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Move all api/ files to use webrtc::RefCountInterface

Browse source 
instead of rtc::RefCountInterface

Bug: webrtc:15622
Change-Id: I085660a097a019c7aa58a7e3f0aceeedd9fcc8c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325460
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41050}
This commit is contained in:
Harald Alvestrand 2023-10-31 13:30:30 +00:00 committed by WebRTC LUCI CQ
parent 082cb56ee7
commit e8a2b3c834
17 changed files with 54 additions and 124 deletions
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27
api/BUILD.gn
View file

@ -141,13 +141,14 @@ rtc_library("media_stream_interface") {
deps = [
":audio_options_api",
":make_ref_counted",
":ref_count",
":ref_count",
":rtp_parameters",
":scoped_refptr",
":sequence_checker",
":video_track_source_constraints",
"../modules/audio_processing:audio_processing_statistics",
"../rtc_base:checks",
"../rtc_base:refcount",
"../rtc_base/system:no_unique_address",
"../rtc_base/system:rtc_export",
"video:recordable_encoded_frame",
@ -195,9 +196,9 @@ rtc_source_set("ice_transport_interface") {
deps = [
":async_dns_resolver",
":packet_socket_factory",
":ref_count",
":rtc_error",
":scoped_refptr",
"../rtc_base:refcount",
"rtc_event_log:rtc_event_log",
]
}
@ -211,9 +212,9 @@ rtc_library("dtls_transport_interface") {
]
deps = [
":ice_transport_interface",
":ref_count",
":rtc_error",
":scoped_refptr",
"../rtc_base:refcount",
"../rtc_base:ssl",
"../rtc_base/system:rtc_export",
]
@ -226,7 +227,7 @@ rtc_library("dtmf_sender_interface") {
sources = [ "dtmf_sender_interface.h" ]
deps = [
":media_stream_interface",
"../rtc_base:refcount",
":ref_count",
]
}
@ -242,11 +243,11 @@ rtc_library("rtp_sender_interface") {
":dtmf_sender_interface",
":frame_transformer_interface",
":media_stream_interface",
":ref_count",
":rtc_error",
":rtp_parameters",
":scoped_refptr",
"../rtc_base:checks",
"../rtc_base:refcount",
"../rtc_base/system:rtc_export",
"crypto:frame_encryptor_interface",
"video_codecs:video_codecs_api",
@ -308,6 +309,7 @@ rtc_library("libjingle_peerconnection_api") {
":network_state_predictor_api",
":packet_socket_factory",
":priority",
":ref_count",
":rtc_error",
":rtc_stats_api",
":rtp_packet_info",
@ -324,7 +326,6 @@ rtc_library("libjingle_peerconnection_api") {
"../rtc_base:logging",
"../rtc_base:network",
"../rtc_base:network_constants",
"../rtc_base:refcount",
"../rtc_base:rtc_certificate_generator",
"../rtc_base:ssl",
"../rtc_base:stringutils",
@ -378,8 +379,8 @@ rtc_source_set("frame_transformer_interface") {
sources = [ "frame_transformer_interface.h" ]
deps = [
":make_ref_counted",
":ref_count",
":scoped_refptr",
"../rtc_base:refcount",
"video:encoded_frame",
"video:video_frame_metadata",
]
@ -466,7 +467,10 @@ rtc_source_set("scoped_refptr") {
rtc_source_set("make_ref_counted") {
visibility = [ "*" ]
sources = [ "make_ref_counted.h" ]
deps = [ "../rtc_base:refcount" ]
deps = [
":ref_count",
"../rtc_base:refcount",
]
}
rtc_source_set("video_quality_analyzer_api") {
@ -787,6 +791,7 @@ rtc_source_set("rtc_stats_api") {
deps = [
":make_ref_counted",
":ref_count",
":scoped_refptr",
"../api:refcountedbase",
"../rtc_base:checks",
@ -919,6 +924,7 @@ rtc_source_set("refcountedbase") {
visibility = [ "*" ]
sources = [ "ref_counted_base.h" ]
deps = [
":ref_count",
"../rtc_base:macromagic",
"../rtc_base:refcount",
]
@ -1217,6 +1223,7 @@ if (rtc_include_tests) {
":array_view",
":libjingle_peerconnection_api",
":make_ref_counted",
":ref_count",
":rtp_parameters",
"../rtc_base:checks",
"../rtc_base:refcount",
@ -1270,6 +1277,7 @@ if (rtc_include_tests) {
deps = [
":libjingle_peerconnection_api",
":ref_count",
"../api:scoped_refptr",
"../rtc_base:refcount",
"../test:test_support",
@ -1410,6 +1418,7 @@ if (rtc_include_tests) {
sources = [ "test/mock_video_track.h" ]
deps = [
":ref_count",
"../api:media_stream_interface",
"../api:scoped_refptr",
"../rtc_base:refcount",
@ -1590,10 +1599,10 @@ rtc_library("frame_transformer_factory") {
]
deps = [
":frame_transformer_interface",
":ref_count",
":scoped_refptr",
"../audio:audio",
"../modules/rtp_rtcp",
"../rtc_base:refcount",
"video:encoded_frame",
"video:video_frame_metadata",
]

