Remove redundant webrtc:: prefixes in api
Created by
tools_webrtc/remove_extra_namespace.py --namespace webrtc
and manual adjustments.
This CL was uploaded by git cl split.
R=eshr@webrtc.org
No-IWYU: Refactoring
Bug: webrtc:42232595
Change-Id: Iaa0f996efce0bc521d3558e4e531ef603a9d8890
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/396160
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44914}
diff --git a/api/array_view.h b/api/array_view.h
index 577d6cb..8b68b9d 100644
--- a/api/array_view.h
+++ b/api/array_view.h
@@ -22,7 +22,7 @@
namespace webrtc {
-// tl;dr: webrtc::ArrayView is the same thing as gsl::span from the Guideline
+// tl;dr: ArrayView is the same thing as gsl::span from the Guideline
// Support Library.
//
// Many functions read from or write to arrays. The obvious way to do this is
@@ -37,7 +37,7 @@
// }
//
// This is flexible, since it doesn't matter how the array is stored (C array,
-// std::vector, webrtc::Buffer, ...), but it's error-prone because the caller
+// std::vector, Buffer, ...), but it's error-prone because the caller
// has to correctly specify the array length:
//
// Contains17(arr, std::size(arr)); // C array
@@ -48,11 +48,11 @@
// It's also kind of messy to have two separate arguments for what is
// conceptually a single thing.
//
-// Enter webrtc::ArrayView<T>. It contains a T pointer (to an array it doesn't
+// Enter ArrayView<T>. It contains a T pointer (to an array it doesn't
// own) and a count, and supports the basic things you'd expect, such as
// indexing and iteration. It allows us to write our function like this:
//
-// bool Contains17(webrtc::ArrayView<const int> arr) {
+// bool Contains17(ArrayView<const int> arr) {
// for (auto e : arr) {
// if (e == 17)
// return true;
@@ -65,7 +65,7 @@
//
// Contains17(arr); // C array
// Contains17(arr); // std::vector
-// Contains17(webrtc::ArrayView<int>(arr, size)); // pointer + size
+// Contains17(ArrayView<int>(arr, size)); // pointer + size
// Contains17(nullptr); // nullptr -> empty ArrayView
// ...
//
@@ -239,8 +239,8 @@
// ArrayView<T, N> to ArrayView<T> or ArrayView<const T>,
// std::vector<T> to ArrayView<T> or ArrayView<const T>,
// const std::vector<T> to ArrayView<const T>,
- // webrtc::Buffer to ArrayView<uint8_t> or ArrayView<const uint8_t>, and
- // const webrtc::Buffer to ArrayView<const uint8_t>.
+ // Buffer to ArrayView<uint8_t> or ArrayView<const uint8_t>, and
+ // const Buffer to ArrayView<const uint8_t>.
template <
typename U,
typename std::enable_if<Size == array_view_internal::kArrayViewVarSize &&
diff --git a/api/async_dns_resolver.h b/api/async_dns_resolver.h
index 11fdf67..e10df20 100644
--- a/api/async_dns_resolver.h
+++ b/api/async_dns_resolver.h
@@ -79,14 +79,14 @@
// Creates an AsyncDnsResolver and starts resolving the name. The callback
// will be called when resolution is finished.
// The callback will be called on the sequence that the caller runs on.
- virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface> CreateAndResolve(
+ virtual std::unique_ptr<AsyncDnsResolverInterface> CreateAndResolve(
const SocketAddress& addr,
absl::AnyInvocable<void()> callback) = 0;
// Creates an AsyncDnsResolver and starts resolving the name to an address
// matching the specified family. The callback will be called when resolution
// is finished. The callback will be called on the sequence that the caller
// runs on.
- virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface> CreateAndResolve(
+ virtual std::unique_ptr<AsyncDnsResolverInterface> CreateAndResolve(
const SocketAddress& addr,
int family,
absl::AnyInvocable<void()> callback) = 0;
@@ -94,7 +94,7 @@
// For backwards compatibility, will be deprecated and removed.
// One has to do a separate Start() call on the
// resolver to start name resolution.
- virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface> Create() = 0;
+ virtual std::unique_ptr<AsyncDnsResolverInterface> Create() = 0;
};
} // namespace webrtc
diff --git a/api/candidate.h b/api/candidate.h
index b02d0ac..69e66ef 100644
--- a/api/candidate.h
+++ b/api/candidate.h
@@ -105,7 +105,7 @@
absl::string_view type_name() const;
// Setting the type requires a constant string (e.g.
