admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Split audio mixer into interface and implementation.

Browse Source 
The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules.

This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC.

It will also create less build dependencies when the new mixer has replaced the old one.

NOTRY=True
TBR=henrik.lundin@webrtc.org
BUG=webrtc:6346

Review-Url: https://codereview.webrtc.org/2411313003
Cr-Commit-Position: refs/heads/master@{#14705}
This commit is contained in:
aleloi 2016-10-20 05:06:39 -07:00 committed by Commit bot
parent 76648da8dc
commit 201dfe90a7
12 changed files with 87 additions and 35 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
.gn
View File

@ -20,6 +20,7 @@ secondary_source = "//build/secondary/"
# "gn check" or "gn gen --check".
# TODO(kjellander): Keep adding paths to this list as work in webrtc:5589 is done.
check_targets = [
"//webrtc/api:audio_mixer_api",
"//webrtc/api:rtc_stats_api",
"//webrtc/modules/audio_coding:g711_test",
"//webrtc/modules/audio_coding:g722_test",

11
webrtc/api/BUILD.gn
View File

@ -321,6 +321,17 @@ rtc_source_set("rtc_stats_api") {
]
}
# GYP version: webrtc/api/api.gyp:audio_mixer_api
rtc_source_set("audio_mixer_api") {
sources = [
"audio/audio_mixer.h",
]
deps = [
"../base:rtc_base_approved",
]
}
if (rtc_include_tests) {
config("peerconnection_unittests_config") {
# The warnings below are enabled by default. Since GN orders compiler flags

11
webrtc/api/api.gyp
View File

@ -232,5 +232,16 @@
'stats/rtcstatsreport.h',
],
}, # target rtc_stats_api
{
# GN version: webrtc/api:audio_mixer_api
'target_name': 'audio_mixer_api',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
],
'sources': [
'audio/audio_mixer.h',
],
}, # target rtc_stats_api
], # targets
}

22
webrtc/modules/audio_mixer/audio_mixer.h → webrtc/api/audio/audio_mixer.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_H_
#define WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_H_
#ifndef WEBRTC_API_AUDIO_AUDIO_MIXER_H_
#define WEBRTC_API_AUDIO_AUDIO_MIXER_H_
#include <memory>
@ -18,6 +18,9 @@
namespace webrtc {
// WORK IN PROGRESS
// This class is under development and is not yet intended for for use outside
// of WebRtc/Libjingle.
class AudioMixer : public rtc::RefCountInterface {
public:
// A callback class that all mixer participants must inherit from/implement.
@ -25,10 +28,11 @@ class AudioMixer : public rtc::RefCountInterface {
public:
enum class AudioFrameInfo {
kNormal, // The samples in audio_frame are valid and should be used.
kMuted, // The samples in audio_frame should not be used, but should be
// implicitly interpreted as zero. Other fields in audio_frame
// may be read and should contain meaningful values.
kError // audio_frame will not be used.
kMuted, // The samples in audio_frame should not be used, but
// should be implicitly interpreted as zero. Other
// fields in audio_frame may be read and should
// contain meaningful values.
kError, // The audio_frame will not be used.
};
struct AudioFrameWithInfo {
@ -47,8 +51,8 @@ class AudioMixer : public rtc::RefCountInterface {
// mixer.
virtual AudioFrameWithInfo GetAudioFrameWithInfo(int sample_rate_hz) = 0;
// A way for a mixer implementation do distinguish participants.
virtual int ssrc() = 0;
// A way for a mixer implementation to distinguish participants.
virtual int Ssrc() = 0;
protected:
virtual ~Source() {}
@ -75,4 +79,4 @@ class AudioMixer : public rtc::RefCountInterface {
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_H_
#endif // WEBRTC_API_AUDIO_AUDIO_MIXER_H_

6
webrtc/modules/BUILD.gn
View File

@ -19,7 +19,7 @@ group("modules") {
"audio_coding",
"audio_conference_mixer",
"audio_device",
"audio_mixer",
"audio_mixer:audio_mixer_impl",
"audio_processing",
"bitrate_controller",
"desktop_capture",
@ -629,6 +629,7 @@ if (rtc_include_tests) {
"../system_wrappers:system_wrappers",
"../test:rtp_test_utils",
"../test:test_common",
"../test:test_support",
"../test:test_support_main",
"../test:video_test_common",
"audio_coding",
@ -645,7 +646,8 @@ if (rtc_include_tests) {
"audio_coding:webrtc_opus",
"audio_conference_mixer",
"audio_device",
"audio_mixer",
"audio_mixer:audio_frame_manipulator",
"audio_mixer:audio_mixer_impl",
"audio_processing",
"audio_processing:audioproc_test_utils",
"bitrate_controller",