80
api/DEPS
View file

@ -81,7 +81,6 @@ specific_include_rules = {
"data_channel_interface\.h": [
"+rtc_base/copy_on_write_buffer.h",
"+rtc_base/ref_count.h",
],
"data_channel_transport_interface\.h": [
@ -89,33 +88,15 @@ specific_include_rules = {
],
"dtls_transport_interface\.h": [
"+rtc_base/ref_count.h",
"+rtc_base/ssl_certificate.h",
],
"dtmf_sender_interface\.h": [
"+rtc_base/ref_count.h",
],
"fec_controller\.h": [
"+modules/include/module_fec_types.h",
],
"frame_transformer_interface\.h": [
"+rtc_base/ref_count.h",
],
"ice_transport_interface\.h": [
"+rtc_base/ref_count.h",
],
"jsep\.h": [
"+rtc_base/ref_count.h",
],
"media_stream_interface\.h": [
"+modules/audio_processing/include/audio_processing_statistics.h",
"+rtc_base/ref_count.h",
],
"packet_socket_factory\.h": [
@ -132,7 +113,6 @@ specific_include_rules = {
"+rtc_base/network.h",
"+rtc_base/network_constants.h",
"+rtc_base/network_monitor_factory.h",
"+rtc_base/ref_count.h",
"+rtc_base/rtc_certificate.h",
"+rtc_base/rtc_certificate_generator.h",
"+rtc_base/socket_address.h",
@ -148,7 +128,6 @@ specific_include_rules = {
],
"ref_counted_base\.h": [
"+rtc_base/ref_count.h",
"+rtc_base/ref_counter.h",
],
@ -159,74 +138,19 @@ specific_include_rules = {
# For private member and constructor.
"+rtc_base/system/file_wrapper.h",
],
"rtp_receiver_interface\.h": [
"+rtc_base/ref_count.h",
],
"rtp_sender_interface\.h": [
"+rtc_base/ref_count.h",
],
"rtp_transceiver_interface\.h": [
"+rtc_base/ref_count.h",
],
"sctp_transport_interface\.h": [
"+rtc_base/ref_count.h",
],
"set_local_description_observer_interface\.h": [
"+rtc_base/ref_count.h",
],
"set_remote_description_observer_interface\.h": [
"+rtc_base/ref_count.h",
],
"legacy_stats_types\.h": [
"+rtc_base/ref_count.h",
"+rtc_base/thread_checker.h",
],
"uma_metrics\.h": [
"+rtc_base/ref_count.h",
],
"audio_mixer\.h": [
"+rtc_base/ref_count.h",
],
"audio_decoder\.h": [
"+rtc_base/buffer.h",
],
"audio_decoder_factory\.h": [
"+rtc_base/ref_count.h",
],
"audio_encoder\.h": [
"+rtc_base/buffer.h",
],
"audio_encoder_factory\.h": [
"+rtc_base/ref_count.h",
],
"frame_decryptor_interface\.h": [
"+rtc_base/ref_count.h",
],
"frame_encryptor_interface\.h": [
"+rtc_base/ref_count.h",
],
"rtc_stats_collector_callback\.h": [
"+rtc_base/ref_count.h",
],
"rtc_stats_report\.h": [
"+rtc_base/ref_count.h",
],
"audioproc_float\.h": [
"+modules/audio_processing/include/audio_processing.h",
],
@ -279,10 +203,6 @@ specific_include_rules = {
"+modules/video_coding/include/video_codec_interface.h"
],
"video_encoder_config\.h": [
"+rtc_base/ref_count.h",
],
"sequence_checker\.h": [
"+rtc_base/synchronization/sequence_checker_internal.h",
"+rtc_base/thread_annotations.h",

