From be74b8058be60096bed9423f218b64a4133e5bb9 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Niels=20M=C3=B6ller?= Date: Fri, 18 Mar 2022 14:10:15 +0100 Subject: [PATCH] Fix spelling of receiver and transceiver. Bug: None Change-Id: I439e217d67283b182833e48da15af9ae367ac14e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256015 Reviewed-by: Harald Alvestrand Commit-Queue: Niels Moller Cr-Commit-Position: refs/heads/main@{#36257} --- media/sctp/usrsctp_transport.h | 2 +- media/sctp/usrsctp_transport_unittest.cc | 2 +- .../codecs/isac/fix/include/isacfix.h | 2 +- .../codecs/isac/fix/source/isacfix.c | 2 +- .../audio_coding/codecs/isac/main/include/isac.h | 2 +- .../audio_coding/codecs/isac/main/source/isac.c | 2 +- modules/audio_device/audio_device_buffer.h | 2 +- ...eceive_side_congestion_controller_unittest.cc | 4 ++-- .../source/rtcp_packet/loss_notification.h | 4 ++-- .../source/rtcp_transceiver_impl_unittest.cc | 2 +- pc/media_session_unittest.cc | 2 +- pc/sdp_offer_answer.h | 2 +- rtc_base/callback_list_unittest.cc | 2 +- rtc_tools/video_replay.cc | 4 ++-- sdk/objc/base/RTCVideoEncoder.h | 2 +- .../components/audio/RTCAudioSession+Private.h | 4 ++-- .../video/default_video_quality_analyzer_test.cc | 2 +- test/scenario/video_stream_unittest.cc | 2 +- video/rtp_video_stream_receiver.cc | 16 ++++++++-------- video/rtp_video_stream_receiver2.cc | 16 ++++++++-------- 20 files changed, 38 insertions(+), 38 deletions(-) diff --git a/media/sctp/usrsctp_transport.h b/media/sctp/usrsctp_transport.h index 2dd6abf9c5..7c7ce8c4a8 100644 --- a/media/sctp/usrsctp_transport.h +++ b/media/sctp/usrsctp_transport.h @@ -63,7 +63,7 @@ struct SctpInboundPacket; // 11. SctpTransport::OnDataFromSctpToTransport(data) // 12. SctpTransport::SignalDataReceived(data) // [from the same thread, methods registered/connected to -// SctpTransport are called with the recieved data] +// SctpTransport are called with the received data] class UsrsctpTransport : public SctpTransportInternal, public sigslot::has_slots<> { public: diff --git a/media/sctp/usrsctp_transport_unittest.cc b/media/sctp/usrsctp_transport_unittest.cc index 59e9c59b3d..8fdbabc14a 100644 --- a/media/sctp/usrsctp_transport_unittest.cc +++ b/media/sctp/usrsctp_transport_unittest.cc @@ -36,7 +36,7 @@ static const int kTransport2Port = 5002; namespace cricket { -// This is essentially a buffer to hold recieved data. It stores only the last +// This is essentially a buffer to hold received data. It stores only the last // received data. Calling OnDataReceived twice overwrites old data with the // newer one. // TODO(ldixon): Implement constraints, and allow new data to be added to old diff --git a/modules/audio_coding/codecs/isac/fix/include/isacfix.h b/modules/audio_coding/codecs/isac/fix/include/isacfix.h index 87956a6997..dcc7b0991d 100644 --- a/modules/audio_coding/codecs/isac/fix/include/isacfix.h +++ b/modules/audio_coding/codecs/isac/fix/include/isacfix.h @@ -394,7 +394,7 @@ int16_t WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct* ISAC_main_inst); /**************************************************************************** * WebRtcIsacfix_GetNewBitStream(...) * - * This function returns encoded data, with the recieved bwe-index in the + * This function returns encoded data, with the received bwe-index in the * stream. It should always return a complete packet, i.e. only called once * even for 60 msec frames * diff --git a/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/modules/audio_coding/codecs/isac/fix/source/isacfix.c index 9a66591de1..a7d44e883d 100644 --- a/modules/audio_coding/codecs/isac/fix/source/isacfix.c +++ b/modules/audio_coding/codecs/isac/fix/source/isacfix.c @@ -381,7 +381,7 @@ int WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst, /**************************************************************************** * WebRtcIsacfix_GetNewBitStream(...) * - * This function returns encoded data, with the recieved bwe-index in the + * This function returns encoded data, with the received bwe-index in the * stream. It should always return a complete packet, i.e. only called once * even for 60 msec frames * diff --git a/modules/audio_coding/codecs/isac/main/include/isac.h b/modules/audio_coding/codecs/isac/main/include/isac.h