38
webrtc/modules/audio_mixer/BUILD.gn
View File

@ -8,15 +8,12 @@
import("../../build/webrtc.gni")
config("audio_conference_mixer_config") {
include_dirs = [ "../../modules/include" ]
}
rtc_static_library("audio_mixer") {
rtc_static_library("audio_mixer_impl") {
visibility = [
"../../audio:audio",
"../../modules/*",
]
sources = [
"audio_frame_manipulator.cc",
"audio_frame_manipulator.h",
"audio_mixer.h",
"audio_mixer_impl.cc",
"audio_mixer_impl.h",
"audio_source_with_mix_status.cc",
@ -24,17 +21,36 @@ rtc_static_library("audio_mixer") {
]
public = [
"audio_mixer.h",
"audio_mixer_impl.h",
]
public_configs = [ ":audio_conference_mixer_config" ]
public_deps = [
"../../api:audio_mixer_api",
]
deps = [
":audio_frame_manipulator",
"../..:webrtc_common",
"../../base:rtc_base_approved",
"../../modules/audio_processing",
"../../modules/utility",
"../../system_wrappers",
"../../voice_engine:level_indicator",
]
}
rtc_static_library("audio_frame_manipulator") {
visibility = [
":*",
"../../modules:*",
]
sources = [
"audio_frame_manipulator.cc",
"audio_frame_manipulator.h",
]
deps = [
"../../base:rtc_base_approved",
"../../modules/utility",
]
}

1
webrtc/modules/audio_mixer/audio_frame_manipulator.cc
View File

@ -12,7 +12,6 @@
#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"
#include "webrtc/typedefs.h"
namespace webrtc {

1
webrtc/modules/audio_mixer/audio_frame_manipulator.h
View File

@ -12,7 +12,6 @@
#define WEBRTC_MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {

22
webrtc/modules/audio_mixer/audio_mixer.gypi
View File

@ -9,24 +9,34 @@
{
'targets': [
{
'target_name': 'audio_mixer',
'target_name': 'audio_mixer_impl',
'type': 'static_library',
'dependencies': [
'audio_frame_manipulator',
'audio_processing',
'webrtc_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/api/api.gyp:audio_mixer_api',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/voice_engine/voice_engine.gyp:level_indicator',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'audio_frame_manipulator.cc',
'audio_frame_manipulator.h',
'audio_mixer.h',
'audio_mixer_impl.cc',
'audio_mixer_impl.h',
'audio_source_with_mix_status.cc',
'audio_source_with_mix_status.h',
],
},
{
'target_name': 'audio_frame_manipulator',
'type': 'static_library',
'dependencies': [
'webrtc_utility',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
],
'sources': [
'audio_frame_manipulator.cc',
'audio_frame_manipulator.h',
],
},
], # targets
}

3
webrtc/modules/audio_mixer/audio_mixer_impl.h
View File

@ -14,14 +14,13 @@
#include <memory>
#include <vector>
#include "webrtc/api/audio/audio_mixer.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/audio_mixer/audio_mixer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/voice_engine/level_indicator.h"
#include "webrtc/voice_engine_configurations.h"
namespace webrtc {

4
webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc
View File

@ -14,10 +14,10 @@
#include <memory>
#include <utility>
#include "webrtc/api/audio/audio_mixer.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/thread.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
#include "webrtc/modules/audio_mixer/audio_mixer.h"
#include "webrtc/test/gmock.h"
using testing::_;
@ -60,7 +60,7 @@ class MockMixerAudioSource : public AudioMixer::Source {
MOCK_METHOD1(GetAudioFrameWithInfo, AudioFrameWithInfo(int sample_rate_hz));
MOCK_METHOD0(ssrc, int());
MOCK_METHOD0(Ssrc, int());
AudioFrame* fake_frame() { return &fake_frame_; }
AudioFrameInfo fake_info() { return fake_audio_frame_info_; }

2
webrtc/modules/audio_mixer/audio_source_with_mix_status.h
View File

@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_MIXER_AUDIO_SOURCE_WITH_MIX_STATUS_H_
#define WEBRTC_MODULES_AUDIO_MIXER_AUDIO_SOURCE_WITH_MIX_STATUS_H_
#include "webrtc/modules/audio_mixer/audio_mixer.h"
#include "webrtc/api/audio/audio_mixer.h"
namespace webrtc {