2
api/data_channel_interface.h
View file

@ -22,10 +22,10 @@
#include "absl/functional/any_invocable.h"
#include "absl/types/optional.h"
#include "api/priority.h"
#include "api/ref_count.h"
#include "api/rtc_error.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {

4
api/dtls_transport_interface.h
View file

@ -16,9 +16,9 @@
#include "absl/types/optional.h"
#include "api/ice_transport_interface.h"
#include "api/ref_count.h"
#include "api/rtc_error.h"
#include "api/scoped_refptr.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/ssl_certificate.h"
#include "rtc_base/system/rtc_export.h"
@ -107,7 +107,7 @@ class DtlsTransportObserverInterface {
// accessed on that thread, except for functions explicitly marked otherwise.
// References can be held by other threads, and destruction can therefore
// be initiated by other threads.
class DtlsTransportInterface : public rtc::RefCountInterface {
class DtlsTransportInterface : public webrtc::RefCountInterface {
public:
// Returns a pointer to the ICE transport that is owned by the DTLS transport.
virtual rtc::scoped_refptr<IceTransportInterface> ice_transport() = 0;

4
api/dtmf_sender_interface.h
View file

@ -14,7 +14,7 @@
#include <string>
#include "api/media_stream_interface.h"
#include "rtc_base/ref_count.h"
#include "api/ref_count.h"
namespace webrtc {
@ -42,7 +42,7 @@ class DtmfSenderObserverInterface {
// The interface of native implementation of the RTCDTMFSender defined by the
// WebRTC W3C Editor's Draft.
// See: https://www.w3.org/TR/webrtc/#peer-to-peer-dtmf
class DtmfSenderInterface : public rtc::RefCountInterface {
class DtmfSenderInterface : public webrtc::RefCountInterface {
public:
// Provides the spec compliant default 2 second delay for the ',' character.
static const int kDtmfDefaultCommaDelayMs = 2000;

2
api/frame_transformer_interface.h
View file

@ -15,10 +15,10 @@
#include <string>
#include <vector>
#include "api/ref_count.h"
#include "api/scoped_refptr.h"
#include "api/video/encoded_frame.h"
#include "api/video/video_frame_metadata.h"
#include "rtc_base/ref_count.h"
namespace webrtc {

4
api/ice_transport_interface.h
View file

@ -15,10 +15,10 @@
#include "api/async_dns_resolver.h"
#include "api/async_resolver_factory.h"
#include "api/ref_count.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/scoped_refptr.h"
#include "rtc_base/ref_count.h"
namespace cricket {
class IceTransportInternal;
@ -33,7 +33,7 @@ class FieldTrialsView;
// An ICE transport, as represented to the outside world.
// This object is refcounted, and is therefore alive until the
// last holder has released it.
class IceTransportInterface : public rtc::RefCountInterface {
class IceTransportInterface : public webrtc::RefCountInterface {
public:
// Accessor for the internal representation of an ICE transport.
// The returned object can only be safely used on the signalling thread.

7
api/jsep.h
View file

@ -27,8 +27,8 @@
#include <vector>
#include "absl/types/optional.h"
#include "api/ref_count.h"
#include "api/rtc_error.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
namespace cricket {
@ -219,7 +219,7 @@ std::unique_ptr<SessionDescriptionInterface> CreateSessionDescription(
// CreateOffer and CreateAnswer callback interface.
class RTC_EXPORT CreateSessionDescriptionObserver
: public rtc::RefCountInterface {
: public webrtc::RefCountInterface {
public:
// This callback transfers the ownership of the `desc`.
// TODO(deadbeef): Make this take an std::unique_ptr<> to avoid confusion
@ -238,7 +238,8 @@ class RTC_EXPORT CreateSessionDescriptionObserver
};
// SetLocalDescription and SetRemoteDescription callback interface.
class RTC_EXPORT SetSessionDescriptionObserver : public rtc::RefCountInterface {
class RTC_EXPORT SetSessionDescriptionObserver
: public webrtc::RefCountInterface {
public:
virtual void OnSuccess() = 0;
// See description in CreateSessionDescriptionObserver for OnFailure.

