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Use backticks not vertical bars to denote variables in comments for /api

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Bug: webrtc:12338
Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34554}
This commit is contained in:
Artem Titov 2021-07-25 21:50:14 +02:00 committed by WebRTC LUCI CQ
parent 7750d802a5
commit 0e61fdd27c
94 changed files with 455 additions and 455 deletions
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2
api/adaptation/resource.h
View file

@ -57,7 +57,7 @@ class RTC_EXPORT Resource : public rtc::RefCountInterface {
~Resource() override;
virtual std::string Name() const = 0;
// The |listener| may be informed of resource usage measurements on any task
// The `listener` may be informed of resource usage measurements on any task
// queue, but not after this method is invoked with the null argument.
virtual void SetResourceListener(ResourceListener* listener) = 0;
};

10
api/async_dns_resolver.h
View file

@ -41,10 +41,10 @@ namespace webrtc {
class AsyncDnsResolverResult {
public:
virtual ~AsyncDnsResolverResult() = default;
// Returns true iff the address from |Start| was successfully resolved.
// If the address was successfully resolved, sets |addr| to a copy of the
// address from |Start| with the IP address set to the top most resolved
// address of |family| (|addr| will have both hostname and the resolved ip).
// Returns true iff the address from `Start` was successfully resolved.
// If the address was successfully resolved, sets `addr` to a copy of the
// address from `Start` with the IP address set to the top most resolved
// address of `family` (`addr` will have both hostname and the resolved ip).
virtual bool GetResolvedAddress(int family,
rtc::SocketAddress* addr) const = 0;
// Returns error from resolver.
@ -55,7 +55,7 @@ class RTC_EXPORT AsyncDnsResolverInterface {
public:
virtual ~AsyncDnsResolverInterface() = default;
// Start address resolution of the hostname in |addr|.
// Start address resolution of the hostname in `addr`.
virtual void Start(const rtc::SocketAddress& addr,
std::function<void()> callback) = 0;
virtual const AsyncDnsResolverResult& result() const = 0;

2
api/audio/audio_frame.cc
View file

@ -52,7 +52,7 @@ void AudioFrame::Reset() {
}
void AudioFrame::ResetWithoutMuting() {
// TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
// TODO(wu): Zero is a valid value for `timestamp_`. We should initialize
// to an invalid value, or add a new member to indicate invalidity.
timestamp_ = 0;
elapsed_time_ms_ = -1;

6
api/audio/audio_frame.h
View file

@ -139,7 +139,7 @@ class AudioFrame {
int64_t profile_timestamp_ms_ = 0;
// Information about packets used to assemble this audio frame. This is needed
// by |SourceTracker| when the frame is delivered to the RTCRtpReceiver's
// by `SourceTracker` when the frame is delivered to the RTCRtpReceiver's
// MediaStreamTrack, in order to implement getContributingSources(). See:
// https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources
//
@ -149,7 +149,7 @@ class AudioFrame {
// sync buffer is the small sample-holding buffer located after the audio
// decoder and before where samples are assembled into output frames.
//
// |RtpPacketInfos| may also be empty if the audio samples did not come from
// `RtpPacketInfos` may also be empty if the audio samples did not come from
// RTP packets. E.g. if the audio were locally generated by packet loss
// concealment, comfort noise generation, etc.
RtpPacketInfos packet_infos_;
@ -165,7 +165,7 @@ class AudioFrame {
// Absolute capture timestamp when this audio frame was originally captured.
// This is only valid for audio frames captured on this machine. The absolute
// capture timestamp of a received frame is found in |packet_infos_|.
// capture timestamp of a received frame is found in `packet_infos_`.
// This timestamp MUST be based on the same clock as rtc::TimeMillis().
absl::optional<int64_t> absolute_capture_timestamp_ms_;

6
api/audio/audio_frame_processor.h
View file

@ -28,12 +28,12 @@ class AudioFrameProcessor {
// Processes the frame received from WebRTC, is called by WebRTC off the
// realtime audio capturing path. AudioFrameProcessor must reply with
// processed frames by calling |sink_callback| if it was provided in SetSink()
// call. |sink_callback| can be called in the context of Process().
// processed frames by calling `sink_callback` if it was provided in SetSink()
// call. `sink_callback` can be called in the context of Process().
virtual void Process(std::unique_ptr<AudioFrame> frame) = 0;
// Atomically replaces the current sink with the new one. Before the
// first call to this function, or if the provided |sink_callback| is nullptr,
// first call to this function, or if the provided `sink_callback` is nullptr,
// processed frames are simply discarded.
virtual void SetSink(OnAudioFrameCallback sink_callback) = 0;
};

6
api/audio/audio_mixer.h
View file

@ -35,9 +35,9 @@ class AudioMixer : public rtc::RefCountInterface {
kError, // The audio_frame will not be used.
};
// Overwrites |audio_frame|. The data_ field is overwritten with
// Overwrites `audio_frame`. The data_ field is overwritten with
// 10 ms of new audio (either 1 or 2 interleaved channels) at
// |sample_rate_hz|. All fields in |audio_frame| must be updated.
// `sample_rate_hz`. All fields in `audio_frame` must be updated.
virtual AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
AudioFrame* audio_frame) = 0;
@ -66,7 +66,7 @@ class AudioMixer : public rtc::RefCountInterface {
// should mix at a rate that doesn't cause quality loss of the
// sources' audio. The mixing rate is one of the rates listed in
// AudioProcessing::NativeRate. All fields in
// |audio_frame_for_mixing| must be updated.
// `audio_frame_for_mixing` must be updated.
virtual void Mix(size_t number_of_channels,
AudioFrame* audio_frame_for_mixing) = 0;

30
api/audio_codecs/audio_decoder.h
View file

@ -53,8 +53,8 @@ class AudioDecoder {
// Returns true if this packet contains DTX.
virtual bool IsDtxPacket() const;
// Decodes this frame of audio and writes the result in |decoded|.
// |decoded| must be large enough to store as many samples as indicated by a
// Decodes this frame of audio and writes the result in `decoded`.
// `decoded` must be large enough to store as many samples as indicated by a
// call to Duration() . On success, returns an absl::optional containing the
// total number of samples across all channels, as well as whether the
// decoder produced comfort noise or speech. On failure, returns an empty
@ -85,8 +85,8 @@ class AudioDecoder {
// Let the decoder parse this payload and prepare zero or more decodable
// frames. Each frame must be between 10 ms and 120 ms long. The caller must
// ensure that the AudioDecoder object outlives any frame objects returned by
// this call. The decoder is free to swap or move the data from the |payload|
// buffer. |timestamp| is the input timestamp, in samples, corresponding to
// this call. The decoder is free to swap or move the data from the `payload`
// buffer. `timestamp` is the input timestamp, in samples, corresponding to
// the start of the payload.
virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
uint32_t timestamp);
@ -95,12 +95,12 @@ class AudioDecoder {
// obsolete; callers should call ParsePayload instead. For now, subclasses
// must still implement DecodeInternal.
// Decodes |encode_len| bytes from |encoded| and writes the result in
// |decoded|. The maximum bytes allowed to be written into |decoded| is
// |max_decoded_bytes|. Returns the total number of samples across all
// channels. If the decoder produced comfort noise, |speech_type|
// Decodes `encode_len` bytes from `encoded` and writes the result in
// `decoded`. The maximum bytes allowed to be written into `decoded` is
// `max_decoded_bytes`. Returns the total number of samples across all
// channels. If the decoder produced comfort noise, `speech_type`
// is set to kComfortNoise, otherwise it is kSpeech. The desired output
// sample rate is provided in |sample_rate_hz|, which must be valid for the
// sample rate is provided in `sample_rate_hz`, which must be valid for the
// codec at hand.
int Decode(const uint8_t* encoded,
size_t encoded_len,
@ -123,11 +123,11 @@ class AudioDecoder {
// Calls the packet-loss concealment of the decoder to update the state after
// one or several lost packets. The caller has to make sure that the
// memory allocated in |decoded| should accommodate |num_frames| frames.
// memory allocated in `decoded` should accommodate `num_frames` frames.
virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
// Asks the decoder to generate packet-loss concealment and append it to the
// end of |concealment_audio|. The concealment audio should be in
// end of `concealment_audio`. The concealment audio should be in
// channel-interleaved format, with as many channels as the last decoded
// packet produced. The implementation must produce at least
// requested_samples_per_channel, or nothing at all. This is a signal to the
@ -146,19 +146,19 @@ class AudioDecoder {
// Returns the last error code from the decoder.
virtual int ErrorCode();
// Returns the duration in samples-per-channel of the payload in |encoded|
// which is |encoded_len| bytes long. Returns kNotImplemented if no duration
// Returns the duration in samples-per-channel of the payload in `encoded`
// which is `encoded_len` bytes long. Returns kNotImplemented if no duration
// estimate is available, or -1 in case of an error.
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
// Returns the duration in samples-per-channel of the redandant payload in
// |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
// `encoded` which is `encoded_len` bytes long. Returns kNotImplemented if no
// duration estimate is available, or -1 in case of an error.
virtual int PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const;
// Detects whether a packet has forward error correction. The packet is
// comprised of the samples in |encoded| which is |encoded_len| bytes long.
// comprised of the samples in `encoded` which is `encoded_len` bytes long.
// Returns true if the packet has FEC and false otherwise.
virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;

4
api/audio_codecs/audio_decoder_factory_template.h
View file

@ -89,8 +89,8 @@ class AudioDecoderFactoryT : public AudioDecoderFactory {
// Each decoder type is given as a template argument to the function; it should
// be a struct with the following static member functions:
//
// // Converts |audio_format| to a ConfigType instance. Returns an empty
// // optional if |audio_format| doesn't correctly specify a decoder of our
// // Converts `audio_format` to a ConfigType instance. Returns an empty
// // optional if `audio_format` doesn't correctly specify a decoder of our
// // type.
// absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
//

10
api/audio_codecs/audio_encoder.h
View file

@ -95,13 +95,13 @@ class AudioEncoder {
// This is the main struct for auxiliary encoding information. Each encoded
// packet should be accompanied by one EncodedInfo struct, containing the
// total number of |encoded_bytes|, the |encoded_timestamp| and the
// |payload_type|. If the packet contains redundant encodings, the |redundant|
// total number of `encoded_bytes`, the `encoded_timestamp` and the
// `payload_type`. If the packet contains redundant encodings, the `redundant`
// vector will be populated with EncodedInfoLeaf structs. Each struct in the
// vector represents one encoding; the order of structs in the vector is the
// same as the order in which the actual payloads are written to the byte
// stream. When EncoderInfoLeaf structs are present in the vector, the main
// struct's |encoded_bytes| will be the sum of all the |encoded_bytes| in the
// struct's `encoded_bytes` will be the sum of all the `encoded_bytes` in the
// vector.
struct EncodedInfo : public EncodedInfoLeaf {
EncodedInfo();
@ -143,7 +143,7 @@ class AudioEncoder {
// Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
// NumChannels() samples). Multi-channel audio must be sample-interleaved.
// The encoder appends zero or more bytes of output to |encoded| and returns
// The encoder appends zero or more bytes of output to `encoded` and returns
// additional encoding information. Encode() checks some preconditions, calls
// EncodeImpl() which does the actual work, and then checks some
// postconditions.
@ -205,7 +205,7 @@ class AudioEncoder {
virtual void DisableAudioNetworkAdaptor();
// Provides uplink packet loss fraction to this encoder to allow it to adapt.
// |uplink_packet_loss_fraction| is in the range [0.0, 1.0].
// `uplink_packet_loss_fraction` is in the range [0.0, 1.0].
virtual void OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction);

4
api/audio_codecs/audio_encoder_factory_template.h
View file

@ -103,8 +103,8 @@ class AudioEncoderFactoryT : public AudioEncoderFactory {
// Each encoder type is given as a template argument to the function; it should
// be a struct with the following static member functions:
//
// // Converts |audio_format| to a ConfigType instance. Returns an empty
// // optional if |audio_format| doesn't correctly specify an encoder of our
// // Converts `audio_format` to a ConfigType instance. Returns an empty
// // optional if `audio_format` doesn't correctly specify an encoder of our
// // type.
// absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
//

2
api/audio_codecs/audio_format.h
View file

@ -39,7 +39,7 @@ struct RTC_EXPORT SdpAudioFormat {
Parameters&& param);
~SdpAudioFormat();
// Returns true if this format is compatible with |o|. In SDP terminology:
// Returns true if this format is compatible with `o`. In SDP terminology:
// would it represent the same codec between an offer and an answer? As
// opposed to operator==, this method disregards codec parameters.
bool Matches(const SdpAudioFormat& o) const;

8
api/audio_codecs/opus/audio_encoder_opus_config.h
View file

@ -49,10 +49,10 @@ struct RTC_EXPORT AudioEncoderOpusConfig {
bool cbr_enabled;
int max_playback_rate_hz;
// |complexity| is used when the bitrate goes above
// |complexity_threshold_bps| + |complexity_threshold_window_bps|;
// |low_rate_complexity| is used when the bitrate falls below
// |complexity_threshold_bps| - |complexity_threshold_window_bps|. In the
// `complexity` is used when the bitrate goes above
// `complexity_threshold_bps` + `complexity_threshold_window_bps`;
// `low_rate_complexity` is used when the bitrate falls below
// `complexity_threshold_bps` - `complexity_threshold_window_bps`. In the
// interval in the middle, we keep using the most recent of the two
// complexity settings.
int complexity;

2
api/call/bitrate_allocation.h
View file

@ -32,7 +32,7 @@ struct BitrateAllocationUpdate {
double packet_loss_ratio = 0;
// Predicted round trip time.
TimeDelta round_trip_time = TimeDelta::PlusInfinity();
// |bwe_period| is deprecated, use |stable_target_bitrate| allocation instead.
// `bwe_period` is deprecated, use `stable_target_bitrate` allocation instead.
TimeDelta bwe_period = TimeDelta::PlusInfinity();
// Congestion window pushback bitrate reduction fraction. Used in
// VideoStreamEncoder to reduce the bitrate by the given fraction

2
api/candidate.cc
View file

@ -92,7 +92,7 @@ uint32_t Candidate::GetPriority(uint32_t type_preference,
// (2^8)*(local preference) +
// (2^0)*(256 - component ID)
// |local_preference| length is 2 bytes, 0-65535 inclusive.
// `local_preference` length is 2 bytes, 0-65535 inclusive.
// In our implemenation we will partion local_preference into
// 0 1
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5

8
api/candidate.h
View file

@ -112,7 +112,7 @@ class RTC_EXPORT Candidate {
uint32_t generation() const { return generation_; }
void set_generation(uint32_t generation) { generation_ = generation; }
// |network_cost| measures the cost/penalty of using this candidate. A network
// `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
// rtc::kNetworkCostMax indicates it should be used only as the last resort.
void set_network_cost(uint16_t network_cost) {
@ -167,9 +167,9 @@ class RTC_EXPORT Candidate {
bool operator!=(const Candidate& o) const;
// Returns a sanitized copy configured by the given booleans. If
// |use_host_address| is true, the returned copy has its IP removed from
// |address()|, which leads |address()| to be a hostname address. If
// |filter_related_address|, the returned copy has its related address reset
// `use_host_address` is true, the returned copy has its IP removed from
// `address()`, which leads `address()` to be a hostname address. If
// `filter_related_address`, the returned copy has its related address reset
// to the wildcard address (i.e. 0.0.0.0 for IPv4 and :: for IPv6). Note that
// setting both booleans to false returns an identical copy to the original
// candidate.

12
api/data_channel_interface.h
View file

@ -42,14 +42,14 @@ struct DataChannelInit {
// The max period of time in milliseconds in which retransmissions will be
// sent. After this time, no more retransmissions will be sent.
//
// Cannot be set along with |maxRetransmits|.
// This is called |maxPacketLifeTime| in the WebRTC JS API.
// Cannot be set along with `maxRetransmits`.
// This is called `maxPacketLifeTime` in the WebRTC JS API.
// Negative values are ignored, and positive values are clamped to [0-65535]
absl::optional<int> maxRetransmitTime;
// The max number of retransmissions.
//
// Cannot be set along with |maxRetransmitTime|.
// Cannot be set along with `maxRetransmitTime`.
// Negative values are ignored, and positive values are clamped to [0-65535]
absl::optional<int> maxRetransmits;
@ -57,7 +57,7 @@ struct DataChannelInit {
std::string protocol;
// True if the channel has been externally negotiated and we do not send an
// in-band signalling in the form of an "open" message. If this is true, |id|
// in-band signalling in the form of an "open" message. If this is true, `id`
// below must be set; otherwise it should be unset and will be negotiated
// in-band.
bool negotiated = false;
@ -70,7 +70,7 @@ struct DataChannelInit {
};
// At the JavaScript level, data can be passed in as a string or a blob, so
// this structure's |binary| flag tells whether the data should be interpreted
// this structure's `binary` flag tells whether the data should be interpreted
// as binary or text.
struct DataBuffer {
DataBuffer(const rtc::CopyOnWriteBuffer& data, bool binary)
@ -180,7 +180,7 @@ class RTC_EXPORT DataChannelInterface : public rtc::RefCountInterface {
// https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.7
virtual void Close() = 0;
// Sends |data| to the remote peer. If the data can't be sent at the SCTP
// Sends `data` to the remote peer. If the data can't be sent at the SCTP
// level (due to congestion control), it's buffered at the data channel level,
// up to a maximum of 16MB. If Send is called while this buffer is full, the
// data channel will be closed abruptly.