- // webrtc::LOCAL_PORT_TYPE). The type should really be an enum rather than a
+ // LOCAL_PORT_TYPE). The type should really be an enum rather than a
// string, but until we make that change the lifetime attribute helps us lock
// things down. See also the `Port` class.
void set_type(IceCandidateType type) { type_ = type; }
@@ -159,10 +159,10 @@
// `network_cost` measures the cost/penalty of using this candidate. A network
// cost of 0 indicates this candidate can be used freely. A value of
- // webrtc::kNetworkCostMax indicates it should be used only as the last
+ // kNetworkCostMax indicates it should be used only as the last
// resort.
void set_network_cost(uint16_t network_cost) {
- RTC_DCHECK_LE(network_cost, webrtc::kNetworkCostMax);
+ RTC_DCHECK_LE(network_cost, kNetworkCostMax);
network_cost_ = network_cost;
}
uint16_t network_cost() const { return network_cost_; }
diff --git a/api/create_peerconnection_factory.cc b/api/create_peerconnection_factory.cc
index aec4f12..12434ee 100644
--- a/api/create_peerconnection_factory.cc
+++ b/api/create_peerconnection_factory.cc
@@ -52,7 +52,7 @@
dependencies.env = CreateEnvironment(std::move(field_trials));
if (network_thread) {
- // TODO(bugs.webrtc.org/13145): Add an webrtc::SocketFactory* argument.
+ // TODO(bugs.webrtc.org/13145): Add an SocketFactory* argument.
dependencies.socket_factory = network_thread->socketserver();
}
dependencies.adm = std::move(default_adm);
diff --git a/api/crypto/frame_decryptor_interface.h b/api/crypto/frame_decryptor_interface.h
index 7ed165f..659f166 100644
--- a/api/crypto/frame_decryptor_interface.h
+++ b/api/crypto/frame_decryptor_interface.h
@@ -60,7 +60,7 @@
// kRecoverable should be returned if the failure was due to something other
// than a decryption failure. kFailedToDecrypt should be returned in all other
// cases.
- virtual Result Decrypt(webrtc::MediaType media_type,
+ virtual Result Decrypt(MediaType media_type,
const std::vector<uint32_t>& csrcs,
ArrayView<const uint8_t> additional_data,
ArrayView<const uint8_t> encrypted_frame,
@@ -69,7 +69,7 @@
// Returns the total required length in bytes for the output of the
// decryption. This can be larger than the actual number of bytes you need but
// must never be smaller as it informs the size of the frame buffer.
- virtual size_t GetMaxPlaintextByteSize(webrtc::MediaType media_type,
+ virtual size_t GetMaxPlaintextByteSize(MediaType media_type,
size_t encrypted_frame_size) = 0;
};
diff --git a/api/crypto/frame_encryptor_interface.h b/api/crypto/frame_encryptor_interface.h
index 9a2b170..6a17972 100644
--- a/api/crypto/frame_encryptor_interface.h
+++ b/api/crypto/frame_encryptor_interface.h
@@ -38,7 +38,7 @@
// must set bytes_written to the number of bytes you wrote in the
// encrypted_frame. 0 must be returned if successful all other numbers can be
// selected by the implementer to represent error codes.
- virtual int Encrypt(webrtc::MediaType media_type,
+ virtual int Encrypt(MediaType media_type,
uint32_t ssrc,
ArrayView<const uint8_t> additional_data,
ArrayView<const uint8_t> frame,
@@ -48,7 +48,7 @@
// Returns the total required length in bytes for the output of the
// encryption. This can be larger than the actual number of bytes you need but
// must never be smaller as it informs the size of the encrypted_frame buffer.
- virtual size_t GetMaxCiphertextByteSize(webrtc::MediaType media_type,
+ virtual size_t GetMaxCiphertextByteSize(MediaType media_type,
size_t frame_size) = 0;
};
diff --git a/api/dtls_transport_interface.h b/api/dtls_transport_interface.h
index c444313..79099f4 100644
--- a/api/dtls_transport_interface.h
+++ b/api/dtls_transport_interface.h
@@ -110,7 +110,7 @@
// 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 webrtc::RefCountInterface {
+class DtlsTransportInterface : public RefCountInterface {
public:
// Returns a pointer to the ICE transport that is owned by the DTLS transport.
virtual scoped_refptr<IceTransportInterface> ice_transport() = 0;
diff --git a/api/dtmf_sender_interface.h b/api/dtmf_sender_interface.h
index 16ce665..3b18483 100644
--- a/api/dtmf_sender_interface.h
+++ b/api/dtmf_sender_interface.h
@@ -41,7 +41,7 @@
// 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 webrtc::RefCountInterface {
+class DtmfSenderInterface : public RefCountInterface {
public:
// Provides the spec compliant default 2 second delay for the ',' character.
static const int kDtmfDefaultCommaDelayMs = 2000;
diff --git a/api/ice_transport_factory.h b/api/ice_transport_factory.h
index cdd9db1..56ac4b3 100644
--- a/api/ice_transport_factory.h
+++ b/api/ice_transport_factory.h
@@ -19,7 +19,7 @@
namespace webrtc {
// Static factory for an IceTransport object that can be created
-// without using a webrtc::PeerConnection.
