtc/audio/utility/audio_frame_operations.h Normal file
View File

@ -0,0 +1,83 @@
/*
* 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.
*/
#ifndef WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
#define WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
#include <stddef.h>
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioFrame;
// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
// Change reference parameters to pointers. Consider using a namespace rather
// than a class.
class AudioFrameOperations {
public:
// Add samples in |frame_to_add| with samples in |result_frame|
// putting the results in |results_frame|. The fields
// |vad_activity_| and |speech_type_| of the result frame are
// updated. If |result_frame| is empty (|samples_per_channel_|==0),
// the samples in |frame_to_add| are added to it. The number of
// channels and number of samples per channel must match except when
// |result_frame| is empty.
static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);
// Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place
// operation, meaning src_audio and dst_audio must point to different
// buffers. It is the caller's responsibility to ensure that |dst_audio| is
// sufficiently large.
static void MonoToStereo(const int16_t* src_audio,
size_t samples_per_channel,
int16_t* dst_audio);
// |frame.num_channels_| will be updated. This version checks for sufficient
// buffer size and that |num_channels_| is mono.
static int MonoToStereo(AudioFrame* frame);
// Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place
// operation, meaning |src_audio| and |dst_audio| may point to the same
// buffer.
static void StereoToMono(const int16_t* src_audio,
size_t samples_per_channel,
int16_t* dst_audio);
// |frame.num_channels_| will be updated. This version checks that
// |num_channels_| is stereo.
static int StereoToMono(AudioFrame* frame);
// Swap the left and right channels of |frame|. Fails silently if |frame| is
// not stereo.
static void SwapStereoChannels(AudioFrame* frame);
// Conditionally zero out contents of |frame| for implementing audio mute:
// |previous_frame_muted| && |current_frame_muted| - Zero out whole frame.
// |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start.
// !|previous_frame_muted| && |current_frame_muted| - Fade-out at frame end.
// !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched.
static void Mute(AudioFrame* frame,
bool previous_frame_muted,
bool current_frame_muted);
// Zero out contents of frame.
static void Mute(AudioFrame* frame);
// Halve samples in |frame|.
static void ApplyHalfGain(AudioFrame* frame);
static int Scale(float left, float right, AudioFrame& frame);
static int ScaleWithSat(float scale, AudioFrame& frame);
};
} // namespace webrtc
#endif // WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_

22
webrtc/modules/utility/source/audio_frame_operations_unittest.cc → webrtc/audio/utility/audio_frame_operations_unittest.cc
View File

@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/test/gtest.h"
namespace webrtc {
@ -365,5 +365,25 @@ TEST_F(AudioFrameOperationsTest, ScaleWithSatSucceeds) {
VerifyFramesAreEqual(scaled_frame, frame_);
}
TEST_F(AudioFrameOperationsTest, AddingXToEmptyGivesX) {
// When samples_per_channel_ is 0, the frame counts as empty and zero.
AudioFrame frame_to_add_to;
frame_to_add_to.samples_per_channel_ = 0;
frame_to_add_to.num_channels_ = frame_.num_channels_;
AudioFrameOperations::Add(frame_, &frame_to_add_to);
VerifyFramesAreEqual(frame_, frame_to_add_to);
}
TEST_F(AudioFrameOperationsTest, AddingTwoFramesProducesTheirSum) {
AudioFrame frame_to_add_to;
frame_to_add_to.samples_per_channel_ = frame_.samples_per_channel_;
frame_to_add_to.num_channels_ = frame_.num_channels_;
SetFrameData(&frame_to_add_to, 1000);
AudioFrameOperations::Add(frame_, &frame_to_add_to);
SetFrameData(&frame_, frame_.data_[0] + 1000);
VerifyFramesAreEqual(frame_, frame_to_add_to);
}
} // namespace
} // namespace webrtc