20
api/dtmf_sender_interface.h
View file

@ -23,8 +23,8 @@ namespace webrtc {
// DtmfSender.
class DtmfSenderObserverInterface {
public:
// Triggered when DTMF |tone| is sent.
// If |tone| is empty that means the DtmfSender has sent out all the given
// Triggered when DTMF `tone` is sent.
// If `tone` is empty that means the DtmfSender has sent out all the given
// tones.
// The callback includes the state of the tone buffer at the time when
// the tone finished playing.
@ -58,7 +58,7 @@ class DtmfSenderInterface : public rtc::RefCountInterface {
// able to send packets, and a "telephone-event" codec must be negotiated.
virtual bool CanInsertDtmf() = 0;
// Queues a task that sends the DTMF |tones|. The |tones| parameter is treated
// Queues a task that sends the DTMF `tones`. The `tones` parameter is treated
// as a series of characters. The characters 0 through 9, A through D, #, and
// * generate the associated DTMF tones. The characters a to d are equivalent
// to A to D. The character ',' indicates a delay of 2 seconds before
@ -66,18 +66,18 @@ class DtmfSenderInterface : public rtc::RefCountInterface {
//
// Unrecognized characters are ignored.
//
// The |duration| parameter indicates the duration in ms to use for each
// character passed in the |tones| parameter. The duration cannot be more
// The `duration` parameter indicates the duration in ms to use for each
// character passed in the `tones` parameter. The duration cannot be more
// than 6000 or less than 70.
//
// The |inter_tone_gap| parameter indicates the gap between tones in ms. The
// |inter_tone_gap| must be at least 50 ms but should be as short as
// The `inter_tone_gap` parameter indicates the gap between tones in ms. The
// `inter_tone_gap` must be at least 50 ms but should be as short as
// possible.
//
// The |comma_delay| parameter indicates the delay after the ','
// character. InsertDtmf specifies |comma_delay| as an argument
// The `comma_delay` parameter indicates the delay after the ','
// character. InsertDtmf specifies `comma_delay` as an argument
// with a default value of 2 seconds as per the WebRTC spec. This parameter
// allows users to comply with legacy WebRTC clients. The |comma_delay|
// allows users to comply with legacy WebRTC clients. The `comma_delay`
// must be at least 50 ms.
//
// If InsertDtmf is called on the same object while an existing task for this

2
api/fec_controller.h
View file

@ -38,7 +38,7 @@ class VCMProtectionCallback {
// FecController calculates how much of the allocated network
// capacity that can be used by an encoder and how much that
// is needed for redundant packets such as FEC and NACK. It uses an
// implementation of |VCMProtectionCallback| to set new FEC parameters and get
// implementation of `VCMProtectionCallback` to set new FEC parameters and get
// the bitrate currently used for FEC and NACK.
// Usage:
// Setup by calling SetProtectionMethod and SetEncodingData.

4
api/frame_transformer_interface.h
View file

@ -30,7 +30,7 @@ class TransformableFrameInterface {
// method call.
virtual rtc::ArrayView<const uint8_t> GetData() const = 0;
// Copies |data| into the owned frame payload data.
// Copies `data` into the owned frame payload data.
virtual void SetData(rtc::ArrayView<const uint8_t> data) = 0;
virtual uint32_t GetTimestamp() const = 0;
@ -78,7 +78,7 @@ class TransformedFrameCallback : public rtc::RefCountInterface {
// the TransformedFrameCallback interface (see above).
class FrameTransformerInterface : public rtc::RefCountInterface {
public:
// Transforms |frame| using the implementing class' processing logic.
// Transforms `frame` using the implementing class' processing logic.
virtual void Transform(
std::unique_ptr<TransformableFrameInterface> transformable_frame) = 0;

14
api/jsep.h
View file

@ -73,7 +73,7 @@ class RTC_EXPORT IceCandidateInterface {
// Creates a IceCandidateInterface based on SDP string.
// Returns null if the sdp string can't be parsed.
// |error| may be null.
// `error` may be null.
RTC_EXPORT IceCandidateInterface* CreateIceCandidate(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& sdp,
@ -91,7 +91,7 @@ class IceCandidateCollection {
public:
virtual ~IceCandidateCollection() {}
virtual size_t count() const = 0;
// Returns true if an equivalent |candidate| exist in the collection.
// Returns true if an equivalent `candidate` exist in the collection.
virtual bool HasCandidate(const IceCandidateInterface* candidate) const = 0;
virtual const IceCandidateInterface* at(size_t index) const = 0;
};
@ -158,7 +158,7 @@ class RTC_EXPORT SessionDescriptionInterface {
virtual SdpType GetType() const;
// kOffer/kPrAnswer/kAnswer
// TODO(steveanton): Remove this in favor of |GetType| that returns SdpType.
// TODO(steveanton): Remove this in favor of `GetType` that returns SdpType.
virtual std::string type() const = 0;
// Adds the specified candidate to the description.
@ -190,7 +190,7 @@ class RTC_EXPORT SessionDescriptionInterface {
// Creates a SessionDescriptionInterface based on the SDP string and the type.
// Returns null if the sdp string can't be parsed or the type is unsupported.
// |error| may be null.
// `error` may be null.
// TODO(steveanton): This function is deprecated. Please use the functions below
// which take an SdpType enum instead. Remove this once it is no longer used.
RTC_EXPORT SessionDescriptionInterface* CreateSessionDescription(
@ -200,8 +200,8 @@ RTC_EXPORT SessionDescriptionInterface* CreateSessionDescription(
// Creates a SessionDescriptionInterface based on the SDP string and the type.
// Returns null if the SDP string cannot be parsed.
// If using the signature with |error_out|, details of the parsing error may be
// written to |error_out| if it is not null.
// If using the signature with `error_out`, details of the parsing error may be
// written to `error_out` if it is not null.
RTC_EXPORT std::unique_ptr<SessionDescriptionInterface>
CreateSessionDescription(SdpType type, const std::string& sdp);
RTC_EXPORT std::unique_ptr<SessionDescriptionInterface>
@ -221,7 +221,7 @@ std::unique_ptr<SessionDescriptionInterface> CreateSessionDescription(
class RTC_EXPORT CreateSessionDescriptionObserver
: public rtc::RefCountInterface {
public:
// This callback transfers the ownership of the |desc|.
// This callback transfers the ownership of the `desc`.
// TODO(deadbeef): Make this take an std::unique_ptr<> to avoid confusion
// around ownership.
virtual void OnSuccess(SessionDescriptionInterface* desc) = 0;

2
api/jsep_ice_candidate.h
View file

@ -37,7 +37,7 @@ class RTC_EXPORT JsepIceCandidate : public IceCandidateInterface {
JsepIceCandidate(const JsepIceCandidate&) = delete;
JsepIceCandidate& operator=(const JsepIceCandidate&) = delete;
~JsepIceCandidate() override;
// |err| may be null.
// `err` may be null.
bool Initialize(const std::string& sdp, SdpParseError* err);
void SetCandidate(const cricket::Candidate& candidate) {
candidate_ = candidate;

2
api/jsep_session_description.h
View file

@ -43,7 +43,7 @@ class JsepSessionDescription : public SessionDescriptionInterface {
absl::string_view session_version);
virtual ~JsepSessionDescription();
// Takes ownership of |description|.
// Takes ownership of `description`.
bool Initialize(std::unique_ptr<cricket::SessionDescription> description,
const std::string& session_id,
const std::string& session_version);

6
api/media_stream_interface.h
View file

@ -200,7 +200,7 @@ class AudioTrackSinkInterface {
RTC_NOTREACHED() << "This method must be overridden, or not used.";
}
// In this method, |absolute_capture_timestamp_ms|, when available, is
// 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
// rtc::TimeMillis().
@ -240,7 +240,7 @@ class RTC_EXPORT AudioSourceInterface : public MediaSourceInterface {
// TODO(deadbeef): Makes all the interfaces pure virtual after they're
// implemented in chromium.
// Sets the volume of the source. |volume| is in the range of [0, 10].
// Sets the volume of the source. `volume` is in the range of [0, 10].
// TODO(tommi): This method should be on the track and ideally volume should
// be applied in the track in a way that does not affect clones of the track.
virtual void SetVolume(double volume) {}
@ -268,7 +268,7 @@ class AudioProcessorInterface : public rtc::RefCountInterface {
AudioProcessingStats apm_statistics;
};
// Get audio processor statistics. The |has_remote_tracks| argument should be
// Get audio processor statistics. The `has_remote_tracks` argument should be
// set if there are active remote tracks (this would usually be true during
// a call). If there are no remote tracks some of the stats will not be set by
// the AudioProcessor, because they only make sense if there is at least one

26
api/neteq/neteq.h
View file

@ -183,7 +183,7 @@ class NetEq {
SdpAudioFormat sdp_format;
};
// Creates a new NetEq object, with parameters set in |config|. The |config|
// Creates a new NetEq object, with parameters set in `config`. The `config`
// object will only have to be valid for the duration of the call to this
// method.
static NetEq* Create(
@ -205,15 +205,15 @@ class NetEq {
virtual void InsertEmptyPacket(const RTPHeader& rtp_header) = 0;
// Instructs NetEq to deliver 10 ms of audio data. The data is written to
// |audio_frame|. All data in |audio_frame| is wiped; |data_|, |speech_type_|,
// |num_channels_|, |sample_rate_hz_|, |samples_per_channel_|, and
// |vad_activity_| are updated upon success. If an error is returned, some
// `audio_frame`. All data in `audio_frame` is wiped; `data_`, `speech_type_`,
// `num_channels_`, `sample_rate_hz_`, `samples_per_channel_`, and
// `vad_activity_` are updated upon success. If an error is returned, some
// fields may not have been updated, or may contain inconsistent values.
// If muted state is enabled (through Config::enable_muted_state), |muted|
// If muted state is enabled (through Config::enable_muted_state), `muted`
// may be set to true after a prolonged expand period. When this happens, the
// |data_| in |audio_frame| is not written, but should be interpreted as being
// `data_` in `audio_frame` is not written, but should be interpreted as being
// all zeros. For testing purposes, an override can be supplied in the
// |action_override| argument, which will cause NetEq to take this action
// `action_override` argument, which will cause NetEq to take this action
// next, instead of the action it would normally choose. An optional output
// argument for fetching the current sample rate can be provided, which
// will return the same value as last_output_sample_rate_hz() but will avoid
@ -228,12 +228,12 @@ class NetEq {
// Replaces the current set of decoders with the given one.
virtual void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) = 0;
// Associates |rtp_payload_type| with the given codec, which NetEq will
// Associates `rtp_payload_type` with the given codec, which NetEq will
// instantiate when it needs it. Returns true iff successful.
virtual bool RegisterPayloadType(int rtp_payload_type,
const SdpAudioFormat& audio_format) = 0;
// Removes |rtp_payload_type| from the codec database. Returns 0 on success,
// Removes `rtp_payload_type` from the codec database. Returns 0 on success,
// -1 on failure. Removing a payload type that is not registered is ok and
// will not result in an error.
virtual int RemovePayloadType(uint8_t rtp_payload_type) = 0;
@ -250,12 +250,12 @@ class NetEq {
// Sets a maximum delay in milliseconds for packet buffer. The latency will
// not exceed the given value, even required delay (given the channel
// conditions) is higher. Calling this method has the same effect as setting
// the |max_delay_ms| value in the NetEq::Config struct.
// the `max_delay_ms` value in the NetEq::Config struct.
virtual bool SetMaximumDelay(int delay_ms) = 0;
// Sets a base minimum delay in milliseconds for packet buffer. The minimum
// delay which is set via |SetMinimumDelay| can't be lower than base minimum
// delay. Calling this method is similar to setting the |min_delay_ms| value
// delay which is set via `SetMinimumDelay` can't be lower than base minimum
// delay. Calling this method is similar to setting the `min_delay_ms` value
// in the NetEq::Config struct. Returns true if the base minimum is
// successfully applied, otherwise false is returned.
virtual bool SetBaseMinimumDelayMs(int delay_ms) = 0;
@ -272,7 +272,7 @@ class NetEq {
// The packet buffer part of the delay is not updated during DTX/CNG periods.
virtual int FilteredCurrentDelayMs() const = 0;
// Writes the current network statistics to |stats|. The statistics are reset
// Writes the current network statistics to `stats`. The statistics are reset
// after the call.
virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;

18
api/neteq/neteq_controller.h
View file

@ -115,13 +115,13 @@ class NetEqController {
virtual void SoftReset() = 0;
// Given info about the latest received packet, and current jitter buffer
// status, returns the operation. |target_timestamp| and |expand_mutefactor|
// are provided for reference. |last_packet_samples| is the number of samples
// status, returns the operation. `target_timestamp` and `expand_mutefactor`
// are provided for reference. `last_packet_samples` is the number of samples
// obtained from the last decoded frame. If there is a packet available, it
// should be supplied in |packet|. The mode resulting from the last call to
// NetEqImpl::GetAudio is supplied in |last_mode|. If there is a DTMF event to
// play, |play_dtmf| should be set to true. The output variable
// |reset_decoder| will be set to true if a reset is required; otherwise it is
// should be supplied in `packet`. The mode resulting from the last call to
// NetEqImpl::GetAudio is supplied in `last_mode`. If there is a DTMF event to
// play, `play_dtmf` should be set to true. The output variable
// `reset_decoder` will be set to true if a reset is required; otherwise it is
// left unchanged (i.e., it can remain true if it was true before the call).
virtual NetEq::Operation GetDecision(const NetEqStatus& status,
bool* reset_decoder) = 0;
@ -144,11 +144,11 @@ class NetEqController {
virtual bool SetBaseMinimumDelay(int delay_ms) = 0;
virtual int GetBaseMinimumDelay() const = 0;
// These methods test the |cng_state_| for different conditions.
// These methods test the `cng_state_` for different conditions.
virtual bool CngRfc3389On() const = 0;
virtual bool CngOff() const = 0;
// Resets the |cng_state_| to kCngOff.
// Resets the `cng_state_` to kCngOff.
virtual void SetCngOff() = 0;
// Reports back to DecisionLogic whether the decision to do expand remains or
@ -157,7 +157,7 @@ class NetEqController {
// sync buffer.
virtual void ExpandDecision(NetEq::Operation operation) = 0;
// Adds |value| to |sample_memory_|.
// Adds `value` to `sample_memory_`.
virtual void AddSampleMemory(int32_t value) = 0;
// Returns the target buffer level in ms.