+// without using a PeerConnection.
// The returned object must be accessed and destroyed on the thread that
// created it.
// The PortAllocator must outlive the created IceTransportInterface object.
@@ -29,7 +29,7 @@
PortAllocator* port_allocator);
// Static factory for an IceTransport object that can be created
-// without using a webrtc::PeerConnection.
+// without using a PeerConnection.
// The returned object must be accessed and destroyed on the thread that
// created it.
// `init.port_allocator()` is required and must outlive the created
diff --git a/api/ice_transport_interface.h b/api/ice_transport_interface.h
index d859ed6..937c916 100644
--- a/api/ice_transport_interface.h
+++ b/api/ice_transport_interface.h
@@ -35,7 +35,7 @@
// The returned object can only be safely used on the signalling thread.
// TODO(crbug.com/907849): Add API calls for the functions that have to
// be exposed to clients, and stop allowing access to the
- // webrtc::IceTransportInternal API.
+ // IceTransportInternal API.
virtual IceTransportInternal* internal() = 0;
};
diff --git a/api/jsep.h b/api/jsep.h
index 829f726..2c626bb 100644
--- a/api/jsep.h
+++ b/api/jsep.h
@@ -317,8 +317,7 @@
std::unique_ptr<SessionDescription> description);
// CreateOffer and CreateAnswer callback interface.
-class RTC_EXPORT CreateSessionDescriptionObserver
- : public webrtc::RefCountInterface {
+class RTC_EXPORT CreateSessionDescriptionObserver : public RefCountInterface {
public:
// This callback transfers the ownership of the `desc`.
// TODO(deadbeef): Make this take an std::unique_ptr<> to avoid confusion
@@ -337,8 +336,7 @@
};
// SetLocalDescription and SetRemoteDescription callback interface.
-class RTC_EXPORT SetSessionDescriptionObserver
- : public webrtc::RefCountInterface {
+class RTC_EXPORT SetSessionDescriptionObserver : public RefCountInterface {
public:
virtual void OnSuccess() = 0;
// See description in CreateSessionDescriptionObserver for OnFailure.
diff --git a/api/legacy_stats_types.h b/api/legacy_stats_types.h
index 8d94541..51b9318 100644
--- a/api/legacy_stats_types.h
+++ b/api/legacy_stats_types.h
@@ -244,7 +244,7 @@
kStatsValueNameLocalCandidateRelayProtocol,
};
- class RTC_EXPORT IdBase : public webrtc::RefCountInterface {
+ class RTC_EXPORT IdBase : public RefCountInterface {
public:
~IdBase() override;
StatsType type() const;
@@ -356,7 +356,7 @@
#endif
private:
- webrtc::SequenceChecker thread_checker_{webrtc::SequenceChecker::kDetached};
+ SequenceChecker thread_checker_{SequenceChecker::kDetached};
mutable int ref_count_ RTC_GUARDED_BY(thread_checker_) = 0;
const Type type_;
@@ -474,7 +474,7 @@
private:
Container list_;
- webrtc::SequenceChecker thread_checker_{SequenceChecker::kDetached};
+ SequenceChecker thread_checker_{SequenceChecker::kDetached};
};
} // namespace webrtc
diff --git a/api/make_ref_counted.h b/api/make_ref_counted.h
index b7781e9..48a0535 100644
--- a/api/make_ref_counted.h
+++ b/api/make_ref_counted.h
@@ -56,7 +56,7 @@
// auto p = scoped_refptr<Foo>(new RefCountedObject<Foo>("bar", 123));
//
// If the class does not inherit from RefCountInterface, but does have
-// AddRef/Release methods (so a T* is convertible to webrtc::scoped_refptr),
+// AddRef/Release methods (so a T* is convertible to scoped_refptr),
// this is equivalent to just
//
// auto p = scoped_refptr<Foo>(new Foo("bar", 123));
diff --git a/api/media_stream_interface.h b/api/media_stream_interface.h
index c3b60f7..07fc72d 100644
--- a/api/media_stream_interface.h
+++ b/api/media_stream_interface.h
@@ -56,7 +56,7 @@
// 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 webrtc::RefCountInterface,
+class RTC_EXPORT MediaSourceInterface : public RefCountInterface,
public NotifierInterface {
public:
enum SourceState { kInitializing, kLive, kEnded, kMuted };
@@ -71,7 +71,7 @@
// C++ version of MediaStreamTrack.
// See: https://www.w3.org/TR/mediacapture-streams/#mediastreamtrack
-class RTC_EXPORT MediaStreamTrackInterface : public webrtc::RefCountInterface,
+class RTC_EXPORT MediaStreamTrackInterface : public RefCountInterface,
public NotifierInterface {
public:
enum TrackState {
@@ -106,7 +106,7 @@
// VideoTrackSourceInterface is a reference counted source used for
// VideoTracks. The same source can be used by multiple VideoTracks.