1
webrtc/modules/BUILD.gn
View File

@ -483,7 +483,6 @@ if (rtc_include_tests) {
"rtp_rtcp/test/testAPI/test_api_audio.cc",
"rtp_rtcp/test/testAPI/test_api_rtcp.cc",
"rtp_rtcp/test/testAPI/test_api_video.cc",
"utility/source/audio_frame_operations_unittest.cc",
"utility/source/file_player_unittests.cc",
"utility/source/process_thread_impl_unittest.cc",
"video_coding/codecs/test/packet_manipulator_unittest.cc",

2
webrtc/modules/audio_conference_mixer/BUILD.gn
View File

@ -39,8 +39,8 @@ rtc_static_library("audio_conference_mixer") {
}
deps = [
"../../audio/utility:audio_frame_operations",
"../../system_wrappers",
"../audio_processing",
"../utility",
]
}

1
webrtc/modules/audio_conference_mixer/DEPS
View File

@ -1,4 +1,5 @@
include_rules = [
"+webrtc/audio/utility/audio_frame_operations.h",
"+webrtc/base",
"+webrtc/system_wrappers",
]

14
webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
View File

@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h"
#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.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"
#include "webrtc/system_wrappers/include/trace.h"
@ -38,9 +38,9 @@ typedef std::list<ParticipantFrameStruct*> ParticipantFrameStructList;
void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) {
assert(mixed_frame->num_channels_ >= frame->num_channels_);
if (use_limiter) {
// Divide by two to avoid saturation in the mixing.
// This is only meaningful if the limiter will be used.
*frame >>= 1;
// This is to avoid saturation in the mixing. It is only
// meaningful if the limiter will be used.
AudioFrameOperations::ApplyHalfGain(frame);
}
if (mixed_frame->num_channels_ > frame->num_channels_) {
// We only support mono-to-stereo.
@ -49,7 +49,7 @@ void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) {
AudioFrameOperations::MonoToStereo(frame);
}
*mixed_frame += *frame;
AudioFrameOperations::Add(*frame, mixed_frame);
}
// Return the max number of channels from a |list| composed of AudioFrames.
@ -303,7 +303,7 @@ void AudioConferenceMixerImpl::Process() {
if(mixedAudio->samples_per_channel_ == 0) {
// Nothing was mixed, set the audio samples to silence.
mixedAudio->samples_per_channel_ = _sampleSize;
mixedAudio->Mute();
AudioFrameOperations::Mute(mixedAudio);
} else {
// Only call the limiter if we have something to mix.
LimitMixedAudio(mixedAudio);
@ -922,7 +922,7 @@ bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame* mixedAudio) const {
//
// Instead we double the frame (with addition since left-shifting a
// negative value is undefined).
*mixedAudio += *mixedAudio;
AudioFrameOperations::Add(*mixedAudio, mixedAudio);
if(error != _limiter->kNoError) {
WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,

6
webrtc/modules/audio_mixer/BUILD.gn
View File

@ -32,10 +32,10 @@ rtc_static_library("audio_mixer_impl") {
deps = [
":audio_frame_manipulator",
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
"../../base:rtc_base_approved",
"../../modules/audio_processing",
"../../modules/utility",
"../../system_wrappers",
"../audio_processing",
]
}
@ -51,7 +51,7 @@ rtc_static_library("audio_frame_manipulator") {
]
deps = [
"../../audio/utility",
"../../base:rtc_base_approved",
"../../modules/utility",
]
}

1
webrtc/modules/audio_mixer/DEPS
View File

@ -1,4 +1,5 @@
include_rules = [
"+webrtc/audio/utility/audio_frame_operations.h",
"+webrtc/base",
"+webrtc/call",
"+webrtc/common_audio",