2
api/neteq/neteq_controller_factory.h
View file

@ -23,7 +23,7 @@ class NetEqControllerFactory {
public:
virtual ~NetEqControllerFactory() = default;
// Creates a new NetEqController object, with parameters set in |config|.
// Creates a new NetEqController object, with parameters set in `config`.
virtual std::unique_ptr<NetEqController> CreateNetEqController(
const NetEqController::Config& config) const = 0;
};

2
api/neteq/neteq_factory.h
View file

@ -24,7 +24,7 @@ class NetEqFactory {
public:
virtual ~NetEqFactory() = default;
// Creates a new NetEq object, with parameters set in |config|. The |config|
// Creates a new NetEq object, with parameters set in `config`. The `config`
// object will only have to be valid for the duration of the call to this
// method.
virtual std::unique_ptr<NetEq> CreateNetEq(

6
api/numerics/samples_stats_counter.h
View file

@ -82,7 +82,7 @@ class SamplesStatsCounter {
// additions were done. This function may not be called if there are no
// samples.
//
// |percentile| has to be in [0; 1]. 0 percentile is the min in the array and
// `percentile` has to be in [0; 1]. 0 percentile is the min in the array and
// 1 percentile is the max in the array.
double GetPercentile(double percentile);
// Returns array view with all samples added into counter. There are no
@ -105,14 +105,14 @@ class SamplesStatsCounter {
bool sorted_ = false;
};
// Multiply all sample values on |value| and return new SamplesStatsCounter
// Multiply all sample values on `value` and return new SamplesStatsCounter
// with resulted samples. Doesn't change origin SamplesStatsCounter.
SamplesStatsCounter operator*(const SamplesStatsCounter& counter, double value);
inline SamplesStatsCounter operator*(double value,
const SamplesStatsCounter& counter) {
return counter * value;
}
// Divide all sample values on |value| and return new SamplesStatsCounter with
// Divide all sample values on `value` and return new SamplesStatsCounter with
// resulted samples. Doesn't change origin SamplesStatsCounter.
SamplesStatsCounter operator/(const SamplesStatsCounter& counter, double value);

86
api/peer_connection_interface.h
View file

@ -235,9 +235,9 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
std::string username;
std::string password;
TlsCertPolicy tls_cert_policy = kTlsCertPolicySecure;
// If the URIs in |urls| only contain IP addresses, this field can be used
// If the URIs in `urls` only contain IP addresses, this field can be used
// to indicate the hostname, which may be necessary for TLS (using the SNI
// extension). If |urls| itself contains the hostname, this isn't
// extension). If `urls` itself contains the hostname, this isn't
// necessary.
std::string hostname;
// List of protocols to be used in the TLS ALPN extension.
@ -526,7 +526,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// re-determining was removed in ICEbis (ICE v2).
bool redetermine_role_on_ice_restart = true;
// This flag is only effective when |continual_gathering_policy| is
// This flag is only effective when `continual_gathering_policy` is
// GATHER_CONTINUALLY.
//
// If true, after the ICE transport type is changed such that new types of
@ -712,8 +712,8 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
};
// Used by GetStats to decide which stats to include in the stats reports.
// |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
// |kStatsOutputLevelDebug| includes both the standard stats and additional
// `kStatsOutputLevelStandard` includes the standard stats for Javascript API;
// `kStatsOutputLevelDebug` includes both the standard stats and additional
// stats for debugging purposes.
enum StatsOutputLevel {
kStatsOutputLevelStandard,
@ -754,10 +754,10 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// Add a new MediaStreamTrack to be sent on this PeerConnection, and return
// the newly created RtpSender. The RtpSender will be associated with the
// streams specified in the |stream_ids| list.
// streams specified in the `stream_ids` list.
//
// Errors:
// - INVALID_PARAMETER: |track| is null, has a kind other than audio or video,
// - INVALID_PARAMETER: `track` is null, has a kind other than audio or video,
// or a sender already exists for the track.
// - INVALID_STATE: The PeerConnection is closed.
virtual RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
@ -774,7 +774,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// corresponding RtpTransceiver direction as no longer sending.
//
// Errors:
// - INVALID_PARAMETER: |sender| is null or (Plan B only) the sender is not
// - INVALID_PARAMETER: `sender` is null or (Plan B only) the sender is not
// associated with this PeerConnection.
// - INVALID_STATE: PeerConnection is closed.
// TODO(bugs.webrtc.org/9534): Rename to RemoveTrack once the other signature
@ -786,7 +786,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// transceivers. Adding a transceiver will cause future calls to CreateOffer
// to add a media description for the corresponding transceiver.
//
// The initial value of |mid| in the returned transceiver is null. Setting a
// The initial value of `mid` in the returned transceiver is null. Setting a
// new session description may change it to a non-null value.
//
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
@ -805,7 +805,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// of the transceiver (and sender/receiver) will be derived from the kind of
// the track.
// Errors:
// - INVALID_PARAMETER: |track| is null.
// - INVALID_PARAMETER: `track` is null.
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track) = 0;
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
@ -815,7 +815,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// Adds a transceiver with the given kind. Can either be MEDIA_TYPE_AUDIO or
// MEDIA_TYPE_VIDEO.
// Errors:
// - INVALID_PARAMETER: |media_type| is not MEDIA_TYPE_AUDIO or
// - INVALID_PARAMETER: `media_type` is not MEDIA_TYPE_AUDIO or
// MEDIA_TYPE_VIDEO.
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(cricket::MediaType media_type) = 0;
@ -830,9 +830,9 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// The standard way to do this would be through "addTransceiver", but we
// don't support that API yet.
//
// |kind| must be "audio" or "video".
// `kind` must be "audio" or "video".
//
// |stream_id| is used to populate the msid attribute; if empty, one will
// `stream_id` is used to populate the msid attribute; if empty, one will
// be generated automatically.
//
// This method is not supported with kUnifiedPlan semantics. Please use
@ -986,7 +986,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// returned by CreateOffer() or CreateAnswer() or else the operation should
// fail. Our implementation however allows some amount of "SDP munging", but
// please note that this is HIGHLY DISCOURAGED. If you do not intent to munge
// SDP, the method below that doesn't take |desc| as an argument will create
// SDP, the method below that doesn't take `desc` as an argument will create
// the offer or answer for you.
//
// The observer is invoked as soon as the operation completes, which could be
@ -1044,10 +1044,10 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
virtual PeerConnectionInterface::RTCConfiguration GetConfiguration() = 0;
// Sets the PeerConnection's global configuration to |config|.
// Sets the PeerConnection's global configuration to `config`.
//
// The members of |config| that may be changed are |type|, |servers|,
// |ice_candidate_pool_size| and |prune_turn_ports| (though the candidate
// The members of `config` that may be changed are `type`, `servers`,
// `ice_candidate_pool_size` and `prune_turn_ports` (though the candidate
// pool size can't be changed after the first call to SetLocalDescription).
// Note that this means the BUNDLE and RTCP-multiplexing policies cannot be
// changed with this method.
@ -1055,14 +1055,14 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// Any changes to STUN/TURN servers or ICE candidate policy will affect the
// next gathering phase, and cause the next call to createOffer to generate
// new ICE credentials, as described in JSEP. This also occurs when
// |prune_turn_ports| changes, for the same reasoning.
// `prune_turn_ports` changes, for the same reasoning.
//
// If an error occurs, returns false and populates |error| if non-null:
// - INVALID_MODIFICATION if |config| contains a modified parameter other
// If an error occurs, returns false and populates `error` if non-null:
// - INVALID_MODIFICATION if `config` contains a modified parameter other
// than one of the parameters listed above.
// - INVALID_RANGE if |ice_candidate_pool_size| is out of range.
// - INVALID_RANGE if `ice_candidate_pool_size` is out of range.
// - SYNTAX_ERROR if parsing an ICE server URL failed.
// - INVALID_PARAMETER if a TURN server is missing |username| or |password|.
// - INVALID_PARAMETER if a TURN server is missing `username` or `password`.
// - INTERNAL_ERROR if an unexpected error occurred.
//
// TODO(nisse): Make this pure virtual once all Chrome subclasses of
@ -1071,9 +1071,9 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
const PeerConnectionInterface::RTCConfiguration& config);
// Provides a remote candidate to the ICE Agent.
// A copy of the |candidate| will be created and added to the remote
// A copy of the `candidate` will be created and added to the remote
// description. So the caller of this method still has the ownership of the
// |candidate|.
// `candidate`.
// TODO(hbos): The spec mandates chaining this operation onto the operations
// chain; deprecate and remove this version in favor of the callback-based
// signature.
@ -1096,13 +1096,13 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// this PeerConnection. Other limitations might affect these limits and
// are respected (for example "b=AS" in SDP).
//
// Setting |current_bitrate_bps| will reset the current bitrate estimate
// Setting `current_bitrate_bps` will reset the current bitrate estimate
// to the provided value.
virtual RTCError SetBitrate(const BitrateSettings& bitrate) = 0;
// Enable/disable playout of received audio streams. Enabled by default. Note
// that even if playout is enabled, streams will only be played out if the
// appropriate SDP is also applied. Setting |playout| to false will stop
// appropriate SDP is also applied. Setting `playout` to false will stop
// playout of the underlying audio device but starts a task which will poll
// for audio data every 10ms to ensure that audio processing happens and the
// audio statistics are updated.
@ -1157,13 +1157,13 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) {}
// Start RtcEventLog using an existing output-sink. Takes ownership of
// |output| and passes it on to Call, which will take the ownership. If the
// `output` and passes it on to Call, which will take the ownership. If the
// operation fails the output will be closed and deallocated. The event log
// will send serialized events to the output object every |output_period_ms|.
// will send serialized events to the output object every `output_period_ms`.
// Applications using the event log should generally make their own trade-off
// regarding the output period. A long period is generally more efficient,
// with potential drawbacks being more bursty thread usage, and more events
// lost in case the application crashes. If the |output_period_ms| argument is
// lost in case the application crashes. If the `output_period_ms` argument is
// omitted, webrtc selects a default deemed to be workable in most cases.
virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) = 0;
@ -1222,7 +1222,7 @@ class PeerConnectionObserver {
// Used to fire spec-compliant onnegotiationneeded events, which should only
// fire when the Operations Chain is empty. The observer is responsible for
// queuing a task (e.g. Chromium: jump to main thread) to maybe fire the
// event. The event identified using |event_id| must only fire if
// event. The event identified using `event_id` must only fire if
// PeerConnection::ShouldFireNegotiationNeededEvent() returns true since it is
// possible for the event to become invalidated by operations subsequently
// chained.
@ -1256,7 +1256,7 @@ class PeerConnectionObserver {
// Gathering of an ICE candidate failed.
// See https://w3c.github.io/webrtc-pc/#event-icecandidateerror
// |host_candidate| is a stringified socket address.
// `host_candidate` is a stringified socket address.
virtual void OnIceCandidateError(const std::string& host_candidate,
const std::string& url,
int error_code,
@ -1393,7 +1393,7 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final {
network_state_predictor_factory;
std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
// This will only be used if CreatePeerConnection is called without a
// |port_allocator|, causing the default allocator and network manager to be
// `port_allocator`, causing the default allocator and network manager to be
// used.
std::unique_ptr<rtc::NetworkMonitorFactory> network_monitor_factory;
std::unique_ptr<NetEqFactory> neteq_factory;
@ -1467,12 +1467,12 @@ class RTC_EXPORT PeerConnectionFactoryInterface
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies);
// Deprecated; |allocator| and |cert_generator| may be null, in which case
// Deprecated; `allocator` and `cert_generator` may be null, in which case
// default implementations will be used.
//
// |observer| must not be null.
// `observer` must not be null.
//
// Note that this method does not take ownership of |observer|; it's the
// Note that this method does not take ownership of `observer`; it's the
// responsibility of the caller to delete it. It can be safely deleted after
// Close has been called on the returned PeerConnection, which ensures no
// more observer callbacks will be invoked.
@ -1483,13 +1483,13 @@ class RTC_EXPORT PeerConnectionFactoryInterface
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer);
// Returns the capabilities of an RTP sender of type |kind|.
// Returns the capabilities of an RTP sender of type `kind`.
// If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
// TODO(orphis): Make pure virtual when all subclasses implement it.
virtual RtpCapabilities GetRtpSenderCapabilities(
cricket::MediaType kind) const;
// Returns the capabilities of an RTP receiver of type |kind|.
// Returns the capabilities of an RTP receiver of type `kind`.
// If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
// TODO(orphis): Make pure virtual when all subclasses implement it.
virtual RtpCapabilities GetRtpReceiverCapabilities(
@ -1499,22 +1499,22 @@ class RTC_EXPORT PeerConnectionFactoryInterface
const std::string& stream_id) = 0;
// Creates an AudioSourceInterface.
// |options| decides audio processing settings.
// `options` decides audio processing settings.
virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const cricket::AudioOptions& options) = 0;
// Creates a new local VideoTrack. The same |source| can be used in several
// Creates a new local VideoTrack. The same `source` can be used in several
// tracks.
virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
const std::string& label,
VideoTrackSourceInterface* source) = 0;
// Creates an new AudioTrack. At the moment |source| can be null.
// Creates an new AudioTrack. At the moment `source` can be null.
virtual rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
const std::string& label,
AudioSourceInterface* source) = 0;
// Starts AEC dump using existing file. Takes ownership of |file| and passes
// Starts AEC dump using existing file. Takes ownership of `file` and passes
// it on to VoiceEngine (via other objects) immediately, which will take
// the ownerhip. If the operation fails, the file will be closed.
// A maximum file size in bytes can be specified. When the file size limit is
@ -1549,8 +1549,8 @@ class RTC_EXPORT PeerConnectionFactoryInterface
// video-specific interfaces, and omit the corresponding modules from its
// build.
//
// If |network_thread| or |worker_thread| are null, the PeerConnectionFactory
// will create the necessary thread internally. If |signaling_thread| is null,
// 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 rtc::Thread object if needed.
RTC_EXPORT rtc::scoped_refptr<PeerConnectionFactoryInterface>

2
api/rtc_error.h
View file

@ -176,7 +176,7 @@ inline std::ostream& operator<<( // no-presubmit-check TODO(webrtc:8982)
#endif // WEBRTC_UNIT_TEST
// Helper macro that can be used by implementations to create an error with a
// message and log it. |message| should be a string literal or movable
// message and log it. `message` should be a string literal or movable
// std::string.
#define LOG_AND_RETURN_ERROR_EX(type, message, severity) \
{ \

2
api/rtc_event_log/rtc_event_log.h
View file

@ -42,7 +42,7 @@ class RtcEventLog {
// which it would be permissible to read and/or modify it.
virtual void StopLogging() = 0;
// Stops logging to file and calls |callback| when the file has been closed.
// Stops logging to file and calls `callback` when the file has been closed.
// Note that it is not safe to call any other members, including the
// destructor, until the callback has been called.
// TODO(srte): Remove default implementation when it's safe to do so.

2
api/rtc_event_log_output.h
View file

@ -29,7 +29,7 @@ class RtcEventLogOutput {
// Write encoded events to an output. Returns true if the output was
// successfully written in its entirety. Otherwise, no guarantee is given
// about how much data was written, if any. The output sink becomes inactive
// after the first time |false| is returned. Write() may not be called on
// after the first time `false` is returned. Write() may not be called on
// an inactive output sink.
virtual bool Write(const std::string& output) = 0;

8
api/rtp_packet_info.h
View file

@ -23,9 +23,9 @@
namespace webrtc {
//
// Structure to hold information about a received |RtpPacket|. It is primarily
// Structure to hold information about a received `RtpPacket`. It is primarily
// used to carry per-packet information from when a packet is received until
// the information is passed to |SourceTracker|.
// the information is passed to `SourceTracker`.
//
class RTC_EXPORT RtpPacketInfo {
public:
@ -102,8 +102,8 @@ class RTC_EXPORT RtpPacketInfo {
// Fields from the Absolute Capture Time header extension:
// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
// To not be confused with |local_capture_clock_offset_|, the
// |estimated_capture_clock_offset| in |absolute_capture_time_| should
// To not be confused with `local_capture_clock_offset_`, the
// `estimated_capture_clock_offset` in `absolute_capture_time_` should
// represent the clock offset between a remote sender and the capturer, and
// thus equals to the corresponding values in the received RTP packets,
// subjected to possible interpolations.

8
api/rtp_parameters.h
View file

@ -126,7 +126,7 @@ struct RTC_EXPORT RtpCodecCapability {
RtpCodecCapability();
~RtpCodecCapability();
// Build MIME "type/subtype" string from |name| and |kind|.
// Build MIME "type/subtype" string from `name` and `kind`.
std::string mime_type() const { return MediaTypeToString(kind) + "/" + name; }
// Used to identify the codec. Equivalent to MIME subtype.
@ -537,7 +537,7 @@ struct RTC_EXPORT RtpCodecParameters {
RtpCodecParameters(const RtpCodecParameters&);
~RtpCodecParameters();
// Build MIME "type/subtype" string from |name| and |kind|.
// Build MIME "type/subtype" string from `name` and `kind`.
std::string mime_type() const { return MediaTypeToString(kind) + "/" + name; }
// Used to identify the codec. Equivalent to MIME subtype.
@ -562,7 +562,7 @@ struct RTC_EXPORT RtpCodecParameters {
absl::optional<int> num_channels;
// The maximum packetization time to be used by an RtpSender.
// If |ptime| is also set, this will be ignored.
// If `ptime` is also set, this will be ignored.
// TODO(deadbeef): Not implemented.
absl::optional<int> max_ptime;
@ -607,7 +607,7 @@ struct RTC_EXPORT RtpCapabilities {
// Supported Forward Error Correction (FEC) mechanisms. Note that the RED,
// ulpfec and flexfec codecs used by these mechanisms will still appear in
// |codecs|.
// `codecs`.
std::vector<FecMechanism> fec;
bool operator==(const RtpCapabilities& o) const {

6
api/rtp_receiver_interface.h
View file

@ -54,7 +54,7 @@ class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface {
// TODO(https://bugs.webrtc.org/907849) remove default implementation
virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
// The list of streams that |track| is associated with. This is the same as
// The list of streams that `track` is associated with. This is the same as
// the [[AssociatedRemoteMediaStreams]] internal slot in the spec.
// https://w3c.github.io/webrtc-pc/#dfn-associatedremotemediastreams
// TODO(hbos): Make pure virtual as soon as Chromium's mock implements this.
@ -84,8 +84,8 @@ class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface {
virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
// Sets the jitter buffer minimum delay until media playout. Actual observed
// delay may differ depending on the congestion control. |delay_seconds| is a
// positive value including 0.0 measured in seconds. |nullopt| means default
// delay may differ depending on the congestion control. `delay_seconds` is a
// positive value including 0.0 measured in seconds. `nullopt` means default
// value must be used.
virtual void SetJitterBufferMinimumDelay(
absl::optional<double> delay_seconds) = 0;