// VideoTrackSourceInterface is designed to be invoked on the signaling thread
-// except for webrtc::VideoSourceInterface<VideoFrame> methods that will be
+// except for VideoSourceInterface<VideoFrame> methods that will be
// invoked on the worker thread via a VideoTrack. A custom implementation of a
// source can inherit AdaptedVideoTrackSource instead of directly implementing
// this interface.
@@ -161,14 +161,14 @@
// The call is expected to happen on the network thread.
// TODO(crbug/1255737): make pure virtual once downstream project adapts.
virtual void ProcessConstraints(
- const webrtc::VideoTrackSourceConstraints& /* constraints */) {}
+ const VideoTrackSourceConstraints& /* constraints */) {}
protected:
~VideoTrackSourceInterface() override = default;
};
// VideoTrackInterface is designed to be invoked on the signaling thread except
-// for webrtc::VideoSourceInterface<VideoFrame> methods that must be invoked
+// for VideoSourceInterface<VideoFrame> methods that must be invoked
// on the worker thread.
// PeerConnectionFactory::CreateVideoTrack can be used for creating a VideoTrack
// that ensures thread safety and that all methods are called on the right
@@ -210,7 +210,7 @@
// In this method, `absolute_capture_timestamp_ms`, when available, is
// supposed to deliver the timestamp when this audio frame was originally
// captured. This timestamp MUST be based on the same clock as
- // webrtc::TimeMillis().
+ // TimeMillis().
virtual void OnData(
const void* audio_data,
int bits_per_sample,
@@ -269,7 +269,7 @@
// Interface of the audio processor used by the audio track to collect
// statistics.
-class AudioProcessorInterface : public webrtc::RefCountInterface {
+class AudioProcessorInterface : public RefCountInterface {
public:
struct AudioProcessorStatistics {
bool typing_noise_detected = false;
@@ -323,7 +323,7 @@
// must be pushed down.
//
// Thus, this interface acts as simply a container for tracks.
-class MediaStreamInterface : public webrtc::RefCountInterface,
+class MediaStreamInterface : public RefCountInterface,
public NotifierInterface {
public:
virtual std::string id() const = 0;
diff --git a/api/media_types.h b/api/media_types.h
index c2d4149..e58abfd 100644
--- a/api/media_types.h
+++ b/api/media_types.h
@@ -24,7 +24,7 @@
DATA,
UNSUPPORTED,
ANY,
- // Backwards compatibility values for webrtc::MediaType users
+ // Backwards compatibility values for MediaType users
// TODO: https://issues.webrtc.org/42222911 - remove
MEDIA_TYPE_AUDIO [[deprecated("Use AUDIO")]] = AUDIO,
MEDIA_TYPE_VIDEO [[deprecated("Use VIDEO")]] = VIDEO,
diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index 9e2e99c..549afb3 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -155,7 +155,7 @@
// IWYU pragma: end_keep
// MediaStream container interface.
-class StreamCollectionInterface : public webrtc::RefCountInterface {
+class StreamCollectionInterface : public RefCountInterface {
public:
// TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
virtual size_t count() = 0;
@@ -169,7 +169,7 @@
~StreamCollectionInterface() override = default;
};
-class StatsObserver : public webrtc::RefCountInterface {
+class StatsObserver : public RefCountInterface {
public:
virtual void OnComplete(const StatsReports& reports) = 0;
@@ -184,7 +184,7 @@
kUnifiedPlan,
};
-class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
+class RTC_EXPORT PeerConnectionInterface : public RefCountInterface {
public:
// See https://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
enum SignalingState {
@@ -613,7 +613,7 @@
// With this class one can modify outgoing TURN messages.
// The object passed in must remain valid until PeerConnection::Close() is
// called.
- webrtc::TurnCustomizer* turn_customizer = nullptr;
+ TurnCustomizer* turn_customizer = nullptr;
// Preferred network interface.
// A candidate pair on a preferred network has a higher precedence in ICE
@@ -864,13 +864,13 @@
const RtpTransceiverInit& init) = 0;
// Adds a transceiver with the given kind. Can either be
- // webrtc::MediaType::AUDIO or webrtc::MediaType::VIDEO. Errors:
- // - INVALID_PARAMETER: `media_type` is not webrtc::MediaType::AUDIO or
- // webrtc::MediaType::VIDEO.
+ // MediaType::AUDIO or MediaType::VIDEO. Errors:
+ // - INVALID_PARAMETER: `media_type` is not MediaType::AUDIO or
+ // MediaType::VIDEO.
virtual RTCErrorOr<scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
- webrtc::MediaType media_type) = 0;
+ MediaType media_type) = 0;
virtual RTCErrorOr<scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
- webrtc::MediaType media_type,
+ MediaType media_type,
const RtpTransceiverInit& init) = 0;
// Creates a sender without a track. Can be used for "early media"/"warmup"
@@ -941,7 +941,7 @@
// version of getStats() in JavaScript. Implementation status is described in
// api/stats/rtcstats_objects.h. For more details on stats, see spec:
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-getstats
- // TODO(hbos): Takes shared ownership, use webrtc::scoped_refptr<> instead.