4
api/legacy_stats_types.h
View file

@ -20,9 +20,9 @@
#include <string>
#include <vector>
#include "api/ref_count.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -240,7 +240,7 @@ class RTC_EXPORT StatsReport {
kStatsValueNameLocalCandidateRelayProtocol,
};
class RTC_EXPORT IdBase : public rtc::RefCountInterface {
class RTC_EXPORT IdBase : public webrtc::RefCountInterface {
public:
~IdBase() override;
StatsType type() const;

10
api/media_stream_interface.h
View file

@ -23,6 +23,7 @@
#include "absl/types/optional.h"
#include "api/audio_options.h"
#include "api/ref_count.h"
#include "api/scoped_refptr.h"
#include "api/video/recordable_encoded_frame.h"
#include "api/video/video_frame.h"
@ -30,7 +31,6 @@
#include "api/video/video_source_interface.h"
#include "api/video_track_source_constraints.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -54,7 +54,7 @@ class NotifierInterface {
// Base class for sources. A MediaStreamTrack has an underlying source that
// provides media. A source can be shared by multiple tracks.
class RTC_EXPORT MediaSourceInterface : public rtc::RefCountInterface,
class RTC_EXPORT MediaSourceInterface : public webrtc::RefCountInterface,
public NotifierInterface {
public:
enum SourceState { kInitializing, kLive, kEnded, kMuted };
@ -69,7 +69,7 @@ class RTC_EXPORT MediaSourceInterface : public rtc::RefCountInterface,
// C++ version of MediaStreamTrack.
// See: https://www.w3.org/TR/mediacapture-streams/#mediastreamtrack
class RTC_EXPORT MediaStreamTrackInterface : public rtc::RefCountInterface,
class RTC_EXPORT MediaStreamTrackInterface : public webrtc::RefCountInterface,
public NotifierInterface {
public:
enum TrackState {
@ -267,7 +267,7 @@ class RTC_EXPORT AudioSourceInterface : public MediaSourceInterface {
// Interface of the audio processor used by the audio track to collect
// statistics.
class AudioProcessorInterface : public rtc::RefCountInterface {
class AudioProcessorInterface : public webrtc::RefCountInterface {
public:
struct AudioProcessorStatistics {
bool typing_noise_detected = false;
@ -321,7 +321,7 @@ typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> > VideoTrackVector;
// must be pushed down.
//
// Thus, this interface acts as simply a container for tracks.
class MediaStreamInterface : public rtc::RefCountInterface,
class MediaStreamInterface : public webrtc::RefCountInterface,
public NotifierInterface {
public:
virtual std::string id() const = 0;

10
api/peer_connection_interface.h
View file

@ -128,11 +128,11 @@
// 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 "api/ref_count.h"
#include "p2p/base/port_allocator.h"
#include "rtc_base/network.h"
#include "rtc_base/network_constants.h"
#include "rtc_base/network_monitor_factory.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/socket_address.h"
@ -151,7 +151,7 @@ namespace webrtc {
class MediaFactory;
// MediaStream container interface.
class StreamCollectionInterface : public rtc::RefCountInterface {
class StreamCollectionInterface : public webrtc::RefCountInterface {
public:
// TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
virtual size_t count() = 0;
@ -165,7 +165,7 @@ class StreamCollectionInterface : public rtc::RefCountInterface {
~StreamCollectionInterface() override = default;
};
class StatsObserver : public rtc::RefCountInterface {
class StatsObserver : public webrtc::RefCountInterface {
public:
virtual void OnComplete(const StatsReports& reports) = 0;
@ -180,7 +180,7 @@ enum class SdpSemantics {
kUnifiedPlan,
};
class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
public:
// See https://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
enum SignalingState {
@ -1487,7 +1487,7 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final {
// CreatePeerConnectionFactory method which accepts threads as input, and use
// the CreatePeerConnection version that takes a PortAllocator as an argument.
class RTC_EXPORT PeerConnectionFactoryInterface
: public rtc::RefCountInterface {
: public webrtc::RefCountInterface {
public:
class Options {
public:

4
api/rtp_receiver_interface.h
View file

@ -22,10 +22,10 @@
#include "api/frame_transformer_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/ref_count.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/transport/rtp/rtp_source.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -44,7 +44,7 @@ class RtpReceiverObserverInterface {
virtual ~RtpReceiverObserverInterface() {}
};
class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface {
class RTC_EXPORT RtpReceiverInterface : public webrtc::RefCountInterface {
public:
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;

4
api/rtp_sender_interface.h
View file

@ -25,18 +25,18 @@
#include "api/frame_transformer_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/ref_count.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>;
class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface {
class RTC_EXPORT RtpSenderInterface : public webrtc::RefCountInterface {
public:
// Returns true if successful in setting the track.
// Fails if an audio track is set on a video RtpSender, or vice-versa.

4
api/rtp_transceiver_interface.h
View file

@ -18,12 +18,12 @@
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/media_types.h"
#include "api/ref_count.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_direction.h"
#include "api/scoped_refptr.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -58,7 +58,7 @@ struct RTC_EXPORT RtpTransceiverInit final {
//
// WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
class RTC_EXPORT RtpTransceiverInterface : public rtc::RefCountInterface {
class RTC_EXPORT RtpTransceiverInterface : public webrtc::RefCountInterface {
public:
// Media type of the transceiver. Any sender(s)/receiver(s) will have this
// type as well.

4
api/sctp_transport_interface.h
View file

@ -13,9 +13,9 @@
#include "absl/types/optional.h"
#include "api/dtls_transport_interface.h"
#include "api/ref_count.h"
#include "api/rtc_error.h"
#include "api/scoped_refptr.h"
#include "rtc_base/ref_count.h"
namespace webrtc {
@ -75,7 +75,7 @@ class SctpTransportObserverInterface {
// accessed on that thread, except for functions explicitly marked otherwise.
// References can be held by other threads, and destruction can therefore
// be initiated by other threads.
class SctpTransportInterface : public rtc::RefCountInterface {
class SctpTransportInterface : public webrtc::RefCountInterface {
public:
// This function can be called from other threads.
virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const = 0;

4
api/set_local_description_observer_interface.h
View file

@ -11,15 +11,15 @@
#ifndef API_SET_LOCAL_DESCRIPTION_OBSERVER_INTERFACE_H_
#define API_SET_LOCAL_DESCRIPTION_OBSERVER_INTERFACE_H_
#include "api/ref_count.h"
#include "api/rtc_error.h"
#include "rtc_base/ref_count.h"
namespace webrtc {
// OnSetLocalDescriptionComplete() invokes as soon as
// PeerConnectionInterface::SetLocalDescription() operation completes, allowing
// the observer to examine the effects of the operation without delay.
class SetLocalDescriptionObserverInterface : public rtc::RefCountInterface {
class SetLocalDescriptionObserverInterface : public webrtc::RefCountInterface {
public:
// On success, `error.ok()` is true.
virtual void OnSetLocalDescriptionComplete(RTCError error) = 0;

4
api/set_remote_description_observer_interface.h
View file

@ -11,8 +11,8 @@
#ifndef API_SET_REMOTE_DESCRIPTION_OBSERVER_INTERFACE_H_
#define API_SET_REMOTE_DESCRIPTION_OBSERVER_INTERFACE_H_
#include "api/ref_count.h"
#include "api/rtc_error.h"
#include "rtc_base/ref_count.h"
namespace webrtc {
@ -20,7 +20,7 @@ namespace webrtc {
// callback is invoked such that the state of the peer connection can be
// examined to accurately reflect the effects of the SetRemoteDescription
// operation.
class SetRemoteDescriptionObserverInterface : public rtc::RefCountInterface {
class SetRemoteDescriptionObserverInterface : public webrtc::RefCountInterface {
public:
// On success, `error.ok()` is true.
virtual void OnSetRemoteDescriptionComplete(RTCError error) = 0;