10
api/scoped_refptr.h
View file

@ -24,13 +24,13 @@
// void some_function() {
// scoped_refptr<MyFoo> foo = new MyFoo();
// foo->Method(param);
// // |foo| is released when this function returns
// // `foo` is released when this function returns
// }
//
// void some_other_function() {
// scoped_refptr<MyFoo> foo = new MyFoo();
// ...
// foo = nullptr; // explicitly releases |foo|
// foo = nullptr; // explicitly releases `foo`
// ...
// if (foo)
// foo->Method(param);
@ -45,10 +45,10 @@
// scoped_refptr<MyFoo> b;
//
// b.swap(a);
// // now, |b| references the MyFoo object, and |a| references null.
// // now, `b` references the MyFoo object, and `a` references null.
// }
//
// To make both |a| and |b| in the above example reference the same MyFoo
// To make both `a` and `b` in the above example reference the same MyFoo
// object, simply use the assignment operator:
//
// {
@ -56,7 +56,7 @@
// scoped_refptr<MyFoo> b;
//
// b = a;
// // now, |a| and |b| each own a reference to the same MyFoo object.
// // now, `a` and `b` each own a reference to the same MyFoo object.
// }
//

42
api/stats/rtc_stats.h
View file

@ -35,7 +35,7 @@ class RTCStatsMemberInterface;
// static const char kType[];
// It is used as a unique class identifier and a string representation of the
// class type, see https://w3c.github.io/webrtc-stats/#rtcstatstype-str*.
// Use the |WEBRTC_RTCSTATS_IMPL| macro when implementing subclasses, see macro
// Use the `WEBRTC_RTCSTATS_IMPL` macro when implementing subclasses, see macro
// for details.
//
// Derived classes list their dictionary members, RTCStatsMember<T>, as public
@ -47,7 +47,7 @@ class RTCStatsMemberInterface;
// foo.baz->push_back("hello world");
// uint32_t x = *foo.bar;
//
// Pointers to all the members are available with |Members|, allowing iteration:
// Pointers to all the members are available with `Members`, allowing iteration:
//
// for (const RTCStatsMemberInterface* member : foo.Members()) {
// printf("%s = %s\n", member->name(), member->ValueToString().c_str());
@ -65,11 +65,11 @@ class RTC_EXPORT RTCStats {
const std::string& id() const { return id_; }
// Time relative to the UNIX epoch (Jan 1, 1970, UTC), in microseconds.
int64_t timestamp_us() const { return timestamp_us_; }
// Returns the static member variable |kType| of the implementing class.
// Returns the static member variable `kType` of the implementing class.
virtual const char* type() const = 0;
// Returns a vector of pointers to all the |RTCStatsMemberInterface| members
// Returns a vector of pointers to all the `RTCStatsMemberInterface` members
// of this class. This allows for iteration of members. For a given class,
// |Members| always returns the same members in the same order.
// `Members` always returns the same members in the same order.
std::vector<const RTCStatsMemberInterface*> Members() const;
// Checks if the two stats objects are of the same type and have the same
// member values. Timestamps are not compared. These operators are exposed for
@ -81,8 +81,8 @@ class RTC_EXPORT RTCStats {
// object, listing all of its members (names and values).
std::string ToJson() const;
// Downcasts the stats object to an |RTCStats| subclass |T|. DCHECKs that the
// object is of type |T|.
// Downcasts the stats object to an `RTCStats` subclass `T`. DCHECKs that the
// object is of type `T`.
template <typename T>
const T& cast_to() const {
RTC_DCHECK_EQ(type(), T::kType);
@ -90,8 +90,8 @@ class RTC_EXPORT RTCStats {
}
protected:
// Gets a vector of all members of this |RTCStats| object, including members
// derived from parent classes. |additional_capacity| is how many more members
// Gets a vector of all members of this `RTCStats` object, including members
// derived from parent classes. `additional_capacity` is how many more members
// shall be reserved in the vector (so that subclasses can allocate a vector
// with room for both parent and child members without it having to resize).
virtual std::vector<const RTCStatsMemberInterface*>
@ -101,21 +101,21 @@ class RTC_EXPORT RTCStats {
int64_t timestamp_us_;
};
// All |RTCStats| classes should use these macros.
// |WEBRTC_RTCSTATS_DECL| is placed in a public section of the class definition.
// |WEBRTC_RTCSTATS_IMPL| is placed outside the class definition (in a .cc).
// All `RTCStats` classes should use these macros.
// `WEBRTC_RTCSTATS_DECL` is placed in a public section of the class definition.
// `WEBRTC_RTCSTATS_IMPL` is placed outside the class definition (in a .cc).
//
// These macros declare (in _DECL) and define (in _IMPL) the static |kType| and
// overrides methods as required by subclasses of |RTCStats|: |copy|, |type| and
// |MembersOfThisObjectAndAncestors|. The |...| argument is a list of addresses
// These macros declare (in _DECL) and define (in _IMPL) the static `kType` and
// overrides methods as required by subclasses of `RTCStats`: `copy`, `type` and
// `MembersOfThisObjectAndAncestors`. The |...| argument is a list of addresses
// to each member defined in the implementing class. The list must have at least
// one member.
//
// (Since class names need to be known to implement these methods this cannot be
// part of the base |RTCStats|. While these methods could be implemented using
// part of the base `RTCStats`. While these methods could be implemented using
// templates, that would only work for immediate subclasses. Subclasses of
// subclasses also have to override these methods, resulting in boilerplate
// code. Using a macro avoids this and works for any |RTCStats| class, including
// code. Using a macro avoids this and works for any `RTCStats` class, including
// grandchildren.)
//
// Sample usage:
@ -215,10 +215,10 @@ enum class NonStandardGroupId {
kRtcStatsRelativePacketArrivalDelay,
};
// Interface for |RTCStats| members, which have a name and a value of a type
// defined in a subclass. Only the types listed in |Type| are supported, these
// Interface for `RTCStats` members, which have a name and a value of a type
// defined in a subclass. Only the types listed in `Type` are supported, these
// are implemented by |RTCStatsMember<T>|. The value of a member may be
// undefined, the value can only be read if |is_defined|.
// undefined, the value can only be read if `is_defined`.
class RTCStatsMemberInterface {
public:
// Member value types.
@ -284,7 +284,7 @@ class RTCStatsMemberInterface {
bool is_defined_;
};
// Template implementation of |RTCStatsMemberInterface|.
// Template implementation of `RTCStatsMemberInterface`.
// The supported types are the ones described by
// |RTCStatsMemberInterface::Type|.
template <typename T>

14
api/stats/rtc_stats_report.h
View file

@ -30,7 +30,7 @@
namespace webrtc {
// A collection of stats.
// This is accessible as a map from |RTCStats::id| to |RTCStats|.
// This is accessible as a map from `RTCStats::id` to `RTCStats`.
class RTC_EXPORT RTCStatsReport final
: public rtc::RefCountedNonVirtual<RTCStatsReport> {
public:
@ -71,8 +71,8 @@ class RTC_EXPORT RTCStatsReport final
const RTCStats* Get(const std::string& id) const;
size_t size() const { return stats_.size(); }
// Gets the stat object of type |T| by ID, where |T| is any class descending
// from |RTCStats|.
// Gets the stat object of type `T` by ID, where `T` is any class descending
// from `RTCStats`.
// Returns null if there is no stats object for the given ID or it is the
// wrong type.
template <typename T>
@ -85,17 +85,17 @@ class RTC_EXPORT RTCStatsReport final
}
// Removes the stats object from the report, returning ownership of it or null
// if there is no object with |id|.
// if there is no object with `id`.
std::unique_ptr<const RTCStats> Take(const std::string& id);
// Takes ownership of all the stats in |other|, leaving it empty.
// Takes ownership of all the stats in `other`, leaving it empty.
void TakeMembersFrom(rtc::scoped_refptr<RTCStatsReport> other);
// Stats iterators. Stats are ordered lexicographically on |RTCStats::id|.
ConstIterator begin() const;
ConstIterator end() const;
// Gets the subset of stats that are of type |T|, where |T| is any class
// descending from |RTCStats|.
// Gets the subset of stats that are of type `T`, where `T` is any class
// descending from `RTCStats`.
template <typename T>
std::vector<const T*> GetStatsOfType() const {
std::vector<const T*> stats_of_type;

20
api/stats/rtcstats_objects.h
View file

@ -197,7 +197,7 @@ class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats {
};
// https://w3c.github.io/webrtc-stats/#icecandidate-dict*
// TODO(hbos): |RTCStatsCollector| only collects candidates that are part of
// TODO(hbos): `RTCStatsCollector` only collects candidates that are part of
// ice candidate pairs, but there could be candidates not paired with anything.
// crbug.com/632723
// TODO(qingsi): Add the stats of STUN binding requests (keepalives) and collect
@ -221,7 +221,7 @@ class RTC_EXPORT RTCIceCandidateStats : public RTCStats {
// TODO(hbos): Support enum types? "RTCStatsMember<RTCIceCandidateType>"?
RTCStatsMember<std::string> candidate_type;
RTCStatsMember<int32_t> priority;
// TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/632723
// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/632723
RTCStatsMember<std::string> url;
protected:
@ -232,8 +232,8 @@ class RTC_EXPORT RTCIceCandidateStats : public RTCStats {
};
// In the spec both local and remote varieties are of type RTCIceCandidateStats.
// But here we define them as subclasses of |RTCIceCandidateStats| because the
// |kType| need to be different ("RTCStatsType type") in the local/remote case.
// But here we define them as subclasses of `RTCIceCandidateStats` because the
// `kType` need to be different ("RTCStatsType type") in the local/remote case.
// https://w3c.github.io/webrtc-stats/#rtcstatstype-str*
// This forces us to have to override copy() and type().
class RTC_EXPORT RTCLocalIceCandidateStats final : public RTCIceCandidateStats {
@ -289,28 +289,28 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
RTCStatsMember<std::string> media_source_id;
RTCStatsMember<bool> remote_source;
RTCStatsMember<bool> ended;
// TODO(hbos): |RTCStatsCollector| does not return stats for detached tracks.
// TODO(hbos): `RTCStatsCollector` does not return stats for detached tracks.
// crbug.com/659137
RTCStatsMember<bool> detached;
// See |RTCMediaStreamTrackKind| for valid values.
// See `RTCMediaStreamTrackKind` for valid values.
RTCStatsMember<std::string> kind;
RTCStatsMember<double> jitter_buffer_delay;
RTCStatsMember<uint64_t> jitter_buffer_emitted_count;
// Video-only members
RTCStatsMember<uint32_t> frame_width;
RTCStatsMember<uint32_t> frame_height;
// TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
RTCStatsMember<double> frames_per_second;
RTCStatsMember<uint32_t> frames_sent;
RTCStatsMember<uint32_t> huge_frames_sent;
RTCStatsMember<uint32_t> frames_received;
RTCStatsMember<uint32_t> frames_decoded;
RTCStatsMember<uint32_t> frames_dropped;
// TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
RTCStatsMember<uint32_t> frames_corrupted;
// TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
RTCStatsMember<uint32_t> partial_frames_lost;
// TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
RTCStatsMember<uint32_t> full_frames_lost;
// Audio-only members
RTCStatsMember<double> audio_level; // Receive-only

2
api/stats_types.cc
View file

@ -835,7 +835,7 @@ StatsReport* StatsCollection::ReplaceOrAddNew(const StatsReport::Id& id) {
return InsertNew(id);
}
// Looks for a report with the given |id|. If one is not found, null
// Looks for a report with the given `id`. If one is not found, null
// will be returned.
StatsReport* StatsCollection::Find(const StatsReport::Id& id) {
RTC_DCHECK(thread_checker_.IsCurrent());

40
api/stats_types.h
View file

@ -39,58 +39,58 @@ class RTC_EXPORT StatsReport {
enum StatsType {
// StatsReport types.
// A StatsReport of |type| = "googSession" contains overall information
// A StatsReport of `type` = "googSession" contains overall information
// about the thing libjingle calls a session (which may contain one
// or more RTP sessions.
kStatsReportTypeSession,
// A StatsReport of |type| = "googTransport" contains information
// A StatsReport of `type` = "googTransport" contains information
// about a libjingle "transport".
kStatsReportTypeTransport,
// A StatsReport of |type| = "googComponent" contains information
// A StatsReport of `type` = "googComponent" contains information
// about a libjingle "channel" (typically, RTP or RTCP for a transport).
// This is intended to be the same thing as an ICE "Component".
kStatsReportTypeComponent,
// A StatsReport of |type| = "googCandidatePair" contains information
// A StatsReport of `type` = "googCandidatePair" contains information
// about a libjingle "connection" - a single source/destination port pair.
// This is intended to be the same thing as an ICE "candidate pair".
kStatsReportTypeCandidatePair,
// A StatsReport of |type| = "VideoBWE" is statistics for video Bandwidth
// Estimation, which is global per-session. The |id| field is "bweforvideo"
// A StatsReport of `type` = "VideoBWE" is statistics for video Bandwidth
// Estimation, which is global per-session. The `id` field is "bweforvideo"
// (will probably change in the future).
kStatsReportTypeBwe,
// A StatsReport of |type| = "ssrc" is statistics for a specific rtp stream.
// The |id| field is the SSRC in decimal form of the rtp stream.
// A StatsReport of `type` = "ssrc" is statistics for a specific rtp stream.
// The `id` field is the SSRC in decimal form of the rtp stream.
kStatsReportTypeSsrc,
// A StatsReport of |type| = "remoteSsrc" is statistics for a specific
// A StatsReport of `type` = "remoteSsrc" is statistics for a specific
// rtp stream, generated by the remote end of the connection.
kStatsReportTypeRemoteSsrc,
// A StatsReport of |type| = "googTrack" is statistics for a specific media
// track. The |id| field is the track id.
// A StatsReport of `type` = "googTrack" is statistics for a specific media
// track. The `id` field is the track id.
kStatsReportTypeTrack,
// A StatsReport of |type| = "localcandidate" or "remotecandidate" is
// A StatsReport of `type` = "localcandidate" or "remotecandidate" is
// attributes on a specific ICE Candidate. It links to its connection pair
// by candidate id. The string value is taken from
// http://w3c.github.io/webrtc-stats/#rtcstatstype-enum*.
kStatsReportTypeIceLocalCandidate,
kStatsReportTypeIceRemoteCandidate,
// A StatsReport of |type| = "googCertificate" contains an SSL certificate
// transmitted by one of the endpoints of this connection. The |id| is
// A StatsReport of `type` = "googCertificate" contains an SSL certificate
// transmitted by one of the endpoints of this connection. The `id` is
// controlled by the fingerprint, and is used to identify the certificate in
// the Channel stats (as "googLocalCertificateId" or
// "googRemoteCertificateId") and in any child certificates (as
// "googIssuerId").
kStatsReportTypeCertificate,
// A StatsReport of |type| = "datachannel" with statistics for a
// A StatsReport of `type` = "datachannel" with statistics for a
// particular DataChannel.
kStatsReportTypeDataChannel,
};
@ -331,7 +331,7 @@ class RTC_EXPORT StatsReport {
bool bool_val() const;
const Id& id_val() const;
// Returns the string representation of |name|.
// Returns the string representation of `name`.
const char* display_name() const;
// Converts the native value to a string representation of the value.
@ -339,7 +339,7 @@ class RTC_EXPORT StatsReport {
Type type() const { return type_; }
// TODO(tommi): Move |name| and |display_name| out of the Value struct.
// TODO(tommi): Move `name` and `display_name` out of the Value struct.
const StatsValueName name;
private:
@ -364,7 +364,7 @@ class RTC_EXPORT StatsReport {
typedef rtc::scoped_refptr<Value> ValuePtr;
typedef std::map<StatsValueName, ValuePtr> Values;
// Ownership of |id| is passed to |this|.
// Ownership of `id` is passed to `this`.
explicit StatsReport(const Id& id);
~StatsReport();
@ -434,13 +434,13 @@ class StatsCollection {
const_iterator end() const;
size_t size() const;
// Creates a new report object with |id| that does not already
// Creates a new report object with `id` that does not already
// exist in the list of reports.
StatsReport* InsertNew(const StatsReport::Id& id);
StatsReport* FindOrAddNew(const StatsReport::Id& id);
StatsReport* ReplaceOrAddNew(const StatsReport::Id& id);
// Looks for a report with the given |id|. If one is not found, null
// Looks for a report with the given `id`. If one is not found, null
// will be returned.
StatsReport* Find(const StatsReport::Id& id);

6
api/task_queue/queued_task.h
View file

@ -20,9 +20,9 @@ class QueuedTask {
virtual ~QueuedTask() = default;
// Main routine that will run when the task is executed on the desired queue.
// The task should return |true| to indicate that it should be deleted or
// |false| to indicate that the queue should consider ownership of the task
// having been transferred. Returning |false| can be useful if a task has
// The task should return `true` to indicate that it should be deleted or
// `false` to indicate that the queue should consider ownership of the task
// having been transferred. Returning `false` can be useful if a task has
// re-posted itself to a different queue or is otherwise being re-used.
virtual bool Run() = 0;
};

4
api/task_queue/task_queue_test.cc
View file

@ -37,7 +37,7 @@ TEST_P(TaskQueueTest, PostAndCheckCurrent) {
rtc::Event event;
auto queue = CreateTaskQueue(factory, "PostAndCheckCurrent");
// We're not running a task, so |queue| shouldn't be current.
// We're not running a task, so `queue` shouldn't be current.
// Note that because rtc::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.
@ -190,7 +190,7 @@ TEST_P(TaskQueueTest, PostAndReuse) {
}
TEST_P(TaskQueueTest, PostALot) {
// Waits until DecrementCount called |count| times. Thread safe.
// Waits until DecrementCount called `count` times. Thread safe.
class BlockingCounter {
public:
explicit BlockingCounter(int initial_count) : count_(initial_count) {}