2
webrtc/modules/audio_mixer/audio_frame_manipulator.cc
View File

@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
namespace webrtc {

14
webrtc/modules/audio_mixer/audio_mixer_impl.cc
View File

@ -14,9 +14,9 @@
#include <functional>
#include <utility>
#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
namespace webrtc {
namespace {
@ -106,12 +106,12 @@ int32_t MixFromList(AudioFrame* mixed_audio,
// Mix |f.frame| into |mixed_audio|, with saturation protection.
// These effect is applied to |f.frame| itself prior to mixing.
if (use_limiter) {
// Divide by two to avoid saturation in the mixing.
// This is only meaningful if the limiter will be used.
*frame >>= 1;
// This is to avoid saturation in the mixing. It is only
// meaningful if the limiter will be used.
AudioFrameOperations::ApplyHalfGain(frame);
}
RTC_DCHECK_EQ(frame->num_channels_, mixed_audio->num_channels_);
*mixed_audio += *frame;
AudioFrameOperations::Add(*frame, mixed_audio);
}
return 0;
}
@ -250,7 +250,7 @@ void AudioMixerImpl::Mix(size_t number_of_channels,
if (audio_frame_for_mixing->samples_per_channel_ == 0) {
// Nothing was mixed, set the audio samples to silence.
audio_frame_for_mixing->samples_per_channel_ = sample_size_;
audio_frame_for_mixing->Mute();
AudioFrameOperations::Mute(audio_frame_for_mixing);
} else {
// Only call the limiter if we have something to mix.
LimitMixedAudio(audio_frame_for_mixing);
@ -357,7 +357,7 @@ bool AudioMixerImpl::LimitMixedAudio(AudioFrame* mixed_audio) const {
//
// Instead we double the frame (with addition since left-shifting a
// negative value is undefined).
*mixed_audio += *mixed_audio;
AudioFrameOperations::Add(*mixed_audio, mixed_audio);
if (error != limiter_->kNoError) {
LOG_F(LS_ERROR) << "Error from AudioProcessing: " << error;

1
webrtc/modules/audio_processing/BUILD.gn
View File

@ -165,6 +165,7 @@ rtc_static_library("audio_processing") {
defines = []
deps = [
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
"../audio_coding:isac",
]

1
webrtc/modules/audio_processing/DEPS
View File

@ -1,4 +1,5 @@
include_rules = [
"+webrtc/audio/utility/audio_frame_operations.h",
"+webrtc/base",
"+webrtc/common_audio",
"+webrtc/system_wrappers",

2
webrtc/modules/audio_processing/vad/standalone_vad.cc
View File

@ -10,9 +10,9 @@
#include "webrtc/modules/audio_processing/vad/standalone_vad.h"
#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/typedefs.h"
namespace webrtc {

46
webrtc/modules/include/module_common_types.h
View File

@ -18,6 +18,8 @@
#include <limits>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/deprecation.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_types.h"
#include "webrtc/common_video/rotation.h"
#include "webrtc/typedefs.h"
@ -520,8 +522,6 @@ class CallStatsObserver {
*
* - Stereo data is interleaved starting with the left channel.
*
* - The +operator assume that you would never add exactly opposite frames when
* deciding the resulting state. To do this use the -operator.
*/
class AudioFrame {
public:
@ -556,26 +556,29 @@ class AudioFrame {
void CopyFrom(const AudioFrame& src);
void Mute();
AudioFrame& operator>>=(const int rhs);
AudioFrame& operator+=(const AudioFrame& rhs);
// These methods are deprecated. Use the functions in
// webrtc/audio/utility instead. These methods will exists for a
// short period of time until webrtc clients have updated. See
// webrtc:6548 for details.
RTC_DEPRECATED void Mute();
RTC_DEPRECATED AudioFrame& operator>>=(const int rhs);
RTC_DEPRECATED AudioFrame& operator+=(const AudioFrame& rhs);
int id_;
// RTP timestamp of the first sample in the AudioFrame.
uint32_t timestamp_;
uint32_t timestamp_ = 0;
// Time since the first frame in milliseconds.
// -1 represents an uninitialized value.
int64_t elapsed_time_ms_;
int64_t elapsed_time_ms_ = -1;
// NTP time of the estimated capture time in local timebase in milliseconds.
// -1 represents an uninitialized value.
int64_t ntp_time_ms_;
int64_t ntp_time_ms_ = -1;
int16_t data_[kMaxDataSizeSamples];
size_t samples_per_channel_;
int sample_rate_hz_;
size_t num_channels_;
SpeechType speech_type_;
VADActivity vad_activity_;
size_t samples_per_channel_ = 0;
int sample_rate_hz_ = 0;
size_t num_channels_ = 0;
SpeechType speech_type_ = kUndefined;
VADActivity vad_activity_ = kVadUnknown;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
@ -585,7 +588,6 @@ class AudioFrame {
// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
inline AudioFrame::AudioFrame()
: data_() {
Reset();
}
inline void AudioFrame::Reset() {
@ -659,18 +661,6 @@ inline AudioFrame& AudioFrame::operator>>=(const int rhs) {
return *this;
}
namespace {
inline int16_t ClampToInt16(int32_t input) {
if (input < -0x00008000) {
return -0x8000;
} else if (input > 0x00007FFF) {
return 0x7FFF;
} else {
return static_cast<int16_t>(input);
}
}
}
inline AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
// Sanity check
assert((num_channels_ > 0) && (num_channels_ < 3));
@ -704,7 +694,7 @@ inline AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
int32_t wrap_guard =
static_cast<int32_t>(data_[i]) + static_cast<int32_t>(rhs.data_[i]);
data_[i] = ClampToInt16(wrap_guard);
data_[i] = rtc::saturated_cast<int16_t>(wrap_guard);
}
}
return *this;

