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Rename Call::Config to CallConfig, keep old name as alias.

Browse Source 
We want api/peerconnectioninterface.h (and corresponding build target)
to not depend on call.h, and generally we treat Call as an internal,
non-api, class. But we need CallFactoryInterface in the api in order to
enable use of PeerConnection with or without support for media.

Making CallConfig a top-level class makes it possible to forward declare
it, together with Call, for use in callfactoryinterface.h and
peerconnectioninterface.h.

Delete the peerconnection_and_implicit_call_api target, replaced by
new target callfactory_api, to link between Call and Peerconnection.

Bug: webrtc:7504
Change-Id: I5e3978ef89bcd6705e94536f8676bcf89fc82fe1
Reviewed-on: https://webrtc-review.googlesource.com/46201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22020}
This commit is contained in:
Niels Möller
2018-02-14 12:20:13 +01:00
committed by Commit Bot
parent fa4fe647ed
commit 8366e177e7
16 changed files with 88 additions and 76 deletions
Download Patch File Download Diff File Expand all files Collapse all files

26
api/BUILD.gn
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@ -37,6 +37,13 @@ rtc_source_set("call_api") {
]
}
rtc_source_set("callfactory_api") {
visibility = [ "*" ]
sources = [
"call/callfactoryinterface.h",
]
}
rtc_static_library("libjingle_peerconnection_api") {
visibility = [ "*" ]
cflags = []
@ -60,6 +67,7 @@ rtc_static_library("libjingle_peerconnection_api") {
"mediatypes.h",
"notifier.h",
"peerconnectionfactoryproxy.h",
"peerconnectioninterface.h",
"peerconnectionproxy.h",
"proxy.cc",
"proxy.h",
@ -94,8 +102,9 @@ rtc_static_library("libjingle_peerconnection_api") {
":array_view",
":audio_mixer_api",
":audio_options_api",
":callfactory_api",
":libjingle_logging_api",
":optional",
":peerconnection_and_implicit_call_api",
":rtc_stats_api",
":video_frame_api",
"audio_codecs:audio_codecs_api",
@ -105,6 +114,7 @@ rtc_static_library("libjingle_peerconnection_api") {
# file, really. All these should arguably go away in time.
"..:typedefs",
"..:webrtc_common",
"../logging:rtc_event_log_api",
"../media:rtc_media_config",
"../modules/audio_processing:audio_processing_statistics",
"../rtc_base:checks",
@ -113,24 +123,17 @@ rtc_static_library("libjingle_peerconnection_api") {
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
]
if (is_nacl) {
# This is needed by .h files included from rtc_base.
deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
}
}
# TODO(bugs.webrtc.org/7504): Dummy target, kept temporarily because
# chromium edpends on it.
rtc_source_set("peerconnection_and_implicit_call_api") {
visibility = [ "*" ]
# The peerconnectioninterface.h file pulls in call/callfactoryinterface.h
# and the entire call module with it. We need to either get rid of this
# dependency or pull most of call/ into the API. For now, silence the warnings
# this creates since it creates a circular dependency (call very much depends
# on API). See bugs.webrtc.org/8667.
check_includes = false
sources = [
"peerconnectioninterface.h",
]
}
rtc_source_set("libjingle_logging_api") {
@ -375,7 +378,6 @@ if (rtc_include_tests) {
"fakemetricsobserver.h",
]
deps = [
"../api:peerconnection_and_implicit_call_api",
"../media:rtc_media_base",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",

1
api/DEPS
View File

@ -4,6 +4,7 @@ include_rules = [
"+media",
"+p2p",
"+pc",
"+logging/rtc_event_log/rtc_event_log_factory_interface.h",
]
specific_include_rules = {

14
call/callfactoryinterface.h → api/call/callfactoryinterface.h
View File

@ -8,15 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef CALL_CALLFACTORYINTERFACE_H_
#define CALL_CALLFACTORYINTERFACE_H_
#ifndef API_CALL_CALLFACTORYINTERFACE_H_
#define API_CALL_CALLFACTORYINTERFACE_H_
#include <memory>
#include "call/call.h"
namespace webrtc {
// These classes are not part of the API, and are treated as opaque pointers.
class Call;
struct CallConfig;
// This interface exists to allow webrtc to be optionally built without media
// support (i.e., if only being used for data channels). PeerConnectionFactory
// is constructed with a CallFactoryInterface, which may or may not be null.
@ -24,11 +26,11 @@ class CallFactoryInterface {
public:
virtual ~CallFactoryInterface() {}
virtual Call* CreateCall(const Call::Config& config) = 0;
virtual Call* CreateCall(const CallConfig& config) = 0;
};
std::unique_ptr<CallFactoryInterface> CreateCallFactory();
} // namespace webrtc
#endif // CALL_CALLFACTORYINTERFACE_H_
#endif // API_CALL_CALLFACTORYINTERFACE_H_