4
api/test/audio_quality_analyzer_interface.h
View file

@ -25,9 +25,9 @@ class AudioQualityAnalyzerInterface : public StatsObserverInterface {
~AudioQualityAnalyzerInterface() override = default;
// Will be called by the framework before the test.
// |test_case_name| is name of test case, that should be used to report all
// `test_case_name` is name of test case, that should be used to report all
// audio metrics.
// |analyzer_helper| is a pointer to a class that will allow track_id to
// `analyzer_helper` is a pointer to a class that will allow track_id to
// stream_id matching. The caller is responsible for ensuring the
// AnalyzerHelper outlives the instance of the AudioQualityAnalyzerInterface.
virtual void Start(std::string test_case_name,

14
api/test/audioproc_float.h
View file

@ -23,12 +23,12 @@ namespace test {
// utility can be used to simulate the audioprocessing module using a recording
// (either an AEC dump or wav files), and generate the output as a wav file.
// Any audio_processing object specified in the input is used for the
// simulation. The optional |audio_processing| object provides the
// simulation. The optional `audio_processing` object provides the
// AudioProcessing instance that is used during the simulation. Note that when
// the audio_processing object is specified all functionality that relies on
// using the AudioProcessingBuilder is deactivated, since the AudioProcessing
// object is already created and the builder is not used in the simulation. It
// is needed to pass the command line flags as |argc| and |argv|, so these can
// is needed to pass the command line flags as `argc` and `argv`, so these can
// be interpreted properly by the utility. To see a list of all supported
// command line flags, run the executable with the '--help' flag.
int AudioprocFloat(rtc::scoped_refptr<AudioProcessing> audio_processing,
@ -38,10 +38,10 @@ int AudioprocFloat(rtc::scoped_refptr<AudioProcessing> audio_processing,
// This is an interface for the audio processing simulation utility. This
// utility can be used to simulate the audioprocessing module using a recording
// (either an AEC dump or wav files), and generate the output as a wav file.
// The |ap_builder| object will be used to create the AudioProcessing instance
// that is used during the simulation. The |ap_builder| supports setting of
// The `ap_builder` object will be used to create the AudioProcessing instance
// that is used during the simulation. The `ap_builder` supports setting of
// injectable components, which will be passed on to the created AudioProcessing
// instance. It is needed to pass the command line flags as |argc| and |argv|,
// instance. It is needed to pass the command line flags as `argc` and `argv`,
// so these can be interpreted properly by the utility.
// To get a fully-working audioproc_f utility, all that is needed is to write a
// main function, create an AudioProcessingBuilder, optionally set custom
@ -56,9 +56,9 @@ int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
// Interface for the audio processing simulation utility, which is similar to
// the one above, but which adds the option of receiving the input as a string
// and returning the output as an array. The first three arguments fulfill the
// same purpose as above. Pass the |input_aecdump| to provide the content of an
// same purpose as above. Pass the `input_aecdump` to provide the content of an
// AEC dump file as a string. After the simulation is completed,
// |processed_capture_samples| will contain the the samples processed on the
// `processed_capture_samples` will contain the the samples processed on the
// capture side.
int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
int argc,

6
api/test/create_frame_generator.h
View file

@ -24,8 +24,8 @@ namespace test {
// Creates a frame generator that produces frames with small squares that
// move randomly towards the lower right corner.
// |type| has the default value FrameGeneratorInterface::OutputType::I420.
// |num_squares| has the default value 10.
// `type` has the default value FrameGeneratorInterface::OutputType::I420.
// `num_squares` has the default value 10.
std::unique_ptr<FrameGeneratorInterface> CreateSquareFrameGenerator(
int width,
int height,
@ -66,7 +66,7 @@ CreateScrollingInputFromYuvFilesFrameGenerator(
// Creates a frame generator that produces randomly generated slides. It fills
// the frames with randomly sized and colored squares.
// |frame_repeat_count| determines how many times each slide is shown.
// `frame_repeat_count` determines how many times each slide is shown.
std::unique_ptr<FrameGeneratorInterface>
CreateSlideFrameGenerator(int width, int height, int frame_repeat_count);

2
api/test/create_peer_connection_quality_test_frame_generator.cc
View file

@ -30,7 +30,7 @@ void ValidateScreenShareConfig(const VideoConfig& video_config,
const ScreenShareConfig& screen_share_config) {
if (screen_share_config.slides_yuv_file_names.empty()) {
if (screen_share_config.scrolling_params) {
// If we have scrolling params, then its |source_width| and |source_heigh|
// If we have scrolling params, then its `source_width` and `source_heigh`
// will be used as width and height of video input, so we have to validate
// it against width and height of default input.
RTC_CHECK_EQ(screen_share_config.scrolling_params->source_width,

2
api/test/create_peer_connection_quality_test_frame_generator.h
View file

@ -21,7 +21,7 @@ namespace webrtc {
namespace webrtc_pc_e2e {
// Creates a frame generator that produces frames with small squares that move
// randomly towards the lower right corner. |type| has the default value
// randomly towards the lower right corner. `type` has the default value
// FrameGeneratorInterface::OutputType::I420. video_config specifies frame
// weight and height.
std::unique_ptr<test::FrameGeneratorInterface> CreateSquareFrameGenerator(

6
api/test/create_peerconnection_quality_test_fixture.h
View file

@ -25,10 +25,10 @@ namespace webrtc_pc_e2e {
// Create test fixture to establish test call between Alice and Bob.
// During the test Alice will be caller and Bob will answer the call.
// |test_case_name| is a name of test case, that will be used for all metrics
// `test_case_name` is a name of test case, that will be used for all metrics
// reporting.
// |time_controller| is used to manage all rtc::Thread's and TaskQueue
// instances. Instance of |time_controller| have to outlive created fixture.
// `time_controller` is used to manage all rtc::Thread's and TaskQueue
// instances. Instance of `time_controller` have to outlive created fixture.
// Returns a non-null PeerConnectionE2EQualityTestFixture instance.
std::unique_ptr<PeerConnectionE2EQualityTestFixture>
CreatePeerConnectionE2EQualityTestFixture(

2
api/test/create_time_controller.h
View file

@ -17,7 +17,7 @@
namespace webrtc {
// Creates a time coltroller that wraps |alarm|.
// Creates a time coltroller that wraps `alarm`.
std::unique_ptr<TimeController> CreateTimeController(
ControlledAlarmClock* alarm);

4
api/test/network_emulation/cross_traffic.h
View file

@ -27,12 +27,12 @@ class CrossTrafficRoute {
public:
virtual ~CrossTrafficRoute() = default;
// Triggers sending of dummy packets with size |packet_size| bytes.
// Triggers sending of dummy packets with size `packet_size` bytes.
virtual void TriggerPacketBurst(size_t num_packets, size_t packet_size) = 0;
// Sends a packet over the nodes. The content of the packet is unspecified;
// only the size metter for the emulation purposes.
virtual void SendPacket(size_t packet_size) = 0;
// Sends a packet over the nodes and runs |action| when it has been delivered.
// Sends a packet over the nodes and runs `action` when it has been delivered.
virtual void NetworkDelayedAction(size_t packet_size,
std::function<void()> action) = 0;
};

12
api/test/network_emulation/network_emulation_interfaces.h
View file

@ -204,9 +204,9 @@ class EmulatedNetworkStats {
class EmulatedEndpoint : public EmulatedNetworkReceiverInterface {
public:
// Send packet into network.
// |from| will be used to set source address for the packet in destination
// `from` will be used to set source address for the packet in destination
// socket.
// |to| will be used for routing verification and picking right socket by port
// `to` will be used for routing verification and picking right socket by port
// on destination endpoint.
virtual void SendPacket(const rtc::SocketAddress& from,
const rtc::SocketAddress& to,
@ -214,12 +214,12 @@ class EmulatedEndpoint : public EmulatedNetworkReceiverInterface {
uint16_t application_overhead = 0) = 0;
// Binds receiver to this endpoint to send and receive data.
// |desired_port| is a port that should be used. If it is equal to 0,
// `desired_port` is a port that should be used. If it is equal to 0,
// endpoint will pick the first available port starting from
// |kFirstEphemeralPort|.
// `kFirstEphemeralPort`.
//
// Returns the port, that should be used (it will be equals to desired, if
// |desired_port| != 0 and is free or will be the one, selected by endpoint)
// `desired_port` != 0 and is free or will be the one, selected by endpoint)
// or absl::nullopt if desired_port in used. Also fails if there are no more
// free ports to bind to.
//
@ -256,7 +256,7 @@ class EmulatedEndpoint : public EmulatedNetworkReceiverInterface {
// they are guranteed to be delivered eventually, even on lossy networks.
class TcpMessageRoute {
public:
// Sends a TCP message of the given |size| over the route, |on_received| is
// Sends a TCP message of the given `size` over the route, `on_received` is
// called when the message has been delivered. Note that the connection
// parameters are reset iff there's no currently pending message on the route.
virtual void SendMessage(size_t size, std::function<void()> on_received) = 0;

32
api/test/network_emulation_manager.h
View file

@ -130,7 +130,7 @@ class EmulatedNetworkManagerInterface {
virtual std::vector<EmulatedEndpoint*> endpoints() const = 0;
// Passes summarized network stats for endpoints for this manager into
// specified |stats_callback|. Callback will be executed on network emulation
// specified `stats_callback`. Callback will be executed on network emulation
// internal task queue.
virtual void GetStats(
std::function<void(std::unique_ptr<EmulatedNetworkStats>)> stats_callback)
@ -180,13 +180,13 @@ class NetworkEmulationManager {
// Creates an emulated network node, which represents single network in
// the emulated network layer. Uses default implementation on network behavior
// which can be configured with |config|. |random_seed| can be provided to
// which can be configured with `config`. `random_seed` can be provided to
// alter randomization behavior.
virtual EmulatedNetworkNode* CreateEmulatedNode(
BuiltInNetworkBehaviorConfig config,
uint64_t random_seed = 1) = 0;
// Creates an emulated network node, which represents single network in
// the emulated network layer. |network_behavior| determines how created node
// the emulated network layer. `network_behavior` determines how created node
// will forward incoming packets to the next receiver.
virtual EmulatedNetworkNode* CreateEmulatedNode(
std::unique_ptr<NetworkBehaviorInterface> network_behavior) = 0;
@ -205,8 +205,8 @@ class NetworkEmulationManager {
// Creates a route between endpoints going through specified network nodes.
// This route is single direction only and describe how traffic that was
// sent by network interface |from| have to be delivered to the network
// interface |to|. Return object can be used to remove created route. The
// sent by network interface `from` have to be delivered to the network
// interface `to`. Return object can be used to remove created route. The
// route must contains at least one network node inside it.
//
// Assume that E{0-9} are endpoints and N{0-9} are network nodes, then
@ -228,7 +228,7 @@ class NetworkEmulationManager {
const std::vector<EmulatedNetworkNode*>& via_nodes,
EmulatedEndpoint* to) = 0;
// Creates a route over the given |via_nodes| creating the required endpoints
// Creates a route over the given `via_nodes` creating the required endpoints
// in the process. The returned EmulatedRoute pointer can be used in other
// calls as a transport route for message or cross traffic.
virtual EmulatedRoute* CreateRoute(
@ -239,7 +239,7 @@ class NetworkEmulationManager {
// packet's destination IP.
//
// This route is single direction only and describe how traffic that was
// sent by network interface |from| have to be delivered in case if routing
// sent by network interface `from` have to be delivered in case if routing
// was unspecified. Return object can be used to remove created route. The
// route must contains at least one network node inside it.
//
@ -269,29 +269,29 @@ class NetworkEmulationManager {
// packets being dropped.
virtual void ClearRoute(EmulatedRoute* route) = 0;
// Creates a simulated TCP connection using |send_route| for traffic and
// |ret_route| for feedback. This can be used to emulate HTTP cross traffic
// Creates a simulated TCP connection using `send_route` for traffic and
// `ret_route` for feedback. This can be used to emulate HTTP cross traffic
// and to implement realistic reliable signaling over lossy networks.
// TODO(srte): Handle clearing of the routes involved.
virtual TcpMessageRoute* CreateTcpRoute(EmulatedRoute* send_route,
EmulatedRoute* ret_route) = 0;
// Creates a route over the given |via_nodes|. Returns an object that can be
// Creates a route over the given `via_nodes`. Returns an object that can be
// used to emulate network load with cross traffic over the created route.
virtual CrossTrafficRoute* CreateCrossTrafficRoute(
const std::vector<EmulatedNetworkNode*>& via_nodes) = 0;
// Starts generating cross traffic using given |generator|. Takes ownership
// Starts generating cross traffic using given `generator`. Takes ownership
// over the generator.
virtual CrossTrafficGenerator* StartCrossTraffic(
std::unique_ptr<CrossTrafficGenerator> generator) = 0;
// Stops generating cross traffic that was started using given |generator|.
// The |generator| shouldn't be used after and the reference may be invalid.
// Stops generating cross traffic that was started using given `generator`.
// The `generator` shouldn't be used after and the reference may be invalid.
virtual void StopCrossTraffic(CrossTrafficGenerator* generator) = 0;
// Creates EmulatedNetworkManagerInterface which can be used then to inject
// network emulation layer into PeerConnection. |endpoints| - are available
// network emulation layer into PeerConnection. `endpoints` - are available
// network interfaces for PeerConnection. If endpoint is enabled, it will be
// immediately available for PeerConnection, otherwise user will be able to
// enable endpoint later to make it available for PeerConnection.
@ -299,8 +299,8 @@ class NetworkEmulationManager {
CreateEmulatedNetworkManagerInterface(
const std::vector<EmulatedEndpoint*>& endpoints) = 0;
// Passes summarized network stats for specified |endpoints| into specified
// |stats_callback|. Callback will be executed on network emulation
// Passes summarized network stats for specified `endpoints` into specified
// `stats_callback`. Callback will be executed on network emulation
// internal task queue.
virtual void GetStats(
rtc::ArrayView<EmulatedEndpoint* const> endpoints,

