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Break out separate compile targets for various classes

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This reduces the degree of interdependency among modules related
to the PeerConnection class, and makes it easier to isolate inappropriate
external dependencies.

Bug: webrtc:11967
Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32235}
This commit is contained in:
Harald Alvestrand
2020-09-29 14:21:47 +00:00
committed by Commit Bot
parent 8036cb791f
commit 445e6b034a
8 changed files with 177 additions and 22 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
examples/BUILD.gn
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@ -672,6 +672,7 @@ if (is_linux || is_chromeos || is_win) {
"../api/video_codecs:video_codecs_api",
"../media:rtc_media_base",
"../p2p:rtc_p2p",
"../pc:video_track_source",
"../rtc_base:checks",
"../rtc_base/third_party/sigslot",
"../system_wrappers:field_trial",
@ -822,6 +823,7 @@ if (is_win || is_android) {
"../modules/video_capture:video_capture_module",
"../pc:libjingle_peerconnection",
"../pc:peerconnection",
"../pc:video_track_source",
"../rtc_base",
"../test:platform_video_capturer",
"../test:video_test_common",

131
pc/BUILD.gn
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@ -166,8 +166,6 @@ rtc_library("peerconnection") {
"data_channel_controller.h",
"data_channel_utils.cc",
"data_channel_utils.h",
"dtmf_sender.cc",
"dtmf_sender.h",
"ice_server_parsing.cc",
"ice_server_parsing.h",
"jitter_buffer_delay.cc",
@ -178,8 +176,6 @@ rtc_library("peerconnection") {
"jsep_session_description.cc",
"local_audio_source.cc",
"local_audio_source.h",
"media_stream.cc",
"media_stream.h",
"media_stream_observer.cc",
"media_stream_observer.h",
"media_stream_track.h",
@ -198,10 +194,6 @@ rtc_library("peerconnection") {
"rtp_data_channel.h",
"rtp_parameters_conversion.cc",
"rtp_parameters_conversion.h",
"rtp_receiver.cc",
"rtp_receiver.h",
"rtp_sender.cc",
"rtp_sender.h",
"rtp_transceiver.cc",
"rtp_transceiver.h",
"sctp_data_channel.cc",
@ -225,8 +217,6 @@ rtc_library("peerconnection") {
"video_rtp_track_source.h",
"video_track.cc",
"video_track.h",
"video_track_source.cc",
"video_track_source.h",
"webrtc_sdp.cc",
"webrtc_sdp.h",
"webrtc_session_description_factory.cc",
@ -234,7 +224,13 @@ rtc_library("peerconnection") {
]
deps = [
":dtmf_sender",
":media_stream",
":rtc_pc_base",
":rtp_receiver",
":rtp_sender",
":stats_collector_interface",
":video_track_source",
"../api:array_view",
"../api:audio_options_api",
"../api:call_api",
@ -294,6 +290,113 @@ rtc_library("peerconnection") {
]
}
rtc_library("rtp_receiver") {
sources = [
"rtp_receiver.cc",
"rtp_receiver.h",
]
deps = [
":media_stream",
":video_track_source",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api/crypto:frame_decryptor_interface",
"../api/video:video_frame",
"../media:rtc_media_base",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("rtp_sender") {
sources = [
"rtp_sender.cc",
"rtp_sender.h",
]
deps = [
":dtmf_sender",
":stats_collector_interface",
"../api:audio_options_api",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../media:rtc_media_base",
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base/synchronization:mutex",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("dtmf_sender") {
sources = [
"dtmf_sender.cc",
"dtmf_sender.h",
]
deps = [
"../api:libjingle_peerconnection_api",
"../rtc_base:checks",
"../rtc_base:rtc_base",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("media_stream") {
sources = [
"media_stream.cc",
"media_stream.h",
]
deps = [
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:scoped_refptr",
"../rtc_base:checks",
"../rtc_base:refcount",
"../rtc_base:rtc_base",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("video_track_source") {
sources = [
"video_track_source.cc",
"video_track_source.h",
]
deps = [
"../api:media_stream_interface",
"../api/video:video_frame",
"../media:rtc_media_base",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base/system:rtc_export",
]
}
rtc_source_set("stats_collector_interface") {
sources = [ "stats_collector_interface.h" ]
deps = [ "../api:media_stream_interface" ]
}
rtc_source_set("libjingle_peerconnection") {
visibility = [ "*" ]
deps = [
@ -472,6 +575,9 @@ if (rtc_include_tests) {
":libjingle_peerconnection",
":peerconnection",
":rtc_pc_base",
":rtp_receiver",
":rtp_sender",
":video_track_source",
"../api:audio_options_api",
"../api:create_frame_generator",
"../api:create_peerconnection_factory",
@ -569,8 +675,13 @@ if (rtc_include_tests) {
}
deps = [
":dtmf_sender",
":media_stream",
":peerconnection",
":rtc_pc_base",
":rtp_receiver",
":rtp_sender",
":video_track_source",
"../api:array_view",
"../api:audio_options_api",
"../api:create_peerconnection_factory",

