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 <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36257}

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:

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

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
  *

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
  *

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

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

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_;
 

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);

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,

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;

pc/media_session_unittest.cc

@@ -4405,7 +4405,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestSetAudioCodecs) {
   std::vector<AudioCodec> 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.

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<RtpTransceiverProxyWithInternal<RtpTransceiver>>
   FindAvailableTransceiverToReceive(cricket::MediaType media_type) const;

rtc_base/callback_list_unittest.cc

@@ -17,7 +17,7 @@
 namespace webrtc {
 namespace {
 
-TEST(CallbackList, NoRecieverSingleMessageTest) {
+TEST(CallbackList, NoReceiverSingleMessageTest) {
   CallbackList<std::string> c;
 
   c.Send("message");

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 {

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;
 

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.

test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc

@@ -1074,7 +1074,7 @@ TEST(DefaultVideoQualityAnalyzerTest,
 }
 
 TEST(DefaultVideoQualityAnalyzerTest,
-     FrameCanBeReceivedByRecieverAfterItWasReceivedBySender) {
+     FrameCanBeReceivedByReceiverAfterItWasReceivedBySender) {
   std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
       test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
                                        /*type=*/absl::nullopt,

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;
 

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<int64_t> remote_to_local_clock_offset_ms =
         ntp_estimator_.EstimateRemoteToLocalClockOffsetMs();

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<int64_t> remote_to_local_clock_offset_ms =
         ntp_estimator_.EstimateRemoteToLocalClockOffsetMs();