66
api/test/peerconnection_quality_test_fixture.h
View file

@ -67,17 +67,17 @@ class PeerConnectionE2EQualityTestFixture {
// bottom right corner of the picture.
//
// In such case source dimensions must be greater or equal to the sliding
// window dimensions. So |source_width| and |source_height| are the dimensions
// of the source frame, while |VideoConfig::width| and |VideoConfig::height|
// window dimensions. So `source_width` and `source_height` are the dimensions
// of the source frame, while `VideoConfig::width` and `VideoConfig::height`
// are the dimensions of the sliding window.
//
// Because |source_width| and |source_height| are dimensions of the source
// Because `source_width` and `source_height` are dimensions of the source
// frame, they have to be width and height of videos from
// |ScreenShareConfig::slides_yuv_file_names|.
// `ScreenShareConfig::slides_yuv_file_names`.
//
// Because scrolling have to be done on single slide it also requires, that
// |duration| must be less or equal to
// |ScreenShareConfig::slide_change_interval|.
// `duration` must be less or equal to
// `ScreenShareConfig::slide_change_interval`.
struct ScrollingParams {
ScrollingParams(TimeDelta duration,
size_t source_width,
@ -110,16 +110,16 @@ class PeerConnectionE2EQualityTestFixture {
// will be applied in such case.
bool generate_slides = false;
// If present scrolling will be applied. Please read extra requirement on
// |slides_yuv_file_names| for scrolling.
// `slides_yuv_file_names` for scrolling.
absl::optional<ScrollingParams> scrolling_params;
// Contains list of yuv files with slides.
//
// If empty, default set of slides will be used. In such case
// |VideoConfig::width| must be equal to |kDefaultSlidesWidth| and
// |VideoConfig::height| must be equal to |kDefaultSlidesHeight| or if
// |scrolling_params| are specified, then |ScrollingParams::source_width|
// must be equal to |kDefaultSlidesWidth| and
// |ScrollingParams::source_height| must be equal to |kDefaultSlidesHeight|.
// `VideoConfig::width` must be equal to `kDefaultSlidesWidth` and
// `VideoConfig::height` must be equal to `kDefaultSlidesHeight` or if
// `scrolling_params` are specified, then `ScrollingParams::source_width`
// must be equal to `kDefaultSlidesWidth` and
// `ScrollingParams::source_height` must be equal to `kDefaultSlidesHeight`.
std::vector<std::string> slides_yuv_file_names;
};
@ -128,7 +128,7 @@ class PeerConnectionE2EQualityTestFixture {
// SVC support is limited:
// During SVC testing there is no SFU, so framework will try to emulate SFU
// behavior in regular p2p call. Because of it there are such limitations:
// * if |target_spatial_index| is not equal to the highest spatial layer
// * if `target_spatial_index` is not equal to the highest spatial layer
// then no packet/frame drops are allowed.
//
// If there will be any drops, that will affect requested layer, then
@ -154,11 +154,11 @@ class PeerConnectionE2EQualityTestFixture {
// Specifies spatial index of the video stream to analyze.
// There are 2 cases:
// 1. simulcast encoder is used:
// in such case |target_spatial_index| will specify the index of
// in such case `target_spatial_index` will specify the index of
// simulcast stream, that should be analyzed. Other streams will be
// dropped.
// 2. SVC encoder is used:
// in such case |target_spatial_index| will specify the top interesting
// in such case `target_spatial_index` will specify the top interesting
// spatial layer and all layers below, including target one will be
// processed. All layers above target one will be dropped.
// If not specified than whatever stream will be received will be analyzed.
@ -166,8 +166,8 @@ class PeerConnectionE2EQualityTestFixture {
// network.
absl::optional<int> target_spatial_index;
// Encoding parameters per simulcast layer. If not empty, |encoding_params|
// size have to be equal to |simulcast_streams_count|. Will be used to set
// Encoding parameters per simulcast layer. If not empty, `encoding_params`
// size have to be equal to `simulcast_streams_count`. Will be used to set
// transceiver send encoding params for simulcast layers. Applicable only
// for codecs that support simulcast (ex. Vp8) and will be ignored
// otherwise. RtpEncodingParameters::rid may be changed by fixture
@ -220,7 +220,7 @@ class PeerConnectionE2EQualityTestFixture {
// was captured during the test for this video stream on sender side.
// It is useful when generator is used as input.
absl::optional<std::string> input_dump_file_name;
// Used only if |input_dump_file_name| is set. Specifies the module for the
// Used only if `input_dump_file_name` is set. Specifies the module for the
// video frames to be dumped. Modulo equals X means every Xth frame will be
// written to the dump file. The value must be greater than 0.
int input_dump_sampling_modulo = 1;
@ -229,7 +229,7 @@ class PeerConnectionE2EQualityTestFixture {
// output files will be appended with indexes. The produced files contains
// what was rendered for this video stream on receiver side.
absl::optional<std::string> output_dump_file_name;
// Used only if |output_dump_file_name| is set. Specifies the module for the
// Used only if `output_dump_file_name` is set. Specifies the module for the
// video frames to be dumped. Modulo equals X means every Xth frame will be
// written to the dump file. The value must be greater than 0.
int output_dump_sampling_modulo = 1;
@ -282,9 +282,9 @@ class PeerConnectionE2EQualityTestFixture {
std::string name = cricket::kVp8CodecName;
// Map of parameters, that have to be specified on SDP codec. Each parameter
// is described by key and value. Codec parameters will match the specified
// map if and only if for each key from |required_params| there will be
// map if and only if for each key from `required_params` there will be
// a parameter with name equal to this key and parameter value will be equal
// to the value from |required_params| for this key.
// to the value from `required_params` for this key.
// If empty then only name will be used to match the codec.
std::map<std::string, std::string> required_params;
};
@ -351,7 +351,7 @@ class PeerConnectionE2EQualityTestFixture {
CapturingDeviceIndex capturing_device_index) = 0;
// Set the list of video codecs used by the peer during the test. These
// codecs will be negotiated in SDP during offer/answer exchange. The order
// of these codecs during negotiation will be the same as in |video_codecs|.
// of these codecs during negotiation will be the same as in `video_codecs`.
// Codecs have to be available in codecs list provided by peer connection to
// be negotiated. If some of specified codecs won't be found, the test will
// crash.
@ -416,9 +416,9 @@ class PeerConnectionE2EQualityTestFixture {
// Invoked by framework after peer connection factory and peer connection
// itself will be created but before offer/answer exchange will be started.
// |test_case_name| is name of test case, that should be used to report all
// `test_case_name` is name of test case, that should be used to report all
// metrics.
// |reporter_helper| is a pointer to a class that will allow track_id to
// `reporter_helper` is a pointer to a class that will allow track_id to
// stream_id matching. The caller is responsible for ensuring the
// TrackIdStreamInfoMap will be valid from Start() to
// StopAndReportResults().
@ -433,14 +433,14 @@ class PeerConnectionE2EQualityTestFixture {
virtual ~PeerConnectionE2EQualityTestFixture() = default;
// Add activity that will be executed on the best effort at least after
// |target_time_since_start| after call will be set up (after offer/answer
// `target_time_since_start` after call will be set up (after offer/answer
// exchange, ICE gathering will be done and ICE candidates will passed to
// remote side). |func| param is amount of time spent from the call set up.
// remote side). `func` param is amount of time spent from the call set up.
virtual void ExecuteAt(TimeDelta target_time_since_start,
std::function<void(TimeDelta)> func) = 0;
// Add activity that will be executed every |interval| with first execution
// on the best effort at least after |initial_delay_since_start| after call
// will be set up (after all participants will be connected). |func| param is
// Add activity that will be executed every `interval` with first execution
// on the best effort at least after `initial_delay_since_start` after call
// will be set up (after all participants will be connected). `func` param is
// amount of time spent from the call set up.
virtual void ExecuteEvery(TimeDelta initial_delay_since_start,
TimeDelta interval,
@ -452,15 +452,15 @@ class PeerConnectionE2EQualityTestFixture {
// Add a new peer to the call and return an object through which caller
// can configure peer's behavior.
// |network_thread| will be used as network thread for peer's peer connection
// |network_manager| will be used to provide network interfaces for peer's
// `network_thread` will be used as network thread for peer's peer connection
// `network_manager` will be used to provide network interfaces for peer's
// peer connection.
// |configurer| function will be used to configure peer in the call.
// `configurer` function will be used to configure peer in the call.
virtual void AddPeer(rtc::Thread* network_thread,
rtc::NetworkManager* network_manager,
rtc::FunctionView<void(PeerConfigurer*)> configurer) = 0;
// Runs the media quality test, which includes setting up the call with
// configured participants, running it according to provided |run_params| and
// configured participants, running it according to provided `run_params` and
// terminating it properly at the end. During call duration media quality
// metrics are gathered, which are then reported to stdout and (if configured)
// to the json/protobuf output file through the WebRTC perf test results

2
api/test/stats_observer_interface.h
View file

@ -23,7 +23,7 @@ class StatsObserverInterface {
virtual ~StatsObserverInterface() = default;
// Method called when stats reports are available for the PeerConnection
// identified by |pc_label|.
// identified by `pc_label`.
virtual void OnStatsReports(
absl::string_view pc_label,
const rtc::scoped_refptr<const RTCStatsReport>& report) = 0;

14
api/test/time_controller.h
View file

@ -44,7 +44,7 @@ class TimeController {
// Creates a process thread.
virtual std::unique_ptr<ProcessThread> CreateProcessThread(
const char* thread_name) = 0;
// Creates an rtc::Thread instance. If |socket_server| is nullptr, a default
// Creates an rtc::Thread instance. If `socket_server` is nullptr, a default
// noop socket server is created.
// Returned thread is not null and started.
virtual std::unique_ptr<rtc::Thread> CreateThread(
@ -55,12 +55,12 @@ class TimeController {
// thread.
virtual rtc::Thread* GetMainThread() = 0;
// Allow task queues and process threads created by this instance to execute
// for the given |duration|.
// for the given `duration`.
virtual void AdvanceTime(TimeDelta duration) = 0;
// Waits until condition() == true, polling condition() in small time
// intervals.
// Returns true if condition() was evaluated to true before |max_duration|
// Returns true if condition() was evaluated to true before `max_duration`
// elapsed and false otherwise.
bool Wait(const std::function<bool()>& condition,
TimeDelta max_duration = TimeDelta::Seconds(5));
@ -75,17 +75,17 @@ class ControlledAlarmClock {
// Gets a clock that tells the alarm clock's notion of time.
virtual Clock* GetClock() = 0;
// Schedules the alarm to fire at |deadline|.
// An alarm clock only supports one deadline. Calls to |ScheduleAlarmAt| with
// Schedules the alarm to fire at `deadline`.
// An alarm clock only supports one deadline. Calls to `ScheduleAlarmAt` with
// an earlier deadline will reset the alarm to fire earlier.Calls to
// |ScheduleAlarmAt| with a later deadline are ignored. Returns true if the
// `ScheduleAlarmAt` with a later deadline are ignored. Returns true if the
// deadline changed, false otherwise.
virtual bool ScheduleAlarmAt(Timestamp deadline) = 0;
// Sets the callback that should be run when the alarm fires.
virtual void SetCallback(std::function<void()> callback) = 0;
// Waits for |duration| to pass, according to the alarm clock.
// Waits for `duration` to pass, according to the alarm clock.
virtual void Sleep(TimeDelta duration) = 0;
};

6
api/test/track_id_stream_info_map.h
View file

@ -16,7 +16,7 @@
namespace webrtc {
namespace webrtc_pc_e2e {
// Instances of |TrackIdStreamInfoMap| provide bookkeeping capabilities that
// Instances of `TrackIdStreamInfoMap` provide bookkeeping capabilities that
// are useful to associate stats reports track_ids to the remote stream info.
class TrackIdStreamInfoMap {
public:
@ -26,12 +26,12 @@ class TrackIdStreamInfoMap {
// StatsObserverInterface::OnStatsReports is invoked.
// Returns a reference to a stream label owned by the TrackIdStreamInfoMap.
// Precondition: |track_id| must be already mapped to stream label.
// Precondition: `track_id` must be already mapped to stream label.
virtual absl::string_view GetStreamLabelFromTrackId(
absl::string_view track_id) const = 0;
// Returns a reference to a sync group name owned by the TrackIdStreamInfoMap.
// Precondition: |track_id| must be already mapped to sync group.
// Precondition: `track_id` must be already mapped to sync group.
virtual absl::string_view GetSyncGroupLabelFromTrackId(
absl::string_view track_id) const = 0;
};

24
api/test/video_quality_analyzer_interface.h
View file

@ -72,9 +72,9 @@ class VideoQualityAnalyzerInterface : public StatsObserverInterface {
~VideoQualityAnalyzerInterface() override = default;
// Will be called by framework before test.
// |test_case_name| is name of test case, that should be used to report all
// `test_case_name` is name of test case, that should be used to report all
// video metrics.
// |threads_count| is number of threads that analyzer can use for heavy
// `threads_count` is number of threads that analyzer can use for heavy
// calculations. Analyzer can perform simple calculations on the calling
// thread in each method, but should remember, that it is the same thread,
// that is used in video pipeline.
@ -83,57 +83,57 @@ class VideoQualityAnalyzerInterface : public StatsObserverInterface {
int max_threads_count) {}
// Will be called when frame was generated from the input stream.
// |peer_name| is name of the peer on which side frame was captured.
// `peer_name` is name of the peer on which side frame was captured.
// Returns frame id, that will be set by framework to the frame.
virtual uint16_t OnFrameCaptured(absl::string_view peer_name,
const std::string& stream_label,
const VideoFrame& frame) = 0;
// Will be called before calling the encoder.
// |peer_name| is name of the peer on which side frame came to encoder.
// `peer_name` is name of the peer on which side frame came to encoder.
virtual void OnFramePreEncode(absl::string_view peer_name,
const VideoFrame& frame) {}
// Will be called for each EncodedImage received from encoder. Single
// VideoFrame can produce multiple EncodedImages. Each encoded image will
// have id from VideoFrame.
// |peer_name| is name of the peer on which side frame was encoded.
// `peer_name` is name of the peer on which side frame was encoded.
virtual void OnFrameEncoded(absl::string_view peer_name,
uint16_t frame_id,
const EncodedImage& encoded_image,
const EncoderStats& stats) {}
// Will be called for each frame dropped by encoder.
// |peer_name| is name of the peer on which side frame drop was detected.
// `peer_name` is name of the peer on which side frame drop was detected.
virtual void OnFrameDropped(absl::string_view peer_name,
EncodedImageCallback::DropReason reason) {}
// Will be called before calling the decoder.
// |peer_name| is name of the peer on which side frame was received.
// `peer_name` is name of the peer on which side frame was received.
virtual void OnFramePreDecode(absl::string_view peer_name,
uint16_t frame_id,
const EncodedImage& encoded_image) {}
// Will be called after decoding the frame.
// |peer_name| is name of the peer on which side frame was decoded.
// `peer_name` is name of the peer on which side frame was decoded.
virtual void OnFrameDecoded(absl::string_view peer_name,
const VideoFrame& frame,
const DecoderStats& stats) {}
// Will be called when frame will be obtained from PeerConnection stack.
// |peer_name| is name of the peer on which side frame was rendered.
// `peer_name` is name of the peer on which side frame was rendered.
virtual void OnFrameRendered(absl::string_view peer_name,
const VideoFrame& frame) {}
// Will be called if encoder return not WEBRTC_VIDEO_CODEC_OK.
// All available codes are listed in
// modules/video_coding/include/video_error_codes.h
// |peer_name| is name of the peer on which side error acquired.
// `peer_name` is name of the peer on which side error acquired.
virtual void OnEncoderError(absl::string_view peer_name,
const VideoFrame& frame,
int32_t error_code) {}
// Will be called if decoder return not WEBRTC_VIDEO_CODEC_OK.
// All available codes are listed in
// modules/video_coding/include/video_error_codes.h
// |peer_name| is name of the peer on which side error acquired.
// `peer_name` is name of the peer on which side error acquired.
virtual void OnDecoderError(absl::string_view peer_name,
uint16_t frame_id,
int32_t error_code) {}
// Will be called every time new stats reports are available for the
// Peer Connection identified by |pc_label|.
// Peer Connection identified by `pc_label`.
void OnStatsReports(
absl::string_view pc_label,
const rtc::scoped_refptr<const RTCStatsReport>& report) override {}

2
api/test/video_quality_test_fixture.h
View file

@ -98,7 +98,7 @@ class VideoQualityTestFixtureInterface {
InterLayerPredMode inter_layer_pred = InterLayerPredMode::kOn;
// If empty, bitrates are generated in VP9Impl automatically.
std::vector<SpatialLayer> spatial_layers;
// If set, default parameters will be used instead of |streams|.
// If set, default parameters will be used instead of `streams`.
bool infer_streams = false;
} ss[2];
struct Logging {

2
api/transport/bitrate_settings.h
View file

@ -18,7 +18,7 @@
namespace webrtc {
// Configuration of send bitrate. The |start_bitrate_bps| value is
// Configuration of send bitrate. The `start_bitrate_bps` value is
// used for multiple purposes, both as a prior in the bandwidth
// estimator, and for initial configuration of the encoder. We may
// want to create separate apis for those, and use a smaller struct

12
api/transport/data_channel_transport_interface.h
View file

@ -48,14 +48,14 @@ struct SendDataParams {
// retransmitted by the transport before it is dropped.
// Setting this value to zero disables retransmission.
// Valid values are in the range [0-UINT16_MAX].
// |max_rtx_count| and |max_rtx_ms| may not be set simultaneously.
// `max_rtx_count` and `max_rtx_ms` may not be set simultaneously.
absl::optional<int> max_rtx_count;
// If set, the maximum number of milliseconds for which the transport
// may retransmit this message before it is dropped.
// Setting this value to zero disables retransmission.
// Valid values are in the range [0-UINT16_MAX].
// |max_rtx_count| and |max_rtx_ms| may not be set simultaneously.
// `max_rtx_count` and `max_rtx_ms` may not be set simultaneously.
absl::optional<int> max_rtx_ms;
};
@ -96,18 +96,18 @@ class DataChannelTransportInterface {
public:
virtual ~DataChannelTransportInterface() = default;
// Opens a data |channel_id| for sending. May return an error if the
// specified |channel_id| is unusable. Must be called before |SendData|.
// Opens a data `channel_id` for sending. May return an error if the
// specified `channel_id` is unusable. Must be called before `SendData`.
virtual RTCError OpenChannel(int channel_id) = 0;
// Sends a data buffer to the remote endpoint using the given send parameters.
// |buffer| may not be larger than 256 KiB. Returns an error if the send
// `buffer` may not be larger than 256 KiB. Returns an error if the send
// fails.
virtual RTCError SendData(int channel_id,
const SendDataParams& params,
const rtc::CopyOnWriteBuffer& buffer) = 0;
// Closes |channel_id| gracefully. Returns an error if |channel_id| is not
// Closes `channel_id` gracefully. Returns an error if `channel_id` is not
// open. Data sent after the closing procedure begins will not be
// transmitted. The channel becomes closed after pending data is transmitted.
virtual RTCError CloseChannel(int channel_id) = 0;

2
api/transport/sctp_transport_factory_interface.h
View file

@ -32,7 +32,7 @@ class SctpTransportFactoryInterface {
public:
virtual ~SctpTransportFactoryInterface() = default;
// Create an SCTP transport using |channel| for the underlying transport.
// Create an SCTP transport using `channel` for the underlying transport.
virtual std::unique_ptr<cricket::SctpTransportInternal> CreateSctpTransport(
rtc::PacketTransportInternal* channel) = 0;
};

