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}
diff --git a/media/sctp/usrsctp_transport.h b/media/sctp/usrsctp_transport.h
index 2dd6abf..7c7ce8c 100644
--- a/media/sctp/usrsctp_transport.h
+++ b/media/sctp/usrsctp_transport.h
@@ -63,7 +63,7 @@
// 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 59e9c59..8fdbabc 100644
--- a/media/sctp/usrsctp_transport_unittest.cc
+++ b/media/sctp/usrsctp_transport_unittest.cc
@@ -36,7 +36,7 @@
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 87956a6..dcc7b09 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 @@
/****************************************************************************
* 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 9a66591..a7d44e8 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 @@
/****************************************************************************
* 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 f45bbb3..3b05a8b 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 @@
/******************************************************************************
* 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 73f132c..456f447 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 @@
/******************************************************************************
* 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 ea6ab9a..9a6a88a 100644
--- a/modules/audio_device/audio_device_buffer.h
+++ b/modules/audio_device/audio_device_buffer.h
@@ -228,7 +228,7 @@
// 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 2aade06c..f2fd6d1 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, 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, 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 b23008c..0f70cf7 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 @@
// 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 e90cf04..ea6b495 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,
- 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 ee71c6b..9d01e07 100644
--- a/pc/media_session_unittest.cc
+++ b/pc/media_session_unittest.cc
@@ -4405,7 +4405,7 @@
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.
diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h
index a32ece9..15c0927 100644
--- a/pc/sdp_offer_answer.h
+++ b/pc/sdp_offer_answer.h
@@ -386,7 +386,7 @@
// 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;
diff --git a/rtc_base/callback_list_unittest.cc b/rtc_base/callback_list_unittest.cc
index 23dfff0..e2bc6d5 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<std::string> c;
c.Send("message");
diff --git a/rtc_tools/video_replay.cc b/rtc_tools/video_replay.cc
index c9fe40f..c03cc6c 100644
--- a/rtc_tools/video_replay.cc
+++ b/rtc_tools/video_replay.cc
@@ -312,7 +312,7 @@
};
// 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 @@
}
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 2b5c952..2445d43 100644
--- a/sdk/objc/base/RTCVideoEncoder.h
+++ b/sdk/objc/base/RTCVideoEncoder.h
@@ -50,7 +50,7 @@
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 4f5107f..2be1b9f 100644
--- a/sdk/objc/components/audio/RTCAudioSession+Private.h
+++ b/sdk/objc/components/audio/RTCAudioSession+Private.h
@@ -73,10 +73,10 @@
/** 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 d5bc2cd..ec2bf58 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,
- FrameCanBeReceivedByRecieverAfterItWasReceivedBySender) {
+ FrameCanBeReceivedByReceiverAfterItWasReceivedBySender) {
std::unique_ptr<test::FrameGeneratorInterface> 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 c1649a3..b37530d 100644
--- a/test/scenario/video_stream_unittest.cc
+++ b/test/scenario/video_stream_unittest.cc
@@ -70,7 +70,7 @@
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 bcee835..093a232 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -1110,18 +1110,18 @@
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();
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index c5594ba..5e9788e 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -1028,18 +1028,18 @@
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();