16
api/peerconnectioninterface.h
View File

@ -80,6 +80,7 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_options.h"
#include "api/call/callfactoryinterface.h"
#include "api/datachannelinterface.h"
#include "api/dtmfsenderinterface.h"
#include "api/jsep.h"
@ -94,12 +95,19 @@
#include "api/statstypes.h"
#include "api/turncustomizer.h"
#include "api/umametrics.h"
#include "call/callfactoryinterface.h"
#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "media/base/mediaconfig.h"
#include "media/base/videocapturer.h"
#include "p2p/base/portallocator.h"
// TODO(bugs.webrtc.org/6353): cricket::VideoCapturer is deprecated and should
// be deleted from the PeerConnection api.
#include "media/base/videocapturer.h" // nogncheck
// TODO(bugs.webrtc.org/7447): We plan to provide a way to let applications
// inject a PacketSocketFactory and/or NetworkManager, and not expose
// PortAllocator in the PeerConnection api.
#include "p2p/base/portallocator.h" // nogncheck
// TODO(nisse): The interface for bitrate allocation strategy belongs in api/.
#include "rtc_base/bitrateallocationstrategy.h"
#include "rtc_base/network.h"
#include "rtc_base/platform_file.h"
#include "rtc_base/rtccertificate.h"
#include "rtc_base/rtccertificategenerator.h"
#include "rtc_base/socketaddress.h"
@ -119,7 +127,7 @@ class WebRtcVideoEncoderFactory;
namespace webrtc {
class AudioDeviceModule;
class AudioMixer;
class CallFactoryInterface;
class AudioProcessing;
class MediaConstraintsInterface;
class VideoDecoderFactory;
class VideoEncoderFactory;

2
call/BUILD.gn
View File

@ -14,7 +14,6 @@ rtc_source_set("call_interfaces") {
"audio_send_stream.h",
"audio_state.h",
"call.h",
"callfactoryinterface.h",
"flexfec_receive_stream.h",
"syncable.cc",
"syncable.h",
@ -139,6 +138,7 @@ rtc_static_library("call") {
":rtp_sender",
":video_stream_api",
"..:webrtc_common",
"../api:callfactory_api",
"../api:optional",
"../api:transport_api",
"../audio",

18
call/call.h
View File

@ -72,13 +72,8 @@ class PacketReceiver {
virtual ~PacketReceiver() {}
};
// A Call instance can contain several send and/or receive streams. All streams
// are assumed to have the same remote endpoint and will share bitrate estimates
// etc.
class Call {
public:
struct Config {
explicit Config(RtcEventLog* event_log) : event_log(event_log) {
struct CallConfig {
explicit CallConfig(RtcEventLog* event_log) : event_log(event_log) {
RTC_DCHECK(event_log);
}
@ -114,7 +109,14 @@ class Call {
// RtcEventLog to use for this call. Required.
// Use webrtc::RtcEventLog::CreateNull() for a null implementation.
RtcEventLog* event_log = nullptr;
};
};
// A Call instance can contain several send and/or receive streams. All streams
// are assumed to have the same remote endpoint and will share bitrate estimates
// etc.
class Call {
public:
using Config = CallConfig;
struct Stats {
std::string ToString(int64_t time_ms) const;

2
call/callfactory.cc
View File

@ -12,6 +12,8 @@
#include <memory>
#include "call/call.h"
namespace webrtc {
Call* CallFactory::CreateCall(const Call::Config& config) {

4
call/callfactory.h
View File

@ -11,14 +11,14 @@
#ifndef CALL_CALLFACTORY_H_
#define CALL_CALLFACTORY_H_
#include "call/callfactoryinterface.h"
#include "api/call/callfactoryinterface.h"
namespace webrtc {
class CallFactory : public CallFactoryInterface {
~CallFactory() override {}
Call* CreateCall(const Call::Config& config) override;
Call* CreateCall(const CallConfig& config) override;
};
} // namespace webrtc

3
examples/BUILD.gn
View File

@ -558,8 +558,8 @@ if (is_linux || is_win) {
configs += [ ":peerconnection_client_warnings_config" ]
deps += [
"../api:libjingle_peerconnection_api",
"../api:libjingle_peerconnection_test_api",
"../api:peerconnection_and_implicit_call_api",
"../api:video_frame_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
@ -692,7 +692,6 @@ if (is_win || is_android) {
deps = [
"../api:libjingle_peerconnection_api",
"../api:libjingle_peerconnection_test_api",
"../api:peerconnection_and_implicit_call_api",
"../api:video_frame_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",