14
api/transport/stun.h
View file

@ -254,11 +254,11 @@ class StunMessage {
// This is used for testing.
void SetStunMagicCookie(uint32_t val);
// Contruct a copy of |this|.
// Contruct a copy of `this`.
std::unique_ptr<StunMessage> Clone() const;
// Check if the attributes of this StunMessage equals those of |other|
// for all attributes that |attribute_type_mask| return true
// Check if the attributes of this StunMessage equals those of `other`
// for all attributes that `attribute_type_mask` return true
bool EqualAttributes(const StunMessage* other,
std::function<bool(int type)> attribute_type_mask) const;
@ -570,11 +570,11 @@ class StunUInt16ListAttribute : public StunAttribute {
std::string StunMethodToString(int msg_type);
// Returns the (successful) response type for the given request type.
// Returns -1 if |request_type| is not a valid request type.
// Returns -1 if `request_type` is not a valid request type.
int GetStunSuccessResponseType(int request_type);
// Returns the error response type for the given request type.
// Returns -1 if |request_type| is not a valid request type.
// Returns -1 if `request_type` is not a valid request type.
int GetStunErrorResponseType(int request_type);
// Returns whether a given message is a request type.
@ -595,13 +595,13 @@ bool ComputeStunCredentialHash(const std::string& username,
const std::string& password,
std::string* hash);
// Make a copy af |attribute| and return a new StunAttribute.
// Make a copy af `attribute` and return a new StunAttribute.
// This is useful if you don't care about what kind of attribute you
// are handling.
//
// The implementation copies by calling Write() followed by Read().
//
// If |tmp_buffer| is supplied this buffer will be used, otherwise
// If `tmp_buffer` is supplied this buffer will be used, otherwise
// a buffer will created in the method.
std::unique_ptr<StunAttribute> CopyStunAttribute(
const StunAttribute& attribute,

2
api/turn_customizer.h
View file

@ -29,7 +29,7 @@ class TurnCustomizer {
cricket::StunMessage* message) = 0;
// TURN can send data using channel data messages or Send indication.
// This method should return false if |data| should be sent using
// This method should return false if `data` should be sent using
// a Send indication instead of a ChannelData message, even if a
// channel is bound.
virtual bool AllowChannelData(cricket::PortInterface* port,

4
api/video/color_space.cc
View file

@ -12,7 +12,7 @@
namespace webrtc {
namespace {
// Try to convert |enum_value| into the enum class T. |enum_bitmask| is created
// Try to convert `enum_value` into the enum class T. `enum_bitmask` is created
// by the funciton below. Returns true if conversion was successful, false
// otherwise.
template <typename T>
@ -43,7 +43,7 @@ constexpr int MakeMask(const int index, const int length, T (&values)[N]) {
}
// Create a bitmask where each bit corresponds to one potential enum value.
// |values| should be an array listing all possible enum values. The bit is set
// `values` should be an array listing all possible enum values. The bit is set
// to one if the corresponding enum exists. Only works for enums with values
// less than 64.
template <typename T, size_t N>

2
api/video/encoded_frame.h
View file

@ -46,7 +46,7 @@ class EncodedFrame : public webrtc::VCMEncodedFrame {
int64_t Id() const { return id_; }
// TODO(philipel): Add simple modify/access functions to prevent adding too
// many |references|.
// many `references`.
size_t num_references = 0;
int64_t references[kMaxFrameReferences];
// Is this subframe the last one in the superframe (In RTP stream that would

4
api/video/encoded_image.h
View file

@ -98,7 +98,7 @@ class RTC_EXPORT EncodedImage {
}
// These methods can be used to set/get size of subframe with spatial index
// |spatial_index| on encoded frames that consist of multiple spatial layers.
// `spatial_index` on encoded frames that consist of multiple spatial layers.
absl::optional<size_t> SpatialLayerFrameSize(int spatial_index) const;
void SetSpatialLayerFrameSize(int spatial_index, size_t size_bytes);
@ -195,7 +195,7 @@ class RTC_EXPORT EncodedImage {
// carries the webrtc::VideoFrame id field from the sender to the receiver.
absl::optional<uint16_t> video_frame_tracking_id_;
// Information about packets used to assemble this video frame. This is needed
// by |SourceTracker| when the frame is delivered to the RTCRtpReceiver's
// by `SourceTracker` when the frame is delivered to the RTCRtpReceiver's
// MediaStreamTrack, in order to implement getContributingSources(). See:
// https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources
RtpPacketInfos packet_infos_;

8
api/video/i010_buffer.h
View file

@ -34,7 +34,7 @@ class I010Buffer : public I010BufferInterface {
// Convert and put I420 buffer into a new buffer.
static rtc::scoped_refptr<I010Buffer> Copy(const I420BufferInterface& buffer);
// Return a rotated copy of |src|.
// Return a rotated copy of `src`.
static rtc::scoped_refptr<I010Buffer> Rotate(const I010BufferInterface& src,
VideoRotation rotation);
@ -55,15 +55,15 @@ class I010Buffer : public I010BufferInterface {
uint16_t* MutableDataU();
uint16_t* MutableDataV();
// Scale the cropped area of |src| to the size of |this| buffer, and
// write the result into |this|.
// Scale the cropped area of `src` to the size of `this` buffer, and
// write the result into `this`.
void CropAndScaleFrom(const I010BufferInterface& src,
int offset_x,
int offset_y,
int crop_width,
int crop_height);
// Scale all of |src| to the size of |this| buffer, with no cropping.
// Scale all of `src` to the size of `this` buffer, with no cropping.
void ScaleFrom(const I010BufferInterface& src);
// Pastes whole picture to canvas at (offset_row, offset_col).

8
api/video/i420_buffer.h
View file

@ -49,7 +49,7 @@ class RTC_EXPORT I420Buffer : public I420BufferInterface {
const uint8_t* data_v,
int stride_v);
// Returns a rotated copy of |src|.
// Returns a rotated copy of `src`.
static rtc::scoped_refptr<I420Buffer> Rotate(const I420BufferInterface& src,
VideoRotation rotation);
// Deprecated.
@ -83,8 +83,8 @@ class RTC_EXPORT I420Buffer : public I420BufferInterface {
uint8_t* MutableDataU();
uint8_t* MutableDataV();
// Scale the cropped area of |src| to the size of |this| buffer, and
// write the result into |this|.
// Scale the cropped area of `src` to the size of `this` buffer, and
// write the result into `this`.
void CropAndScaleFrom(const I420BufferInterface& src,
int offset_x,
int offset_y,
@ -95,7 +95,7 @@ class RTC_EXPORT I420Buffer : public I420BufferInterface {
// aspect ratio without distorting the image.
void CropAndScaleFrom(const I420BufferInterface& src);
// Scale all of |src| to the size of |this| buffer, with no cropping.
// Scale all of `src` to the size of `this` buffer, with no cropping.
void ScaleFrom(const I420BufferInterface& src);
// Pastes whole picture to canvas at (offset_row, offset_col).

4
api/video/nv12_buffer.h
View file

@ -56,8 +56,8 @@ class RTC_EXPORT NV12Buffer : public NV12BufferInterface {
// are resolved in a better way. Or in the mean time, use SetBlack.
void InitializeData();
// Scale the cropped area of |src| to the size of |this| buffer, and
// write the result into |this|.
// Scale the cropped area of `src` to the size of `this` buffer, and
// write the result into `this`.
void CropAndScaleFrom(const NV12BufferInterface& src,
int offset_x,
int offset_y,

4
api/video/video_bitrate_allocation.h
View file

@ -50,8 +50,8 @@ class RTC_EXPORT VideoBitrateAllocation {
// Get the sum of all the temporal layer for a specific spatial layer.
uint32_t GetSpatialLayerSum(size_t spatial_index) const;
// Sum of bitrates of temporal layers, from layer 0 to |temporal_index|
// inclusive, of specified spatial layer |spatial_index|. Bitrates of lower
// Sum of bitrates of temporal layers, from layer 0 to `temporal_index`
// inclusive, of specified spatial layer `spatial_index`. Bitrates of lower
// spatial layers are not included.
uint32_t GetTemporalLayerSum(size_t spatial_index,
size_t temporal_index) const;

2
api/video/video_frame.h
View file

@ -272,7 +272,7 @@ class RTC_EXPORT VideoFrame {
// update_rect() will return a rectangle corresponding to the entire frame.
absl::optional<UpdateRect> update_rect_;
// Information about packets used to assemble this video frame. This is needed
// by |SourceTracker| when the frame is delivered to the RTCRtpReceiver's
// by `SourceTracker` when the frame is delivered to the RTCRtpReceiver's
// MediaStreamTrack, in order to implement getContributingSources(). See:
// https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources
RtpPacketInfos packet_infos_;

4
api/video/video_frame_buffer.h
View file

@ -84,8 +84,8 @@ class RTC_EXPORT VideoFrameBuffer : public rtc::RefCountInterface {
// A format specific scale function. Default implementation works by
// converting to I420. But more efficient implementations may override it,
// especially for kNative.
// First, the image is cropped to |crop_width| and |crop_height| and then
// scaled to |scaled_width| and |scaled_height|.
// First, the image is cropped to `crop_width` and `crop_height` and then
// scaled to `scaled_width` and `scaled_height`.
virtual rtc::scoped_refptr<VideoFrameBuffer> CropAndScale(int offset_x,
int offset_y,
int crop_width,

6
api/video/video_source_interface.h
View file

@ -54,7 +54,7 @@ struct RTC_EXPORT VideoSinkWants {
int max_framerate_fps = std::numeric_limits<int>::max();
// Tells the source that the sink wants width and height of the video frames
// to be divisible by |resolution_alignment|.
// to be divisible by `resolution_alignment`.
// For example: With I420, this value would be a multiple of 2.
// Note that this field is unrelated to any horizontal or vertical stride
// requirements the encoder has on the incoming video frame buffers.
@ -71,13 +71,13 @@ struct RTC_EXPORT VideoSinkWants {
// to scaleResolutionDownBy or turning off simulcast or SVC layers.
//
// For example, we may capture at 720p and due to adaptation (e.g. applying
// |max_pixel_count| constraints) create webrtc::VideoFrames of size 480p, but
// `max_pixel_count` constraints) create webrtc::VideoFrames of size 480p, but
// if we do scaleResolutionDownBy:2 then the only resolution we end up
// encoding is 240p. In this case we still need to provide webrtc::VideoFrames
// of size 480p but we can optimize internal buffers for 240p, avoiding
// downsampling to 480p if possible.
//
// Note that the |resolutions| can change while frames are in flight and
// Note that the `resolutions` can change while frames are in flight and
// should only be used as a hint when constructing the webrtc::VideoFrame.
std::vector<FrameSize> resolutions;
};

2
api/video/video_stream_decoder_create.h
View file

@ -20,7 +20,7 @@
#include "api/video_codecs/sdp_video_format.h"
namespace webrtc {
// The |decoder_settings| parameter is a map between:
// The `decoder_settings` parameter is a map between:
// <payload type> --> <<video format>, <number of cores>>.
// The video format is used when instantiating a decoder, and
// the number of cores is used when initializing the decoder.

20
api/video/video_stream_encoder_interface.h
View file

@ -68,9 +68,9 @@ class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> {
GetAdaptationResources() = 0;
// Sets the source that will provide video frames to the VideoStreamEncoder's
// OnFrame method. |degradation_preference| control whether or not resolution
// OnFrame method. `degradation_preference` control whether or not resolution
// or frame rate may be reduced. The VideoStreamEncoder registers itself with
// |source|, and signals adaptation decisions to the source in the form of
// `source`, and signals adaptation decisions to the source in the form of
// VideoSinkWants.
// TODO(nisse): When adaptation logic is extracted from this class,
// it no longer needs to know the source.
@ -78,8 +78,8 @@ class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> {
rtc::VideoSourceInterface<VideoFrame>* source,
const DegradationPreference& degradation_preference) = 0;
// Sets the |sink| that gets the encoded frames. |rotation_applied| means
// that the source must support rotation. Only set |rotation_applied| if the
// Sets the `sink` that gets the encoded frames. `rotation_applied` means
// that the source must support rotation. Only set `rotation_applied` if the
// remote side does not support the rotation extension.
virtual void SetSink(EncoderSink* sink, bool rotation_applied) = 0;
@ -102,13 +102,13 @@ class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> {
virtual void OnLossNotification(
const VideoEncoder::LossNotification& loss_notification) = 0;
// Set the currently estimated network properties. A |target_bitrate|
// Set the currently estimated network properties. A `target_bitrate`
// of zero pauses the encoder.
// |stable_target_bitrate| is a filtered version of |target_bitrate|. It is
// `stable_target_bitrate` is a filtered version of `target_bitrate`. It is
// always less or equal to it. It can be used to avoid rapid changes of
// expensive encoding settings, such as resolution.
// |link_allocation| is the bandwidth available for this video stream on the
// network link. It is always at least |target_bitrate| but may be higher
// `link_allocation` is the bandwidth available for this video stream on the
// network link. It is always at least `target_bitrate` but may be higher
// if we are not network constrained.
virtual void OnBitrateUpdated(DataRate target_bitrate,
DataRate stable_target_bitrate,
@ -122,8 +122,8 @@ class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> {
virtual void SetFecControllerOverride(
FecControllerOverride* fec_controller_override) = 0;
// Creates and configures an encoder with the given |config|. The
// |max_data_payload_length| is used to support single NAL unit
// Creates and configures an encoder with the given `config`. The
// `max_data_payload_length` is used to support single NAL unit
// packetization for H.264.
virtual void ConfigureEncoder(VideoEncoderConfig config,
size_t max_data_payload_length) = 0;

2
api/video/video_stream_encoder_observer.h
View file

@ -101,7 +101,7 @@ class VideoStreamEncoderObserver : public CpuOveruseMetricsObserver {
const VideoBitrateAllocation& allocation) {}
// Informes observer if an internal encoder scaler has reduced video
// resolution or not. |is_scaled| is a flag indicating if the video is scaled
// resolution or not. `is_scaled` is a flag indicating if the video is scaled
// down.
virtual void OnEncoderInternalScalerUpdate(bool is_scaled) {}

2
api/video/video_timing.h
View file

@ -55,7 +55,7 @@ struct TimingFrameInfo {
// synchronized, -1 otherwise.
int64_t EndToEndDelay() const;
// Returns true if current frame took longer to process than |other| frame.
// Returns true if current frame took longer to process than `other` frame.
// If other frame's clocks are not synchronized, current frame is always
// preferred.
bool IsLongerThan(const TimingFrameInfo& other) const;

2
api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc
View file

@ -75,7 +75,7 @@ class VideoDecoderSoftwareFallbackWrapperTest : public ::testing::Test {
int reset_count_ = 0;
};
test::ScopedFieldTrials override_field_trials_;
// |fake_decoder_| is owned and released by |fallback_wrapper_|.
// `fake_decoder_` is owned and released by `fallback_wrapper_`.
CountingFakeDecoder* fake_decoder_;
std::unique_ptr<VideoDecoder> fallback_wrapper_;
};

2
api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
View file

@ -172,7 +172,7 @@ class VideoEncoderSoftwareFallbackWrapperTestBase : public ::testing::Test {
test::ScopedFieldTrials override_field_trials_;
FakeEncodedImageCallback callback_;
// |fake_encoder_| is owned and released by |fallback_wrapper_|.
// `fake_encoder_` is owned and released by `fallback_wrapper_`.
CountingFakeEncoder* fake_encoder_;
CountingFakeEncoder* fake_sw_encoder_;
bool wrapper_initialized_;

2
api/video_codecs/video_decoder_factory.h
View file

@ -40,7 +40,7 @@ class RTC_EXPORT VideoDecoderFactory {
// power efficient, which is currently interpreted as if there is support for
// hardware acceleration.
// See https://w3c.github.io/webrtc-svc/#scalabilitymodes* for a specification
// of valid values for |scalability_mode|.
// of valid values for `scalability_mode`.
// NOTE: QueryCodecSupport is currently an experimental feature that is
// subject to change without notice.
virtual CodecSupport QueryCodecSupport(

