Move MediaConfig to its own header file and target.
To eliminate circular dependencies, we need to eliminate the include
of media/base/mediachannel.h from api/peerconnectioninterface.h.
MediaConfig is one of the types the PeerConnection api depends on,
since it's part of PeerConnectionInterface::RTCConfiguration. It's
formally a public member, but the intention is that applications should use
accessor mehtods on RTCConfiguration and never access the contents of
MediaConfig directly.
Bug: webrtc:7504
Change-Id: Idfab6f69132d6b90d1628fa4543a393e22db79ac
Reviewed-on: https://webrtc-review.googlesource.com/41260
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21731}diff --git a/api/BUILD.gn b/api/BUILD.gn
index 95891df..e9eba77 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -103,6 +103,7 @@
# file, really. All these should arguably go away in time.
"..:typedefs",
"..:webrtc_common",
+ "../media:rtc_media_config",
"../modules/audio_processing:audio_processing_statistics",
"../rtc_base:checks",
"../rtc_base:deprecation",
diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index 3cbf263..7087de6 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -95,7 +95,7 @@
#include "api/umametrics.h"
#include "call/callfactoryinterface.h"
#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
-#include "media/base/mediachannel.h"
+#include "media/base/mediaconfig.h"
#include "media/base/videocapturer.h"
#include "p2p/base/portallocator.h"
#include "rtc_base/network.h"
@@ -446,6 +446,9 @@
// standard priority order.
bool prioritize_most_likely_ice_candidate_pairs = false;
+ // Implementation defined settings. A public member only for the benefit of
+ // the implementation. Applications must not access it directly, and should
+ // instead use provided accessor methods, e.g., set_cpu_adaptation.
struct cricket::MediaConfig media_config;
// If set to true, only one preferred TURN allocation will be used per
diff --git a/media/BUILD.gn b/media/BUILD.gn
index dfadbae..b856bb5 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -55,6 +55,13 @@
]
}
+rtc_source_set("rtc_media_config") {
+ visibility = [ "*" ]
+ sources = [
+ "base/mediaconfig.h",
+ ]
+}
+
rtc_static_library("rtc_media_base") {
visibility = [ "*" ]
defines = []
@@ -106,6 +113,7 @@
deps += [
":rtc_h264_profile_id",
+ ":rtc_media_config",
"..:webrtc_common",
"../api:libjingle_peerconnection_api",
"../api:optional",
diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index 3bcb596..32ab479 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -28,6 +28,7 @@
#include "api/videosourceinterface.h"
#include "call/video_config.h"
#include "media/base/codec.h"
+#include "media/base/mediaconfig.h"
#include "media/base/mediaconstants.h"
#include "media/base/streamparams.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
@@ -88,71 +89,6 @@
return ost.str();
}
-// Construction-time settings, passed on when creating
-// MediaChannels.
-struct MediaConfig {
- // Set DSCP value on packets. This flag comes from the
- // PeerConnection constraint 'googDscp'.
- bool enable_dscp = false;
-
- // Video-specific config.
- struct Video {
- // Enable WebRTC CPU Overuse Detection. This flag comes from the
- // PeerConnection constraint 'googCpuOveruseDetection'.
- bool enable_cpu_adaptation = true;
-
- // Enable WebRTC suspension of video. No video frames will be sent
- // when the bitrate is below the configured minimum bitrate. This
- // flag comes from the PeerConnection constraint
- // 'googSuspendBelowMinBitrate', and WebRtcVideoChannel copies it
- // to VideoSendStream::Config::suspend_below_min_bitrate.
- bool suspend_below_min_bitrate = false;
-
- // Set to true if the renderer has an algorithm of frame selection.
- // If the value is true, then WebRTC will hand over a frame as soon as
- // possible without delay, and rendering smoothness is completely the duty
- // of the renderer;
- // If the value is false, then WebRTC is responsible to delay frame release
- // in order to increase rendering smoothness.
- //
- // This flag comes from PeerConnection's RtcConfiguration, but is
- // currently only set by the command line flag
- // 'disable-rtc-smoothness-algorithm'.
- // WebRtcVideoChannel::AddRecvStream copies it to the created
- // WebRtcVideoReceiveStream, where it is returned by the
- // SmoothsRenderedFrames method. This method is used by the
- // VideoReceiveStream, where the value is passed on to the
- // IncomingVideoStream constructor.
- bool enable_prerenderer_smoothing = true;
-
- // Enables periodic bandwidth probing in application-limited region.
- bool periodic_alr_bandwidth_probing = false;
-
- // Enables the new method to estimate the cpu load from encoding, used for
- // cpu adaptation. This flag is intended to be controlled primarily by a
- // Chrome origin-trial.
- // TODO(bugs.webrtc.org/8504): If all goes well, the flag will be removed
- // together with the old method of estimation.