+ // TODO(hbos): Takes shared ownership, use scoped_refptr<> instead.
// This requires stop overriding the current version in third party or making
// third party calls explicit to avoid ambiguity during switch. Make the
// future version abstract as soon as third party projects implement it.
@@ -1128,7 +1128,7 @@
// networks come and go. Note that the candidates' transport_name must be set
// to the MID of the m= section that generated the candidate.
// TODO(bugs.webrtc.org/8395): Use IceCandidate instead of
- // webrtc::Candidate, which would avoid the transport_name oddity.
+ // Candidate, which would avoid the transport_name oddity.
virtual bool RemoveIceCandidates(
const std::vector<Candidate>& candidates) = 0;
@@ -1390,13 +1390,11 @@
// PacketSocketFactory when creating the PeerConnectionFactory.
std::unique_ptr<PortAllocator> allocator;
// Factory for creating resolvers that look up hostnames in DNS
- std::unique_ptr<webrtc::AsyncDnsResolverFactoryInterface>
- async_dns_resolver_factory;
- std::unique_ptr<webrtc::IceTransportFactory> ice_transport_factory;
+ std::unique_ptr<AsyncDnsResolverFactoryInterface> async_dns_resolver_factory;
+ std::unique_ptr<IceTransportFactory> ice_transport_factory;
std::unique_ptr<RTCCertificateGeneratorInterface> cert_generator;
std::unique_ptr<SSLCertificateVerifier> tls_cert_verifier;
- std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
- video_bitrate_allocator_factory;
+ std::unique_ptr<VideoBitrateAllocatorFactory> video_bitrate_allocator_factory;
// Optional network controller factory to use.
// Overrides that set in PeerConnectionFactoryDependencies.
std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
@@ -1497,8 +1495,7 @@
// of networking classes, it should use the alternate
// CreatePeerConnectionFactory method which accepts threads as input, and use
// the CreatePeerConnection version that takes a PortAllocator as an argument.
-class RTC_EXPORT PeerConnectionFactoryInterface
- : public webrtc::RefCountInterface {
+class RTC_EXPORT PeerConnectionFactoryInterface : public RefCountInterface {
public:
class Options {
public:
@@ -1541,16 +1538,14 @@
PeerConnectionDependencies dependencies) = 0;
// Returns the capabilities of an RTP sender of type `kind`.
- // If for some reason you pass in webrtc::MediaType::DATA, returns an empty
+ // If for some reason you pass in MediaType::DATA, returns an empty
// structure.
- virtual RtpCapabilities GetRtpSenderCapabilities(
- webrtc::MediaType kind) const = 0;
+ virtual RtpCapabilities GetRtpSenderCapabilities(MediaType kind) const = 0;
// Returns the capabilities of an RTP receiver of type `kind`.
- // If for some reason you pass in webrtc::MediaType::DATA, returns an empty
+ // If for some reason you pass in MediaType::DATA, returns an empty
// structure.
- virtual RtpCapabilities GetRtpReceiverCapabilities(
- webrtc::MediaType kind) const = 0;
+ virtual RtpCapabilities GetRtpReceiverCapabilities(MediaType kind) const = 0;
virtual scoped_refptr<MediaStreamInterface> CreateLocalMediaStream(
const std::string& stream_id) = 0;
@@ -1616,7 +1611,7 @@
// If `network_thread` or `worker_thread` are null, the PeerConnectionFactory
// will create the necessary thread internally. If `signaling_thread` is null,
// the PeerConnectionFactory will use the thread on which this method is called
-// as the signaling thread, wrapping it in an webrtc::Thread object if needed.
+// as the signaling thread, wrapping it in an Thread object if needed.
RTC_EXPORT scoped_refptr<PeerConnectionFactoryInterface>
CreateModularPeerConnectionFactory(
PeerConnectionFactoryDependencies dependencies);
diff --git a/api/priority.h b/api/priority.h
index bb2f881..e74e7d3 100644
--- a/api/priority.h
+++ b/api/priority.h
@@ -27,7 +27,7 @@
};
class RTC_EXPORT PriorityValue
- : public webrtc::StrongAlias<class PriorityValueTag, uint16_t> {
+ : public StrongAlias<class PriorityValueTag, uint16_t> {
public:
explicit PriorityValue(Priority priority);
explicit PriorityValue(uint16_t priority) : StrongAlias(priority) {}
diff --git a/api/ref_count.h b/api/ref_count.h
index b3fb763..0b9d56c 100644
--- a/api/ref_count.h
+++ b/api/ref_count.h
@@ -22,7 +22,7 @@
// you're borrowing someone else's reference. (A newly created object is a
// special case: the reference count is zero on construction, and the code that
// creates the object should immediately call AddRef(), bringing the reference
-// count from zero to one, e.g., by constructing an webrtc::scoped_refptr).