9
webrtc/modules/module_common_types_unittest.cc
View File

@ -112,15 +112,6 @@ TEST(LatestTimestamp, Wrap) {
EXPECT_EQ(0x0000FFFFu, LatestTimestamp(0xFFFF0000, 0x0000FFFF));
}
TEST(ClampToInt16, TestCases) {
EXPECT_EQ(0x0000, ClampToInt16(0x00000000));
EXPECT_EQ(0x0001, ClampToInt16(0x00000001));
EXPECT_EQ(0x7FFF, ClampToInt16(0x00007FFF));
EXPECT_EQ(0x7FFF, ClampToInt16(0x7FFFFFFF));
EXPECT_EQ(-0x0001, ClampToInt16(-0x00000001));
EXPECT_EQ(-0x8000, ClampToInt16(-0x8000));
EXPECT_EQ(-0x8000, ClampToInt16(-0x7FFFFFFF));
}
TEST(SequenceNumberUnwrapper, Limits) {
SequenceNumberUnwrapper unwrapper;

4
webrtc/modules/utility/BUILD.gn
View File

@ -16,7 +16,6 @@ rtc_static_library("utility") {
"include/helpers_android.h",
"include/jvm_android.h",
"include/process_thread.h",
"source/audio_frame_operations.cc",
"source/coder.cc",
"source/coder.h",
"source/file_player.cc",
@ -38,10 +37,13 @@ rtc_static_library("utility") {
deps = [
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
"../../base:rtc_task_queue",
"../../common_audio",
"../../system_wrappers",
"../audio_coding",
"../audio_coding:builtin_audio_decoder_factory",
"../audio_coding:rent_a_codec",
"../media_file",
]
}

3
webrtc/modules/utility/DEPS
View File

@ -1,4 +1,7 @@
include_rules = [
# TODO(aleloi): remove when clients update. See
# bugs.webrtc.org/6548.
"+webrtc/audio/utility/audio_frame_operations.h",
"+webrtc/base",
"+webrtc/common_audio",
"+webrtc/common_video",

