Rename Call::Config to CallConfig, keep old name as alias.
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}diff --git a/api/BUILD.gn b/api/BUILD.gn
index 3952cab..fcd3497 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -37,6 +37,13 @@
]
}
+rtc_source_set("callfactory_api") {
+ visibility = [ "*" ]
+ sources = [
+ "call/callfactoryinterface.h",
+ ]
+}
+
rtc_static_library("libjingle_peerconnection_api") {
visibility = [ "*" ]
cflags = []
@@ -60,6 +67,7 @@
"mediatypes.h",
"notifier.h",
"peerconnectionfactoryproxy.h",
+ "peerconnectioninterface.h",
"peerconnectionproxy.h",
"proxy.cc",
"proxy.h",
@@ -94,8 +102,9 @@
":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 @@
# 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_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 @@
"fakemetricsobserver.h",
]
deps = [
- "../api:peerconnection_and_implicit_call_api",
"../media:rtc_media_base",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
diff --git a/api/DEPS b/api/DEPS
index 01a3e73..9ba7245 100644
--- a/api/DEPS
+++ b/api/DEPS
@@ -4,6 +4,7 @@
"+media",
"+p2p",
"+pc",
+ "+logging/rtc_event_log/rtc_event_log_factory_interface.h",
]
specific_include_rules = {
diff --git a/call/callfactoryinterface.h b/api/call/callfactoryinterface.h
similarity index 74%
rename from call/callfactoryinterface.h
rename to api/call/callfactoryinterface.h
index a3cf6eb..a7f3245 100644
--- a/call/callfactoryinterface.h
+++ b/api/call/callfactoryinterface.h
@@ -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 @@
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_
diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index 22514ac..8b4db9a 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -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 @@
namespace webrtc {
class AudioDeviceModule;
class AudioMixer;
-class CallFactoryInterface;
+class AudioProcessing;
class MediaConstraintsInterface;
class VideoDecoderFactory;
class VideoEncoderFactory;
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 084148b..6be92fd 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -14,7 +14,6 @@
"audio_send_stream.h",
"audio_state.h",
"call.h",
- "callfactoryinterface.h",
"flexfec_receive_stream.h",
"syncable.cc",
"syncable.h",
@@ -139,6 +138,7 @@
":rtp_sender",
":video_stream_api",
"..:webrtc_common",
+ "../api:callfactory_api",
"../api:optional",
"../api:transport_api",
"../audio",
diff --git a/call/call.h b/call/call.h
index eb23e8b..884a21b 100644
--- a/call/call.h
+++ b/call/call.h
@@ -72,49 +72,51 @@
virtual ~PacketReceiver() {}
};
+struct CallConfig {
+ explicit CallConfig(RtcEventLog* event_log) : event_log(event_log) {
+ RTC_DCHECK(event_log);
+ }
+
+ static constexpr int kDefaultStartBitrateBps = 300000;
+
+ // Bitrate config used until valid bitrate estimates are calculated. Also
+ // used to cap total bitrate used. This comes from the remote connection.
+ struct BitrateConfig {
+ int min_bitrate_bps = 0;
+ int start_bitrate_bps = kDefaultStartBitrateBps;
+ int max_bitrate_bps = -1;
+ } bitrate_config;
+
+ // The local client's bitrate preferences. The actual configuration used
+ // is a combination of this and |bitrate_config|. The combination is
+ // currently more complicated than a simple mask operation (see
+ // SetBitrateConfig and SetBitrateConfigMask). Assumes that 0 <= min <=
+ // start <= max holds for set parameters.
+ struct BitrateConfigMask {
+ rtc::Optional<int> min_bitrate_bps;
+ rtc::Optional<int> start_bitrate_bps;
+ rtc::Optional<int> max_bitrate_bps;
+ };
+
+ // AudioState which is possibly shared between multiple calls.
+ // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+ rtc::scoped_refptr<AudioState> audio_state;
+
+ // Audio Processing Module to be used in this call.
+ // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+ AudioProcessing* audio_processing = nullptr;
+
+ // 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:
- struct Config {
- explicit Config(RtcEventLog* event_log) : event_log(event_log) {
- RTC_DCHECK(event_log);
- }
-
- static constexpr int kDefaultStartBitrateBps = 300000;
-
- // Bitrate config used until valid bitrate estimates are calculated. Also
- // used to cap total bitrate used. This comes from the remote connection.