+// count from zero to one, e.g., by constructing an scoped_refptr).
//
// AddRef() creates a new reference to the object.
//
@@ -41,8 +41,8 @@
// result of another thread calling Release().
//
// Calling AddRef() and Release() manually is discouraged. It's recommended to
-// use webrtc::scoped_refptr to manage all pointers to reference counted
-// objects. Note that webrtc::scoped_refptr depends on compile-time duck-typing;
+// use scoped_refptr to manage all pointers to reference counted
+// objects. Note that scoped_refptr depends on compile-time duck-typing;
// formally implementing the below RefCountInterface is not required.
enum class RefCountReleaseStatus { kDroppedLastRef, kOtherRefsRemained };
diff --git a/api/ref_counted_base.h b/api/ref_counted_base.h
index 8862ebd..a411257 100644
--- a/api/ref_counted_base.h
+++ b/api/ref_counted_base.h
@@ -41,7 +41,7 @@
virtual ~RefCountedBase() = default;
private:
- mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
+ mutable webrtc_impl::RefCounter ref_count_{0};
};
// Template based version of `RefCountedBase` for simple implementations that do
@@ -49,11 +49,11 @@
// vtable.
//
// To use:
-// struct MyInt : public webrtc::RefCountedNonVirtual<MyInt> {
+// struct MyInt : public RefCountedNonVirtual<MyInt> {
// int foo_ = 0;
// };
//
-// webrtc::scoped_refptr<MyInt> my_int(new MyInt());
+// scoped_refptr<MyInt> my_int(new MyInt());
//
// sizeof(MyInt) on a 32 bit system would then be 8, int + refcount and no
// vtable generated.
@@ -73,7 +73,7 @@
// so the virtual attribute(s) can be removed.
// 2) The virtual methods are a part of the design of the class. In this
// case you can consider using `RefCountedBase` instead or alternatively
- // use `webrtc::RefCountedObject`.
+ // use `RefCountedObject`.
static_assert(!std::is_polymorphic<T>::value,
"T has virtual methods. RefCountedBase is a better fit.");
const auto status = ref_count_.DecRef();
@@ -91,7 +91,7 @@
~RefCountedNonVirtual() = default;
private:
- mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
+ mutable webrtc_impl::RefCounter ref_count_{0};
};
} // namespace webrtc
diff --git a/api/rtp_packet_info.h b/api/rtp_packet_info.h
index 8e8236d..4059108 100644
--- a/api/rtp_packet_info.h
+++ b/api/rtp_packet_info.h
@@ -87,7 +87,7 @@
std::vector<uint32_t> csrcs_;
uint32_t rtp_timestamp_;
- // Local `webrtc::Clock`-based timestamp of when the packet was received.
+ // Local `Clock`-based timestamp of when the packet was received.
Timestamp receive_time_;
// Fields from the Audio Level header extension:
diff --git a/api/rtp_parameters.cc b/api/rtp_parameters.cc
index c6560d4..06c5811 100644
--- a/api/rtp_parameters.cc
+++ b/api/rtp_parameters.cc
@@ -202,7 +202,7 @@
// srtpfilter. External auth is used by Chromium and replaces the
// extension header value of "kAbsSendTimeUri", so it must not be
// encrypted (which can't be done by Chromium).
- uri != webrtc::RtpExtension::kAbsSendTimeUri &&
+ uri != RtpExtension::kAbsSendTimeUri &&
#endif
uri != RtpExtension::kEncryptHeaderExtensionsUri;
}
diff --git a/api/rtp_parameters.h b/api/rtp_parameters.h
index 3bb65d8..5d7c266 100644
--- a/api/rtp_parameters.h
+++ b/api/rtp_parameters.h
@@ -137,7 +137,7 @@
std::string name;
// The media type of this codec. Equivalent to MIME top-level type.
- webrtc::MediaType kind = webrtc::MediaType::AUDIO;
+ MediaType kind = MediaType::AUDIO;
// If unset, the implementation default is used.
std::optional<int> clock_rate;
@@ -195,7 +195,7 @@
template <typename Sink>
friend void AbslStringify(Sink& sink, const RtpCodecCapability& cap) {
- if (cap.kind == webrtc::MediaType::AUDIO) {
+ if (cap.kind == MediaType::AUDIO) {
absl::Format(&sink, "[audio/%s/%d/%d]", cap.name,
cap.clock_rate.value_or(0), cap.num_channels.value_or(1));
} else {
@@ -214,7 +214,7 @@
//
// Note that ORTC includes a "kind" field, but we omit this because it's
// redundant; if you call
-// "RtpReceiver::GetCapabilities(webrtc::MediaType::AUDIO)", you know you're
+// "RtpReceiver::GetCapabilities(MediaType::AUDIO)", you know you're
// getting audio capabilities.
struct RTC_EXPORT RtpHeaderExtensionCapability {
// URI of this extension, as defined in RFC8285.