7
pc/rtp_sender.cc
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@ -17,8 +17,7 @@
#include "api/audio_options.h"
#include "api/media_stream_interface.h"
#include "media/base/media_engine.h"
#include "pc/peer_connection.h"
#include "pc/stats_collector.h"
#include "pc/stats_collector_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/helpers.h"
#include "rtc_base/location.h"
@ -418,7 +417,7 @@ void LocalAudioSinkAdapter::SetSink(cricket::AudioSource::Sink* sink) {
rtc::scoped_refptr<AudioRtpSender> AudioRtpSender::Create(
rtc::Thread* worker_thread,
const std::string& id,
StatsCollector* stats,
StatsCollectorInterface* stats,
SetStreamsObserver* set_streams_observer) {
return rtc::scoped_refptr<AudioRtpSender>(
new rtc::RefCountedObject<AudioRtpSender>(worker_thread, id, stats,
@ -427,7 +426,7 @@ rtc::scoped_refptr<AudioRtpSender> AudioRtpSender::Create(
AudioRtpSender::AudioRtpSender(rtc::Thread* worker_thread,
const std::string& id,
StatsCollector* stats,
StatsCollectorInterface* stats,
SetStreamsObserver* set_streams_observer)
: RtpSenderBase(worker_thread, id, set_streams_observer),
stats_(stats),

8
pc/rtp_sender.h
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@ -28,7 +28,7 @@
namespace webrtc {
class StatsCollector;
class StatsCollectorInterface;
bool UnimplementedRtpParameterHasValue(const RtpParameters& parameters);
@ -257,7 +257,7 @@ class AudioRtpSender : public DtmfProviderInterface, public RtpSenderBase {
static rtc::scoped_refptr<AudioRtpSender> Create(
rtc::Thread* worker_thread,
const std::string& id,
StatsCollector* stats,
StatsCollectorInterface* stats,
SetStreamsObserver* set_streams_observer);
virtual ~AudioRtpSender();
@ -281,7 +281,7 @@ class AudioRtpSender : public DtmfProviderInterface, public RtpSenderBase {
protected:
AudioRtpSender(rtc::Thread* worker_thread,
const std::string& id,
StatsCollector* stats,
StatsCollectorInterface* stats,
SetStreamsObserver* set_streams_observer);
void SetSend() override;
@ -303,7 +303,7 @@ class AudioRtpSender : public DtmfProviderInterface, public RtpSenderBase {
}
sigslot::signal0<> SignalDestroyed;
StatsCollector* stats_ = nullptr;
StatsCollectorInterface* stats_ = nullptr;
rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender_proxy_;
bool cached_track_enabled_ = false;

1
pc/stats_collector.cc
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@ -16,7 +16,6 @@
#include <vector>
#include "pc/channel.h"
#include "pc/peer_connection.h"
#include "rtc_base/checks.h"
#include "rtc_base/third_party/base64/base64.h"
#include "system_wrappers/include/field_trial.h"

9
pc/stats_collector.h
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@ -27,6 +27,7 @@
#include "api/stats_types.h"
#include "p2p/base/port.h"
#include "pc/peer_connection_internal.h"
#include "pc/stats_collector_interface.h"
#include "rtc_base/network_constants.h"
#include "rtc_base/ssl_certificate.h"
@ -44,7 +45,7 @@ const char* AdapterTypeToStatsType(rtc::AdapterType type);
// A mapping between track ids and their StatsReport.
typedef std::map<std::string, StatsReport*> TrackIdMap;
class StatsCollector {
class StatsCollector : public StatsCollectorInterface {
public:
// The caller is responsible for ensuring that the pc outlives the
// StatsCollector instance.
@ -57,11 +58,13 @@ class StatsCollector {
void AddTrack(MediaStreamTrackInterface* track);
// Adds a local audio track that is used for getting some voice statistics.
void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc);
void AddLocalAudioTrack(AudioTrackInterface* audio_track,
uint32_t ssrc) override;
// Removes a local audio tracks that is used for getting some voice
// statistics.
void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc);
void RemoveLocalAudioTrack(AudioTrackInterface* audio_track,
uint32_t ssrc) override;
// Gather statistics from the session and store them for future use.
void UpdateStats(PeerConnectionInterface::StatsOutputLevel level);

40
pc/stats_collector_interface.h Normal file
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@ -0,0 +1,40 @@
/*
* Copyright 2020 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.
*/
// This file contains an interface for the (obsolete) StatsCollector class that
// is used by compilation units that do not wish to depend on the StatsCollector
// implementation.
#ifndef PC_STATS_COLLECTOR_INTERFACE_H_
#define PC_STATS_COLLECTOR_INTERFACE_H_
#include <stdint.h>
#include "api/media_stream_interface.h"
namespace webrtc {
class StatsCollectorInterface {
public:
virtual ~StatsCollectorInterface() {}
// Adds a local audio track that is used for getting some voice statistics.
virtual void AddLocalAudioTrack(AudioTrackInterface* audio_track,
uint32_t ssrc) = 0;
// Removes a local audio tracks that is used for getting some voice
// statistics.
virtual void RemoveLocalAudioTrack(AudioTrackInterface* audio_track,
uint32_t ssrc) = 0;
};
} // namespace webrtc
#endif // PC_STATS_COLLECTOR_INTERFACE_H_

1
test/pc/e2e/BUILD.gn
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@ -307,6 +307,7 @@ if (!build_with_chromium) {
"../../../api:peer_connection_quality_test_fixture_api",
"../../../api/video:video_frame",
"../../../pc:peerconnection",
"../../../pc:video_track_source",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:variant" ]
}