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Move MediaConfig to its own header file and target.

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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}
This commit is contained in:
Niels Möller 2018-01-23 10:37:42 +01:00 committed by Commit Bot
parent 63e83c77ae
commit 6daa278156
6 changed files with 99 additions and 69 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
api/BUILD.gn
View File

@ -103,6 +103,7 @@ rtc_static_library("libjingle_peerconnection_api") {
# 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",

5
api/peerconnectioninterface.h
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@ -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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
// 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

8
media/BUILD.gn
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@ -55,6 +55,13 @@ rtc_source_set("rtc_h264_profile_id") {
]
}
rtc_source_set("rtc_media_config") {
visibility = [ "*" ]
sources = [
"base/mediaconfig.h",
]
}
rtc_static_library("rtc_media_base") {
visibility = [ "*" ]
defines = []
@ -106,6 +113,7 @@ rtc_static_library("rtc_media_base") {
deps += [
":rtc_h264_profile_id",
":rtc_media_config",
"..:webrtc_common",
"../api:libjingle_peerconnection_api",
"../api:optional",

66
media/base/mediachannel.h
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@ -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 @@ static std::string VectorToString(const std::vector<T>& vals) {
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,

83
media/base/mediaconfig.h Normal file
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@ -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_

5
pc/rtpsender.h
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@ -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"