@@ -394,7 +394,7 @@
static constexpr char kRepairedRidUri[] =
"urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id";
- // Header extension to propagate webrtc::VideoFrame id field
+ // Header extension to propagate VideoFrame id field
static constexpr char kVideoFrameTrackingIdUri[] =
"http://www.webrtc.org/experiments/rtp-hdrext/video-frame-tracking-id";
diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
index 29df678..bf87f46 100644
--- a/api/rtp_receiver_interface.h
+++ b/api/rtp_receiver_interface.h
@@ -40,13 +40,13 @@
// In the future, it's likely that an RtpReceiver will only call
// OnFirstPacketReceived when a packet is received specifically for its
// SSRC/mid.
- virtual void OnFirstPacketReceived(webrtc::MediaType media_type) = 0;
+ virtual void OnFirstPacketReceived(MediaType media_type) = 0;
protected:
virtual ~RtpReceiverObserverInterface() {}
};
-class RTC_EXPORT RtpReceiverInterface : public webrtc::RefCountInterface,
+class RTC_EXPORT RtpReceiverInterface : public RefCountInterface,
public FrameTransformerHost {
public:
virtual scoped_refptr<MediaStreamTrackInterface> track() const = 0;
@@ -68,7 +68,7 @@
virtual std::vector<scoped_refptr<MediaStreamInterface>> streams() const;
// Audio or video receiver?
- virtual webrtc::MediaType media_type() const = 0;
+ virtual MediaType media_type() const = 0;
// Not to be confused with "mid", this is a field we can temporarily use
// to uniquely identify a receiver until we implement Unified Plan SDP.
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index 478e15d..2a7907d 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -40,7 +40,7 @@
public:
// The observer is called when the first media packet is sent for the observed
// sender. It is called immediately if the first packet was already sent.
- virtual void OnFirstPacketSent(webrtc::MediaType media_type) = 0;
+ virtual void OnFirstPacketSent(MediaType media_type) = 0;
protected:
virtual ~RtpSenderObserverInterface() {}
@@ -48,7 +48,7 @@
using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>;
-class RTC_EXPORT RtpSenderInterface : public webrtc::RefCountInterface,
+class RTC_EXPORT RtpSenderInterface : public RefCountInterface,
public FrameTransformerHost {
public:
// Returns true if successful in setting the track.
@@ -68,7 +68,7 @@
virtual uint32_t ssrc() const = 0;
// Audio or video sender?
- virtual webrtc::MediaType media_type() const = 0;
+ virtual MediaType media_type() const = 0;
// Not to be confused with "mid", this is a field we can temporarily use
// to uniquely identify a receiver until we implement Unified Plan SDP.
diff --git a/api/rtp_transceiver_interface.h b/api/rtp_transceiver_interface.h
index e35c5b7..0d9671f 100644
--- a/api/rtp_transceiver_interface.h
+++ b/api/rtp_transceiver_interface.h
@@ -59,11 +59,11 @@
//
// WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
-class RTC_EXPORT RtpTransceiverInterface : public webrtc::RefCountInterface {
+class RTC_EXPORT RtpTransceiverInterface : public RefCountInterface {
public:
// Media type of the transceiver. Any sender(s)/receiver(s) will have this
// type as well.
- virtual webrtc::MediaType media_type() const = 0;
+ virtual MediaType media_type() const = 0;
// The mid attribute is the mid negotiated and present in the local and
// remote descriptions. Before negotiation is complete, the mid value may be
@@ -165,7 +165,7 @@
// The SetHeaderExtensionsToNegotiate method modifies the next SDP negotiation
// so that it negotiates use of header extensions which are not kStopped.
// https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface
- virtual webrtc::RTCError SetHeaderExtensionsToNegotiate(
+ virtual RTCError SetHeaderExtensionsToNegotiate(
ArrayView<const RtpHeaderExtensionCapability> header_extensions) = 0;
protected:
diff --git a/api/sctp_transport_interface.h b/api/sctp_transport_interface.h
index a9fd115..ba68ca0 100644
--- a/api/sctp_transport_interface.h
+++ b/api/sctp_transport_interface.h
@@ -75,7 +75,7 @@
// 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 webrtc::RefCountInterface {
+class SctpTransportInterface : public RefCountInterface {
public:
// This function can be called from other threads.
virtual scoped_refptr<DtlsTransportInterface> dtls_transport() const = 0;
diff --git a/api/set_local_description_observer_interface.h b/api/set_local_description_observer_interface.h
index 40f85b9..3ba19f9 100644
--- a/api/set_local_description_observer_interface.h
+++ b/api/set_local_description_observer_interface.h
@@ -19,7 +19,7 @@
// OnSetLocalDescriptionComplete() invokes as soon as
// PeerConnectionInterface::SetLocalDescription() operation completes, allowing
// the observer to examine the effects of the operation without delay.