53
webrtc/modules/utility/include/audio_frame_operations.h
View File

@ -10,54 +10,11 @@
#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
// The contents of this file have moved to
// //webrtc/audio/utility. This file is deprecated.
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioFrame;
// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
// Change reference parameters to pointers. Consider using a namespace rather
// than a class.
class AudioFrameOperations {
public:
// Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place
// operation, meaning src_audio and dst_audio must point to different
// buffers. It is the caller's responsibility to ensure that |dst_audio| is
// sufficiently large.
static void MonoToStereo(const int16_t* src_audio, size_t samples_per_channel,
int16_t* dst_audio);
// |frame.num_channels_| will be updated. This version checks for sufficient
// buffer size and that |num_channels_| is mono.
static int MonoToStereo(AudioFrame* frame);
// Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place
// operation, meaning |src_audio| and |dst_audio| may point to the same
// buffer.
static void StereoToMono(const int16_t* src_audio, size_t samples_per_channel,
int16_t* dst_audio);
// |frame.num_channels_| will be updated. This version checks that
// |num_channels_| is stereo.
static int StereoToMono(AudioFrame* frame);
// Swap the left and right channels of |frame|. Fails silently if |frame| is
// not stereo.
static void SwapStereoChannels(AudioFrame* frame);
// Conditionally zero out contents of |frame| for implementing audio mute:
// |previous_frame_muted| && |current_frame_muted| - Zero out whole frame.
// |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start.
// !|previous_frame_muted| && |current_frame_muted| - Fade-out at frame end.
// !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched.
static void Mute(AudioFrame* frame, bool previous_frame_muted,
bool current_frame_muted);
static int Scale(float left, float right, AudioFrame& frame);
static int ScaleWithSat(float scale, AudioFrame& frame);
};
} // namespace webrtc
// TODO(aleloi): Remove this file when clients have updated their
// includes. See bugs.webrtc.org/6548.
#include "webrtc/audio/utility/audio_frame_operations.h"
#endif // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_

1
webrtc/voice_engine/BUILD.gn
View File

@ -90,6 +90,7 @@ rtc_static_library("voice_engine") {
"../api:audio_mixer_api",
"../api:call_api",
"../api:transport_api",
"../audio/utility:audio_frame_operations",
"../base:rtc_base_approved",
"../common_audio",
"../logging:rtc_event_log_api",

1
webrtc/voice_engine/DEPS
View File

@ -1,4 +1,5 @@
include_rules = [
"+webrtc/audio/utility/audio_frame_operations.h",
"+webrtc/base",
"+webrtc/call",
"+webrtc/common_audio",

4
webrtc/voice_engine/channel.cc
View File

@ -13,6 +13,7 @@
#include <algorithm>
#include <utility>
#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/base/array_view.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/criticalsection.h"
@ -32,7 +33,6 @@
#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/include/voe_external_media.h"
@ -613,7 +613,7 @@ MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
// TODO(henrik.lundin): We should be able to do better than this. But we
// will have to go through all the cases below where the audio samples may
// be used, and handle the muted case in some way.
audioFrame->Mute();
AudioFrameOperations::Mute(audioFrame);
}
// Convert module ID to internal VoE channel ID

2
webrtc/voice_engine/output_mixer.cc
View File

@ -10,9 +10,9 @@
#include "webrtc/voice_engine/output_mixer.h"
#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/include/voe_external_media.h"

2
webrtc/voice_engine/transmit_mixer.cc
View File

@ -12,9 +12,9 @@
#include <memory>
#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/channel.h"

2
webrtc/voice_engine/utility.cc
View File

@ -10,13 +10,13 @@
#include "webrtc/voice_engine/utility.h"
#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
namespace webrtc {

3
webrtc/voice_engine/voe_external_media_impl.cc
View File

@ -10,6 +10,7 @@
#include "webrtc/voice_engine/voe_external_media_impl.h"
#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/channel.h"
#include "webrtc/voice_engine/include/voe_errors.h"
@ -149,7 +150,7 @@ int VoEExternalMediaImpl::GetAudioFrame(int channel, int desired_sample_rate_hz,
desired_sample_rate_hz == 0 ? -1 : desired_sample_rate_hz;
auto ret = channelPtr->GetAudioFrameWithMuted(channel, frame);
if (ret == MixerParticipant::AudioFrameInfo::kMuted) {
frame->Mute();
AudioFrameOperations::Mute(frame);
}
return ret == MixerParticipant::AudioFrameInfo::kError ? -1 : 0;
}