- bool experiment_cpu_load_estimator = false;
- } video;
-
- bool operator==(const MediaConfig& o) const {
- return enable_dscp == o.enable_dscp &&
- video.enable_cpu_adaptation ==
- o.video.enable_cpu_adaptation &&
- video.suspend_below_min_bitrate ==
- o.video.suspend_below_min_bitrate &&
- video.enable_prerenderer_smoothing ==
- o.video.enable_prerenderer_smoothing &&
- video.periodic_alr_bandwidth_probing ==
- o.video.periodic_alr_bandwidth_probing &&
- video.experiment_cpu_load_estimator ==
- o.video.experiment_cpu_load_estimator;
- }
-
- bool operator!=(const MediaConfig& o) const { return !(*this == o); }
-};
-
// Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
// Used to be flags, but that makes it hard to selectively apply options.
// We are moving all of the setting of options to structs like this,
diff --git a/media/base/mediaconfig.h b/media/base/mediaconfig.h
new file mode 100644
index 0000000..5e84871
--- /dev/null
+++ b/media/base/mediaconfig.h
@@ -0,0 +1,83 @@
+/*
+ * Copyright (c) 2018 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 MEDIA_BASE_MEDIACONFIG_H_
+#define MEDIA_BASE_MEDIACONFIG_H_
+
+namespace cricket {
+
+// Construction-time settings, passed on when creating
+// MediaChannels.
+struct MediaConfig {
+ // Set DSCP value on packets. This flag comes from the
+ // PeerConnection constraint 'googDscp'.
+ bool enable_dscp = false;
+
+ // Video-specific config.
+ struct Video {
+ // Enable WebRTC CPU Overuse Detection. This flag comes from the
+ // PeerConnection constraint 'googCpuOveruseDetection'.
+ bool enable_cpu_adaptation = true;
+
+ // Enable WebRTC suspension of video. No video frames will be sent
+ // when the bitrate is below the configured minimum bitrate. This
+ // flag comes from the PeerConnection constraint
+ // 'googSuspendBelowMinBitrate', and WebRtcVideoChannel copies it
+ // to VideoSendStream::Config::suspend_below_min_bitrate.
+ bool suspend_below_min_bitrate = false;
+
+ // Set to true if the renderer has an algorithm of frame selection.
+ // If the value is true, then WebRTC will hand over a frame as soon as
+ // possible without delay, and rendering smoothness is completely the duty
+ // of the renderer;
+ // If the value is false, then WebRTC is responsible to delay frame release
+ // in order to increase rendering smoothness.
+ //
+ // This flag comes from PeerConnection's RtcConfiguration, but is
+ // currently only set by the command line flag
+ // 'disable-rtc-smoothness-algorithm'.
+ // WebRtcVideoChannel::AddRecvStream copies it to the created
+ // WebRtcVideoReceiveStream, where it is returned by the
+ // SmoothsRenderedFrames method. This method is used by the
+ // VideoReceiveStream, where the value is passed on to the
+ // IncomingVideoStream constructor.
+ bool enable_prerenderer_smoothing = true;
+
+ // Enables periodic bandwidth probing in application-limited region.
+ bool periodic_alr_bandwidth_probing = false;
+
+ // Enables the new method to estimate the cpu load from encoding, used for
+ // cpu adaptation. This flag is intended to be controlled primarily by a
+ // Chrome origin-trial.
+ // TODO(bugs.webrtc.org/8504): If all goes well, the flag will be removed
+ // together with the old method of estimation.
+ bool experiment_cpu_load_estimator = false;
+ } video;
+
+ bool operator==(const MediaConfig& o) const {
+ return enable_dscp == o.enable_dscp &&
+ video.enable_cpu_adaptation ==
+ o.video.enable_cpu_adaptation &&
+ video.suspend_below_min_bitrate ==
+ o.video.suspend_below_min_bitrate &&
+ video.enable_prerenderer_smoothing ==
+ o.video.enable_prerenderer_smoothing &&
+ video.periodic_alr_bandwidth_probing ==
+ o.video.periodic_alr_bandwidth_probing &&
+ video.experiment_cpu_load_estimator ==
+ o.video.experiment_cpu_load_estimator;
+ }
+
+ bool operator!=(const MediaConfig& o) const { return !(*this == o); }
+};
+
+} // namespace cricket
+
+#endif // MEDIA_BASE_MEDIACONFIG_H_
diff --git a/pc/rtpsender.h b/pc/rtpsender.h
index 119d91f..1fd7b37 100644
--- a/pc/rtpsender.h
+++ b/pc/rtpsender.h
@@ -23,9 +23,8 @@
#include "api/rtpsenderinterface.h"
#include "rtc_base/basictypes.h"
#include "rtc_base/criticalsection.h"
-// Adding 'nogncheck' to disable the gn include headers check to support modular
-// WebRTC build targets.
-#include "media/base/audiosource.h" // nogncheck
+#include "media/base/audiosource.h"
+#include "media/base/mediachannel.h"
#include "pc/dtmfsender.h"
#include "pc/statscollector.h"