-class SetLocalDescriptionObserverInterface : public webrtc::RefCountInterface {
+class SetLocalDescriptionObserverInterface : public RefCountInterface {
public:
// On success, `error.ok()` is true.
virtual void OnSetLocalDescriptionComplete(RTCError error) = 0;
diff --git a/api/set_remote_description_observer_interface.h b/api/set_remote_description_observer_interface.h
index c162541..4e77be0 100644
--- a/api/set_remote_description_observer_interface.h
+++ b/api/set_remote_description_observer_interface.h
@@ -20,7 +20,7 @@
// 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 webrtc::RefCountInterface {
+class SetRemoteDescriptionObserverInterface : public RefCountInterface {
public:
// On success, `error.ok()` is true.
virtual void OnSetRemoteDescriptionComplete(RTCError error) = 0;
diff --git a/api/task_queue/pending_task_safety_flag.h b/api/task_queue/pending_task_safety_flag.h
index 71e8710..e514b41 100644
--- a/api/task_queue/pending_task_safety_flag.h
+++ b/api/task_queue/pending_task_safety_flag.h
@@ -40,7 +40,7 @@
//
// class ExampleClass {
// ....
-// webrtc::scoped_refptr<PendingTaskSafetyFlag> flag = safety_flag_;
+// scoped_refptr<PendingTaskSafetyFlag> flag = safety_flag_;
// my_task_queue_->PostTask(
// [flag = std::move(flag), this] {
// // Now running on the main thread.
diff --git a/api/task_queue/task_queue_test.cc b/api/task_queue/task_queue_test.cc
index a7c6226..b0054de 100644
--- a/api/task_queue/task_queue_test.cc
+++ b/api/task_queue/task_queue_test.cc
@@ -54,7 +54,7 @@
auto queue = CreateTaskQueue(factory, "PostAndCheckCurrent");
// We're not running a task, so `queue` shouldn't be current.
- // Note that because webrtc::Thread also supports the TQ interface and
+ // Note that because Thread also supports the TQ interface and
// TestMainImpl::Init wraps the main test thread (bugs.webrtc.org/9714), that
// means that TaskQueueBase::Current() will still return a valid value.
EXPECT_FALSE(queue->IsCurrent());
diff --git a/api/task_queue/task_queue_test.h b/api/task_queue/task_queue_test.h
index 214f950..6237f7f 100644
--- a/api/task_queue/task_queue_test.h
+++ b/api/task_queue/task_queue_test.h
@@ -27,7 +27,7 @@
// using ::testing::Values;
// using ::webrtc::TaskQueueTest;
//
-// std::unique_ptr<webrtc::TaskQueueFactory> CreateMyFactory();
+// std::unique_ptr<TaskQueueFactory> CreateMyFactory();
//
// INSTANTIATE_TEST_SUITE_P(My, TaskQueueTest, Values(CreateMyFactory));
//
diff --git a/api/voip/test/mock_voip_engine.h b/api/voip/test/mock_voip_engine.h
index 6f9e281..97006a5 100644
--- a/api/voip/test/mock_voip_engine.h
+++ b/api/voip/test/mock_voip_engine.h
@@ -69,13 +69,11 @@
public:
MOCK_METHOD(VoipResult,
ReceivedRTPPacket,
- (ChannelId channel_id,
- webrtc::ArrayView<const uint8_t> rtp_packet),
+ (ChannelId channel_id, ArrayView<const uint8_t> rtp_packet),
(override));
MOCK_METHOD(VoipResult,
ReceivedRTCPPacket,
- (ChannelId channel_id,
- webrtc::ArrayView<const uint8_t> rtcp_packet),
+ (ChannelId channel_id, ArrayView<const uint8_t> rtcp_packet),
(override));
};
diff --git a/api/voip/voip_base.h b/api/voip/voip_base.h
index f7f4432..72cd4b7 100644
--- a/api/voip/voip_base.h
+++ b/api/voip/voip_base.h
@@ -58,7 +58,7 @@
// Creates a channel.
// Each channel handle maps into one audio media session where each has
// its own separate module for send/receive rtp packet with one peer.
- // Caller must set `transport`, webrtc::Transport callback pointer to
+ // Caller must set `transport`, Transport callback pointer to
// receive rtp/rtcp packets from corresponding media session in VoIP engine.
// VoipEngine framework expects applications to handle network I/O directly
// and injection for incoming RTP from remote endpoint is handled via