index f45bbb3897..3b05a8bcda 100644 --- a/modules/audio_coding/codecs/isac/main/include/isac.h +++ b/modules/audio_coding/codecs/isac/main/include/isac.h @@ -453,7 +453,7 @@ int16_t WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst, /****************************************************************************** * WebRtcIsac_GetNewBitStream(...) * - * This function returns encoded data, with the recieved bwe-index in the + * This function returns encoded data, with the received bwe-index in the * stream. If the rate is set to a value less than bottleneck of codec * the new bistream will be re-encoded with the given target rate. * It should always return a complete packet, i.e. only called once diff --git a/modules/audio_coding/codecs/isac/main/source/isac.c b/modules/audio_coding/codecs/isac/main/source/isac.c index 73f132c228..456f447d9a 100644 --- a/modules/audio_coding/codecs/isac/main/source/isac.c +++ b/modules/audio_coding/codecs/isac/main/source/isac.c @@ -678,7 +678,7 @@ int WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, /****************************************************************************** * WebRtcIsac_GetNewBitStream(...) * - * This function returns encoded data, with the recieved bwe-index in the + * This function returns encoded data, with the received bwe-index in the * stream. If the rate is set to a value less than bottleneck of codec * the new bistream will be re-encoded with the given target rate. * It should always return a complete packet, i.e. only called once diff --git a/modules/audio_device/audio_device_buffer.h b/modules/audio_device/audio_device_buffer.h index ea6ab9a93e..9a6a88a1be 100644 --- a/modules/audio_device/audio_device_buffer.h +++ b/modules/audio_device/audio_device_buffer.h @@ -228,7 +228,7 @@ class AudioDeviceBuffer { // being printed in the LogStats() task. bool log_stats_ RTC_GUARDED_BY(task_queue_); - // Used for converting capture timestaps (recieved from AudioRecordThread + // Used for converting capture timestaps (received from AudioRecordThread // via AudioRecordJni::DataIsRecorded) to RTC clock. rtc::TimestampAligner timestamp_aligner_; diff --git a/modules/congestion_controller/receive_side_congestion_controller_unittest.cc b/modules/congestion_controller/receive_side_congestion_controller_unittest.cc index 2aade06cbc..f2fd6d11d7 100644 --- a/modules/congestion_controller/receive_side_congestion_controller_unittest.cc +++ b/modules/congestion_controller/receive_side_congestion_controller_unittest.cc @@ -81,7 +81,7 @@ TEST(ReceiveSideCongestionControllerTest, } TEST(ReceiveSideCongestionControllerTest, ConvergesToCapacity) { - Scenario s("recieve_cc_unit/converge"); + Scenario s("receive_cc_unit/converge"); NetworkSimulationConfig net_conf; net_conf.bandwidth = DataRate::KilobitsPerSec(1000); net_conf.delay = TimeDelta::Millis(50); @@ -100,7 +100,7 @@ TEST(ReceiveSideCongestionControllerTest, ConvergesToCapacity) { } TEST(ReceiveSideCongestionControllerTest, IsFairToTCP) { - Scenario s("recieve_cc_unit/tcp_fairness"); + Scenario s("receive_cc_unit/tcp_fairness"); NetworkSimulationConfig net_conf; net_conf.bandwidth = DataRate::KilobitsPerSec(1000); net_conf.delay = TimeDelta::Millis(50); diff --git a/modules/rtp_rtcp/source/rtcp_packet/loss_notification.h b/modules/rtp_rtcp/source/rtcp_packet/loss_notification.h index b23008c528..0f70cf75c3 100644 --- a/modules/rtp_rtcp/source/rtcp_packet/loss_notification.h +++ b/modules/rtp_rtcp/source/rtcp_packet/loss_notification.h @@ -42,8 +42,8 @@ class LossNotification : public Psfb { // Set all of the values transmitted by the loss notification message. // If the values may not be represented by a loss notification message, // false is returned, and no change is made to the object; this happens - // when `last_recieved` is ahead of `last_decoded` by more than 0x7fff. - // This is because `last_recieved` is represented on the wire as a delta, + // when `last_received` is ahead of `last_decoded` by more than 0x7fff. + // This is because `last_received` is represented on the wire as a delta, // and only 15 bits are available for that delta. ABSL_MUST_USE_RESULT bool Set(uint16_t last_decoded, diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc b/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc index e90cf047f2..ea6b49525a 100644 --- a/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc +++ b/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc @@ -1405,7 +1405,7 @@ TEST(RtcpTransceiverImplTest, ParsesRemb) { } TEST(RtcpTransceiverImplTest, - CombinesReportBlocksFromSenderAndRecieverReports) { + CombinesReportBlocksFromSenderAndReceiverReports) { MockNetworkLinkRtcpObserver link_observer; RtcpTransceiverConfig config = DefaultTestConfig(); config.network_link_observer = &link_observer; diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc index ee71c6ba7d..9d01e0772f 100644 --- a/pc/media_session_unittest.cc +++ b/pc/media_session_unittest.cc @@ -4405,7 +4405,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestSetAudioCodecs) { std::vector recv_codecs = MAKE_VECTOR(kAudioCodecs2); // The merged list of codecs should contain any send codecs that are also - // nominally in the recieve codecs list. Payload types should be picked from + // nominally in the receive codecs list. Payload types should be picked from // the send codecs and a number-of-channels of 0 and 1 should be equivalent // (set to 1). This equals what happens when the send codecs are used in an // offer and the receive codecs are used in the following answer. diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h index a32ece930b..15c092723e 100644 --- a/pc/sdp_offer_answer.h +++ b/pc/sdp_offer_answer.h @@ -386,7 +386,7 @@ class SdpOfferAnswerHandler : public SdpStateProvider, // to the SDP semantics. void FillInMissingRemoteMids(cricket::SessionDescription* remote_description); - // Returns an RtpTransciever, if available, that can be used to receive the + // Returns an RtpTransceiver, if available, that can be used to receive the // given media type according to JSEP rules. rtc::scoped_refptr> FindAvailableTransceiverToReceive(cricket::MediaType media_type) const; diff --git a/rtc_base/callback_list_unittest.cc b/rtc_base/callback_list_unittest.cc index 23dfff0bdd..e2bc6d515e 100644 --- a/rtc_base/callback_list_unittest.cc +++ b/rtc_base/callback_list_unittest.cc @@ -17,7 +17,7 @@ namespace webrtc { namespace { -TEST(CallbackList, NoRecieverSingleMessageTest) { +TEST(CallbackList, NoReceiverSingleMessageTest) { CallbackList c; c.Send("message"); diff --git a/rtc_tools/video_replay.cc b/rtc_tools/video_replay.cc index c9fe40ffc3..c03cc6cbc4 100644 --- a/rtc_tools/video_replay.cc +++ b/rtc_tools/video_replay.cc @@ -312,7 +312,7 @@ class DecoderIvfFileWriter : public test::FakeDecoder { }; // The RtpReplayer is responsible for parsing the configuration provided by the -// user, setting up the windows, recieve streams and decoders and then replaying +// user, setting up the windows, receive streams and decoders and then replaying // the provided RTP dump. class RtpReplayer final { public: @@ -382,7 +382,7 @@ class RtpReplayer final { } private: - // Holds all the shared memory structures required for a recieve stream. This + // Holds all the shared memory structures required for a receive stream. This // structure is used to prevent members being deallocated before the replay // has been finished. struct StreamState { diff --git a/sdk/objc/base/RTCVideoEncoder.h b/sdk/objc/base/RTCVideoEncoder.h index 2b5c952afa..2445d432d6 100644 --- a/sdk/objc/base/RTCVideoEncoder.h +++ b/sdk/objc/base/RTCVideoEncoder.h @@ -50,7 +50,7 @@ RTC_OBJC_EXPORT scaled, all resolutions comply with 'resolutionAlignment'. */ @property(nonatomic, readonly) BOOL applyAlignmentToAllSimulcastLayers; -/** If YES, the reciever is expected to resample/scale the source texture to the expected output +/** If YES, the receiver is expected to resample/scale the source texture to the expected output size. */ @property(nonatomic, readonly) BOOL supportsNativeHandle; diff --git a/sdk/objc/components/audio/RTCAudioSession+Private.h b/sdk/objc/components/audio/RTCAudioSession+Private.h index 4f5107f7e9..2be1b9fb3d 100644 --- a/sdk/objc/components/audio/RTCAudioSession+Private.h +++ b/sdk/objc/components/audio/RTCAudioSession+Private.h @@ -73,10 +73,10 @@ NS_ASSUME_NONNULL_BEGIN /** Returns a configuration error with the given description. */ - (NSError *)configurationErrorWithDescription:(NSString *)description; -/** Notifies the reciever that a playout glitch was detected. */ +/** Notifies the receiver that a playout glitch was detected. */ - (void)notifyDidDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches; -/** Notifies the reciever that there was an error when starting an audio unit. */ +/** Notifies the receiver that there was an error when starting an audio unit. */ - (void)notifyAudioUnitStartFailedWithError:(OSStatus)error; // Properties and methods for tests. diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc index d5bc2cd3f2..ec2bf587ba 100644 --- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc +++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc @@ -1074,7 +1074,7 @@ TEST(DefaultVideoQualityAnalyzerTest, } TEST(DefaultVideoQualityAnalyzerTest, - FrameCanBeReceivedByRecieverAfterItWasReceivedBySender) { + FrameCanBeReceivedByReceiverAfterItWasReceivedBySender) { std::unique_ptr frame_generator = test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight, /*type=*/absl::nullopt, diff --git a/test/scenario/video_stream_unittest.cc b/test/scenario/video_stream_unittest.cc index c1649a39b3..b37530dc50 100644 --- a/test/scenario/video_stream_unittest.cc +++ b/test/scenario/video_stream_unittest.cc @@ -70,7 +70,7 @@ TEST(VideoStreamTest, ReceivesFramesFromFileBasedStreams) { EXPECT_GE(frame_counts[1], expected_counts[1]); } -TEST(VideoStreamTest, RecievesVp8SimulcastFrames) { +TEST(VideoStreamTest, ReceivesVp8SimulcastFrames) { TimeDelta kRunTime = TimeDelta::Millis(500); int kFrameRate = 30; diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc index bcee8350b1..093a232907 100644 --- a/video/rtp_video_stream_receiver.cc +++ b/video/rtp_video_stream_receiver.cc @@ -1110,18 +1110,18 @@ bool RtpVideoStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet, uint32_t ntp_secs = 0; uint32_t ntp_frac = 0; uint32_t rtp_timestamp = 0; - uint32_t recieved_ntp_secs = 0; - uint32_t recieved_ntp_frac = 0; - if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, &recieved_ntp_secs, - &recieved_ntp_frac, &rtp_timestamp) != 0) { + uint32_t received_ntp_secs = 0; + uint32_t received_ntp_frac = 0; + if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, &received_ntp_secs, + &received_ntp_frac, &rtp_timestamp) != 0) { // Waiting for RTCP. return true; } - NtpTime recieved_ntp(recieved_ntp_secs, recieved_ntp_frac); - int64_t time_since_recieved = - clock_->CurrentNtpInMilliseconds() - recieved_ntp.ToMs(); + NtpTime received_ntp(received_ntp_secs, received_ntp_frac); + int64_t time_since_received = + clock_->CurrentNtpInMilliseconds() - received_ntp.ToMs(); // Don't use old SRs to estimate time. - if (time_since_recieved <= 1) { + if (time_since_received <= 1) { ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); absl::optional remote_to_local_clock_offset_ms = ntp_estimator_.EstimateRemoteToLocalClockOffsetMs(); diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc index c5594ba96e..5e9788ecf9 100644 --- a/video/rtp_video_stream_receiver2.cc +++ b/video/rtp_video_stream_receiver2.cc @@ -1028,18 +1028,18 @@ bool RtpVideoStreamReceiver2::DeliverRtcp(const uint8_t* rtcp_packet, uint32_t ntp_secs = 0; uint32_t ntp_frac = 0; uint32_t rtp_timestamp = 0; - uint32_t recieved_ntp_secs = 0; - uint32_t recieved_ntp_frac = 0; - if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, &recieved_ntp_secs, - &recieved_ntp_frac, &rtp_timestamp) != 0) { + uint32_t received_ntp_secs = 0; + uint32_t received_ntp_frac = 0; + if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, &received_ntp_secs, + &received_ntp_frac, &rtp_timestamp) != 0) { // Waiting for RTCP. return true; } - NtpTime recieved_ntp(recieved_ntp_secs, recieved_ntp_frac); - int64_t time_since_recieved = - clock_->CurrentNtpInMilliseconds() - recieved_ntp.ToMs(); + NtpTime received_ntp(received_ntp_secs, received_ntp_frac); + int64_t time_since_received = + clock_->CurrentNtpInMilliseconds() - received_ntp.ToMs(); // Don't use old SRs to estimate time. - if (time_since_recieved <= 1) { + if (time_since_received <= 1) { ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); absl::optional remote_to_local_clock_offset_ms = ntp_estimator_.EstimateRemoteToLocalClockOffsetMs();