- struct BitrateConfig {
- int min_bitrate_bps = 0;
- int start_bitrate_bps = kDefaultStartBitrateBps;
- int max_bitrate_bps = -1;
- } bitrate_config;
-
- // The local client's bitrate preferences. The actual configuration used
- // is a combination of this and |bitrate_config|. The combination is
- // currently more complicated than a simple mask operation (see
- // SetBitrateConfig and SetBitrateConfigMask). Assumes that 0 <= min <=
- // start <= max holds for set parameters.
- struct BitrateConfigMask {
- rtc::Optional<int> min_bitrate_bps;
- rtc::Optional<int> start_bitrate_bps;
- rtc::Optional<int> max_bitrate_bps;
- };
-
- // AudioState which is possibly shared between multiple calls.
- // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
- rtc::scoped_refptr<AudioState> audio_state;
-
- // Audio Processing Module to be used in this call.
- // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
- AudioProcessing* audio_processing = nullptr;
-
- // RtcEventLog to use for this call. Required.
- // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
- RtcEventLog* event_log = nullptr;
- };
+ using Config = CallConfig;
struct Stats {
std::string ToString(int64_t time_ms) const;
diff --git a/call/callfactory.cc b/call/callfactory.cc
index 82acb65..0f2eecb 100644
--- a/call/callfactory.cc
+++ b/call/callfactory.cc
@@ -12,6 +12,8 @@
#include <memory>
+#include "call/call.h"
+
namespace webrtc {
Call* CallFactory::CreateCall(const Call::Config& config) {
diff --git a/call/callfactory.h b/call/callfactory.h
index 167b82a..1c57bd4 100644
--- a/call/callfactory.h
+++ b/call/callfactory.h
@@ -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
diff --git a/examples/BUILD.gn b/examples/BUILD.gn
index f39e23a..e20ff4d 100644
--- a/examples/BUILD.gn
+++ b/examples/BUILD.gn
@@ -558,8 +558,8 @@
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 @@
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",
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 06166e2..cf506bf 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -182,7 +182,6 @@
"../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 @@
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 @@
"..: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 @@
":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 @@
":libjingle_peerconnection",
":pc_test_utils",
"..:webrtc_common",
+ "../api:callfactory_api",
"../api:fakemetricsobserver",
"../api:libjingle_peerconnection_test_api",
"../api:optional",
diff --git a/pc/createpeerconnectionfactory.cc b/pc/createpeerconnectionfactory.cc
index 48867be..197df8a 100644
--- a/pc/createpeerconnectionfactory.cc
+++ b/pc/createpeerconnectionfactory.cc
@@ -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"
diff --git a/pc/peerconnection_media_unittest.cc b/pc/peerconnection_media_unittest.cc
index cfd4803..3a88e80 100644
--- a/pc/peerconnection_media_unittest.cc
+++ b/pc/peerconnection_media_unittest.cc
@@ -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"
diff --git a/pc/peerconnectioninternal.h b/pc/peerconnectioninternal.h
index 0f942d2..c276b3d 100644
--- a/pc/peerconnectioninternal.h
+++ b/pc/peerconnectioninternal.h
@@ -18,6 +18,7 @@
#include <vector>
#include "api/peerconnectioninterface.h"
+#include "call/call.h"
#include "pc/datachannel.h"
#include "pc/rtptransceiver.h"
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index ddebdce..30bfb0d 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -340,7 +340,6 @@
":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 @@
":native_video",
":peerconnectionfactory_base_objc",
"../api:libjingle_peerconnection_api",
- "../api:peerconnection_and_implicit_call_api",
"../rtc_base:rtc_base",
]
}
@@ -569,7 +567,6 @@
":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",
diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index a1c0cbf..18d9987 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -364,6 +364,7 @@
deps = [
":base_jni",
+ "../../api:callfactory_api",
"../../api/video_codecs:video_codecs_api",
"../../call:call_interfaces",
"../../logging:rtc_event_log_api",
@@ -506,7 +507,6 @@
":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",
diff --git a/sdk/android/src/jni/pc/media.cc b/sdk/android/src/jni/pc/media.cc
index 4705cf2..55ebae4 100644
--- a/sdk/android/src/jni/pc/media.cc
+++ b/sdk/android/src/jni/pc/media.cc
@@ -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"