6
pc/BUILD.gn
View File

@ -182,7 +182,6 @@ rtc_static_library("peerconnection") {
"../api:call_api",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:peerconnection_and_implicit_call_api",
"../api:rtc_stats_api",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
@ -216,8 +215,8 @@ rtc_static_library("create_pc_factory") {
deps = [
"../api:audio_mixer_api",
"../api:callfactory_api",
"../api:libjingle_peerconnection_api",
"../api:peerconnection_and_implicit_call_api",
"../api/audio_codecs:audio_codecs_api",
"../api/video_codecs:video_codecs_api",
"../call",
@ -354,7 +353,6 @@ if (rtc_include_tests) {
"..:webrtc_common",
"../api:libjingle_peerconnection_api",
"../api:libjingle_peerconnection_test_api",
"../api:peerconnection_and_implicit_call_api",
"../api:rtc_stats_api",
"../call:call_interfaces",
"../logging:rtc_event_log_api",
@ -462,7 +460,6 @@ if (rtc_include_tests) {
":rtc_pc_base",
"../api:libjingle_peerconnection_api",
"../api:mock_rtp",
"../api:peerconnection_and_implicit_call_api",
"../rtc_base:checks",
"../rtc_base:stringutils",
]
@ -474,6 +471,7 @@ if (rtc_include_tests) {
":libjingle_peerconnection",
":pc_test_utils",
"..:webrtc_common",
"../api:callfactory_api",
"../api:fakemetricsobserver",
"../api:libjingle_peerconnection_test_api",
"../api:optional",

2
pc/createpeerconnectionfactory.cc
View File

@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "api/call/callfactoryinterface.h"
#include "api/peerconnectioninterface.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "call/callfactoryinterface.h"
#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "media/engine/webrtcmediaengine.h"
#include "modules/audio_device/include/audio_device.h"

2
pc/peerconnection_media_unittest.cc
View File

@ -14,7 +14,7 @@
#include <tuple>
#include "call/callfactoryinterface.h"
#include "api/call/callfactoryinterface.h"
#include "logging/rtc_event_log/rtc_event_log_factory.h"
#include "media/base/fakemediaengine.h"
#include "p2p/base/fakeportallocator.h"

1
pc/peerconnectioninternal.h
View File

@ -18,6 +18,7 @@
#include <vector>
#include "api/peerconnectioninterface.h"
#include "call/call.h"
#include "pc/datachannel.h"
#include "pc/rtptransceiver.h"

3
sdk/BUILD.gn
View File

@ -340,7 +340,6 @@ if (is_ios || is_mac) {
":videotoolbox_objc",
":videotracksource_objc",
"../api:libjingle_peerconnection_api",
"../api:peerconnection_and_implicit_call_api",
"../api:video_frame_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
@ -438,7 +437,6 @@ if (is_ios || is_mac) {
":native_video",
":peerconnectionfactory_base_objc",
"../api:libjingle_peerconnection_api",
"../api:peerconnection_and_implicit_call_api",
"../rtc_base:rtc_base",
]
}
@ -569,7 +567,6 @@ if (is_ios || is_mac) {
":native_video",
":videotracksource_objc",
"../api:libjingle_peerconnection_api",
"../api:peerconnection_and_implicit_call_api",
"../api:video_frame_api",
"../api/video_codecs:video_codecs_api",
"../common_video",

2
sdk/android/BUILD.gn
View File

@ -364,6 +364,7 @@ rtc_static_library("media_jni") {
deps = [
":base_jni",
"../../api:callfactory_api",
"../../api/video_codecs:video_codecs_api",
"../../call:call_interfaces",
"../../logging:rtc_event_log_api",
@ -506,7 +507,6 @@ rtc_static_library("peerconnection_jni") {
":native_api_jni",
"../..:webrtc_common",
"../../api:libjingle_peerconnection_api",
"../../api:peerconnection_and_implicit_call_api",
"../../api/video_codecs:video_codecs_api",
"../../logging:rtc_event_log_api",
"../../logging:rtc_event_log_impl_base",

2
sdk/android/src/jni/pc/media.cc
View File

@ -11,9 +11,9 @@
#include <utility>
#include "api/call/callfactoryinterface.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "call/callfactoryinterface.h"
#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "media/engine/webrtcmediaengine.h"
#include "modules/audio_device/include/audio_device.h"