26
api/video_codecs/video_encoder.h
View file

@ -167,7 +167,7 @@ class RTC_EXPORT VideoEncoder {
ScalingSettings scaling_settings;
// The width and height of the incoming video frames should be divisible
// by |requested_resolution_alignment|. If they are not, the encoder may
// by `requested_resolution_alignment`. If they are not, the encoder may
// drop the incoming frame.
// For example: With I420, this value would be a multiple of 2.
// Note that this field is unrelated to any horizontal or vertical stride
@ -175,12 +175,12 @@ class RTC_EXPORT VideoEncoder {
int requested_resolution_alignment;
// Same as above but if true, each simulcast layer should also be divisible
// by |requested_resolution_alignment|.
// Note that scale factors |scale_resolution_down_by| may be adjusted so a
// by `requested_resolution_alignment`.
// Note that scale factors `scale_resolution_down_by` may be adjusted so a
// common multiple is not too large to avoid largely cropped frames and
// possibly with an aspect ratio far from the original.
// Warning: large values of scale_resolution_down_by could be changed
// considerably, especially if |requested_resolution_alignment| is large.
// considerably, especially if `requested_resolution_alignment` is large.
bool apply_alignment_to_all_simulcast_layers;
// If true, encoder supports working with a native handle (e.g. texture
@ -215,7 +215,7 @@ class RTC_EXPORT VideoEncoder {
bool has_internal_source;
// For each spatial layer (simulcast stream or SVC layer), represented as an
// element in |fps_allocation| a vector indicates how many temporal layers
// element in `fps_allocation` a vector indicates how many temporal layers
// the encoder is using for that spatial layer.
// For each spatial/temporal layer pair, the frame rate fraction is given as
// an 8bit unsigned integer where 0 = 0% and 255 = 100%.
@ -243,8 +243,8 @@ class RTC_EXPORT VideoEncoder {
// Recommended bitrate limits for different resolutions.
std::vector<ResolutionBitrateLimits> resolution_bitrate_limits;
// Obtains the limits from |resolution_bitrate_limits| that best matches the
// |frame_size_pixels|.
// Obtains the limits from `resolution_bitrate_limits` that best matches the
// `frame_size_pixels`.
absl::optional<ResolutionBitrateLimits>
GetEncoderBitrateLimitsForResolution(int frame_size_pixels) const;
@ -279,7 +279,7 @@ class RTC_EXPORT VideoEncoder {
VideoBitrateAllocation bitrate;
// Target framerate, in fps. A value <= 0.0 is invalid and should be
// interpreted as framerate target not available. In this case the encoder
// should fall back to the max framerate specified in |codec_settings| of
// should fall back to the max framerate specified in `codec_settings` of
// the last InitEncode() call.
double framerate_fps;
// The network bandwidth available for video. This is at least
@ -299,15 +299,15 @@ class RTC_EXPORT VideoEncoder {
uint32_t timestamp_of_last_received;
// Describes whether the dependencies of the last received frame were
// all decodable.
// |false| if some dependencies were undecodable, |true| if all dependencies
// were decodable, and |nullopt| if the dependencies are unknown.
// `false` if some dependencies were undecodable, `true` if all dependencies
// were decodable, and `nullopt` if the dependencies are unknown.
absl::optional<bool> dependencies_of_last_received_decodable;
// Describes whether the received frame was decodable.
// |false| if some dependency was undecodable or if some packet belonging
// `false` if some dependency was undecodable or if some packet belonging
// to the last received frame was missed.
// |true| if all dependencies were decodable and all packets belonging
// `true` if all dependencies were decodable and all packets belonging
// to the last received frame were received.
// |nullopt| if no packet belonging to the last frame was missed, but the
// `nullopt` if no packet belonging to the last frame was missed, but the
// last packet in the frame was not yet received.
absl::optional<bool> last_received_decodable;
};

6
api/video_codecs/video_encoder_config.h
View file

@ -24,7 +24,7 @@
namespace webrtc {
// The |VideoStream| struct describes a simulcast layer, or "stream".
// The `VideoStream` struct describes a simulcast layer, or "stream".
struct VideoStream {
VideoStream();
~VideoStream();
@ -46,7 +46,7 @@ struct VideoStream {
int max_bitrate_bps;
// Scaling factor applied to the stream size.
// |width| and |height| values are already scaled down.
// `width` and `height` values are already scaled down.
double scale_resolution_down_by;
// Maximum Quantization Parameter to use when encoding the stream.
@ -171,7 +171,7 @@ class VideoEncoderConfig {
// The simulcast layer's configurations set by the application for this video
// sender. These are modified by the video_stream_factory before being passed
// down to lower layers for the video encoding.
// |simulcast_layers| is also used for configuring non-simulcast (when there
// `simulcast_layers` is also used for configuring non-simulcast (when there
// is a single VideoStream).
std::vector<VideoStream> simulcast_layers;

4
api/video_codecs/video_encoder_factory.h
View file

@ -29,7 +29,7 @@ class VideoEncoderFactory {
public:
// TODO(magjed): Try to get rid of this struct.
struct CodecInfo {
// |has_internal_source| is true if encoders created by this factory of the
// `has_internal_source` is true if encoders created by this factory of the
// given codec will use internal camera sources, meaning that they don't
// require/expect frames to be delivered via webrtc::VideoEncoder::Encode.
// This flag is used as the internal_source parameter to
@ -88,7 +88,7 @@ class VideoEncoderFactory {
// power efficient, which is currently interpreted as if there is support for
// hardware acceleration.
// See https://w3c.github.io/webrtc-svc/#scalabilitymodes* for a specification
// of valid values for |scalability_mode|.
// of valid values for `scalability_mode`.
// NOTE: QueryCodecSupport is currently an experimental feature that is
// subject to change without notice.
virtual CodecSupport QueryCodecSupport(

8
api/video_codecs/video_encoder_software_fallback_wrapper.cc
View file

@ -39,8 +39,8 @@ namespace {
// If forced fallback is allowed, either:
//
// 1) The forced fallback is requested if the resolution is less than or equal
// to |max_pixels_|. The resolution is allowed to be scaled down to
// |min_pixels_|.
// to `max_pixels_`. The resolution is allowed to be scaled down to
// `min_pixels_`.
//
// 2) The forced fallback is requested if temporal support is preferred and the
// SW fallback supports temporal layers while the HW encoder does not.
@ -274,8 +274,8 @@ bool VideoEncoderSoftwareFallbackWrapper::InitFallbackEncoder(bool is_forced) {
void VideoEncoderSoftwareFallbackWrapper::SetFecControllerOverride(
FecControllerOverride* fec_controller_override) {
// It is important that only one of those would ever interact with the
// |fec_controller_override| at a given time. This is the responsibility
// of |this| to maintain.
// `fec_controller_override` at a given time. This is the responsibility
// of `this` to maintain.
fec_controller_override_ = fec_controller_override;
current_encoder()->SetFecControllerOverride(fec_controller_override);

2
api/video_codecs/video_encoder_software_fallback_wrapper.h
View file

@ -32,7 +32,7 @@ CreateVideoEncoderSoftwareFallbackWrapper(
bool prefer_temporal_support);
// Default fallback for call-sites not yet updated with
// |prefer_temporal_support|.
// `prefer_temporal_support`.
// TODO(sprang): Remove when usage is gone.
RTC_EXPORT inline std::unique_ptr<VideoEncoder>
CreateVideoEncoderSoftwareFallbackWrapper(

22
api/video_codecs/vp8_frame_buffer_controller.h
View file

@ -66,7 +66,7 @@ struct Vp8EncoderConfig {
// Number of active temporal layers. Set to 0 if not used.
uint32_t ts_number_layers;
// Arrays of length |ts_number_layers|, indicating (cumulative) target
// Arrays of length `ts_number_layers`, indicating (cumulative) target
// bitrate and rate decimator (e.g. 4 if every 4th frame is in the given
// layer) for each active temporal layer, starting with temporal id 0.
std::array<uint32_t, kMaxLayers> ts_target_bitrate;
@ -75,7 +75,7 @@ struct Vp8EncoderConfig {
// The periodicity of the temporal pattern. Set to 0 if not used.
uint32_t ts_periodicity;
// Array of length |ts_periodicity| indicating the sequence of temporal id's
// Array of length `ts_periodicity` indicating the sequence of temporal id's
// to assign to incoming frames.
std::array<uint32_t, kMaxPeriodicity> ts_layer_id;
};
@ -106,7 +106,7 @@ class Vp8FrameBufferController {
// The limits are suggestion-only; the controller is allowed to exceed them.
virtual void SetQpLimits(size_t stream_index, int min_qp, int max_qp) = 0;
// Number of streamed controlled by |this|.
// Number of streamed controlled by `this`.
virtual size_t StreamCount() const = 0;
// If this method returns true, the encoder is free to drop frames for
@ -121,7 +121,7 @@ class Vp8FrameBufferController {
virtual bool SupportsEncoderFrameDropping(size_t stream_index) const = 0;
// New target bitrate for a stream (each entry in
// |bitrates_bps| is for another temporal layer).
// `bitrates_bps` is for another temporal layer).
virtual void OnRatesUpdated(size_t stream_index,
const std::vector<uint32_t>& bitrates_bps,
int framerate_fps) = 0;
@ -130,7 +130,7 @@ class Vp8FrameBufferController {
// the controller wishes to enact in the encoder's configuration.
// If a value is not overridden, previous overrides are still in effect.
// However, if |Vp8EncoderConfig::reset_previous_configuration_overrides|
// is set to |true|, all previous overrides are reset.
// is set to `true`, all previous overrides are reset.
virtual Vp8EncoderConfig UpdateConfiguration(size_t stream_index) = 0;
// Returns the recommended VP8 encode flags needed.
@ -142,13 +142,13 @@ class Vp8FrameBufferController {
virtual Vp8FrameConfig NextFrameConfig(size_t stream_index,
uint32_t rtp_timestamp) = 0;
// Called after the encode step is done. |rtp_timestamp| must match the
// Called after the encode step is done. `rtp_timestamp` must match the
// parameter use in the NextFrameConfig() call.
// |is_keyframe| must be true iff the encoder decided to encode this frame as
// `is_keyframe` must be true iff the encoder decided to encode this frame as
// a keyframe.
// If |info| is not null, the encoder may update |info| with codec specific
// data such as temporal id. |qp| should indicate the frame-level QP this
// frame was encoded at. If the encoder does not support extracting this, |qp|
// If `info` is not null, the encoder may update `info` with codec specific
// data such as temporal id. `qp` should indicate the frame-level QP this
// frame was encoded at. If the encoder does not support extracting this, `qp`
// should be set to 0.
virtual void OnEncodeDone(size_t stream_index,
uint32_t rtp_timestamp,
@ -161,7 +161,7 @@ class Vp8FrameBufferController {
virtual void OnFrameDropped(size_t stream_index, uint32_t rtp_timestamp) = 0;
// Called by the encoder when the packet loss rate changes.
// |packet_loss_rate| runs between 0.0 (no loss) and 1.0 (everything lost).
// `packet_loss_rate` runs between 0.0 (no loss) and 1.0 (everything lost).
virtual void OnPacketLossRateUpdate(float packet_loss_rate) = 0;
// Called by the encoder when the round trip time changes.

4
api/video_track_source_proxy_factory.h
View file

@ -15,9 +15,9 @@
namespace webrtc {
// Creates a proxy source for |source| which makes sure the real
// Creates a proxy source for `source` which makes sure the real
// VideoTrackSourceInterface implementation is destroyed on the signaling thread
// and marshals calls to |worker_thread| and |signaling_thread|.
// and marshals calls to `worker_thread` and `signaling_thread`.
rtc::scoped_refptr<VideoTrackSourceInterface> RTC_EXPORT
CreateVideoTrackSourceProxy(rtc::Thread* signaling_thread,
rtc::Thread* worker_thread,

26
api/voip/voip_base.h
View file

@ -56,53 +56,53 @@ class VoipBase {
// 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`, webrtc::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
// VoipNetwork interface. |local_ssrc| is optional and when local_ssrc is not
// VoipNetwork interface. `local_ssrc` is optional and when local_ssrc is not
// set, some random value will be used by voip engine.
// Returns a ChannelId created for caller to handle subsequent Channel
// operations.
virtual ChannelId CreateChannel(Transport* transport,
absl::optional<uint32_t> local_ssrc) = 0;
// Releases |channel_id| that no longer has any use.
// Releases `channel_id` that no longer has any use.
// Returns following VoipResult;
// kOk - |channel_id| is released.
// kInvalidArgument - |channel_id| is invalid.
// kOk - `channel_id` is released.
// kInvalidArgument - `channel_id` is invalid.
// kInternal - Fails to stop audio output device.
virtual VoipResult ReleaseChannel(ChannelId channel_id) = 0;
// Starts sending on |channel_id|. This starts microphone if not started yet.
// Starts sending on `channel_id`. This starts microphone if not started yet.
// Returns following VoipResult;
// kOk - Channel successfully started to send.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
// kFailedPrecondition - Missing prerequisite on VoipCodec::SetSendCodec.
// kInternal - initialization has failed on selected microphone.
virtual VoipResult StartSend(ChannelId channel_id) = 0;
// Stops sending on |channel_id|. If this is the last active channel, it will
// Stops sending on `channel_id`. If this is the last active channel, it will
// stop microphone input from underlying audio platform layer.
// Returns following VoipResult;
// kOk - Channel successfully stopped to send.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
// kInternal - Failed to stop the active microphone device.
virtual VoipResult StopSend(ChannelId channel_id) = 0;
// Starts playing on speaker device for |channel_id|.
// Starts playing on speaker device for `channel_id`.
// This will start underlying platform speaker device if not started.
// Returns following VoipResult;
// kOk - Channel successfully started to play out.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
// kFailedPrecondition - Missing prerequisite on VoipCodec::SetReceiveCodecs.
// kInternal - Failed to initializate the selected speaker device.
virtual VoipResult StartPlayout(ChannelId channel_id) = 0;
// Stops playing on speaker device for |channel_id|.
// Stops playing on speaker device for `channel_id`.
// Returns following VoipResult;
// kOk - Channel successfully stopped t play out.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult StopPlayout(ChannelId channel_id) = 0;
protected:

4
api/voip/voip_codec.h
View file

@ -31,7 +31,7 @@ class VoipCodec {
// Set encoder type here along with its payload type to use.
// Returns following VoipResult;
// kOk - sending codec is set as provided.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult SetSendCodec(ChannelId channel_id,
int payload_type,
const SdpAudioFormat& encoder_spec) = 0;
@ -42,7 +42,7 @@ class VoipCodec {
// direction.
// Returns following VoipResult;
// kOk - receiving codecs are set as provided.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult SetReceiveCodecs(
ChannelId channel_id,
const std::map<int, SdpAudioFormat>& decoder_specs) = 0;

6
api/voip/voip_dtmf.h
View file

@ -45,20 +45,20 @@ class VoipDtmf {
// type has been negotiated with remote.
// Returns following VoipResult;
// kOk - telephone event type is registered as provided.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult RegisterTelephoneEventType(ChannelId channel_id,
int rtp_payload_type,
int sample_rate_hz) = 0;
// Send DTMF named event as specified by
// https://tools.ietf.org/html/rfc4733#section-3.2
// |duration_ms| specifies the duration of DTMF packets that will be emitted
// `duration_ms` specifies the duration of DTMF packets that will be emitted
// in place of real RTP packets instead.
// Must be called after RegisterTelephoneEventType and VoipBase::StartSend
// have been called.
// Returns following VoipResult;
// kOk - requested DTMF event is successfully scheduled.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
// kFailedPrecondition - Missing prerequisite on RegisterTelephoneEventType
// or sending state.
virtual VoipResult SendDtmfEvent(ChannelId channel_id,

4
api/voip/voip_network.h
View file

@ -24,7 +24,7 @@ class VoipNetwork {
// The data received from the network including RTP header is passed here.
// Returns following VoipResult;
// kOk - received RTP packet is processed.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult ReceivedRTPPacket(
ChannelId channel_id,
rtc::ArrayView<const uint8_t> rtp_packet) = 0;
@ -32,7 +32,7 @@ class VoipNetwork {
// The data received from the network including RTCP header is passed here.
// Returns following VoipResult;
// kOk - received RTCP packet is processed.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult ReceivedRTCPPacket(
ChannelId channel_id,
rtc::ArrayView<const uint8_t> rtcp_packet) = 0;

8
api/voip/voip_statistics.h
View file

@ -75,17 +75,17 @@ struct ChannelStatistics {
// the jitter buffer (NetEq) performance.
class VoipStatistics {
public:
// Gets the audio ingress statistics by |ingress_stats| reference.
// Gets the audio ingress statistics by `ingress_stats` reference.
// Returns following VoipResult;
// kOk - successfully set provided IngressStatistics reference.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult GetIngressStatistics(ChannelId channel_id,
IngressStatistics& ingress_stats) = 0;
// Gets the channel statistics by |channel_stats| reference.
// Gets the channel statistics by `channel_stats` reference.
// Returns following VoipResult;
// kOk - successfully set provided ChannelStatistics reference.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult GetChannelStatistics(ChannelId channel_id,
ChannelStatistics& channel_stats) = 0;

12
api/voip/voip_volume_control.h
View file

@ -37,21 +37,21 @@ class VoipVolumeControl {
// mute doesn't affect audio input level and energy values as input sample is
// silenced after the measurement.
// Returns following VoipResult;
// kOk - input source muted or unmuted as provided by |enable|.
// kInvalidArgument - |channel_id| is invalid.
// kOk - input source muted or unmuted as provided by `enable`.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult SetInputMuted(ChannelId channel_id, bool enable) = 0;
// Gets the microphone volume info via |volume_info| reference.
// Gets the microphone volume info via `volume_info` reference.
// Returns following VoipResult;
// kOk - successfully set provided input volume info.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult GetInputVolumeInfo(ChannelId channel_id,
VolumeInfo& volume_info) = 0;
// Gets the speaker volume info via |volume_info| reference.
// Gets the speaker volume info via `volume_info` reference.
// Returns following VoipResult;
// kOk - successfully set provided output volume info.
// kInvalidArgument - |channel_id| is invalid.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult GetOutputVolumeInfo(ChannelId channel_id,
VolumeInfo& volume_info) = 0;