diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h index e0b15dc0c9..b40750891c 100644 --- a/audio/audio_send_stream.h +++ b/audio/audio_send_stream.h @@ -189,7 +189,7 @@ class AudioSendStream final : public webrtc::AudioSendStream, BitrateAllocatorInterface* const bitrate_allocator_ RTC_GUARDED_BY(rtp_transport_queue_); - // Constrains cached to be accessed from |rtp_transport_queue_|. + // Constrains cached to be accessed from `rtp_transport_queue_`. absl::optional cached_constraints_ RTC_GUARDED_BY(rtp_transport_queue_) = absl::nullopt; RtpTransportControllerSendInterface* const rtp_transport_; diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc index 357e08040c..db42efc373 100644 --- a/audio/audio_send_stream_unittest.cc +++ b/audio/audio_send_stream_unittest.cc @@ -172,7 +172,7 @@ struct ConfigHelper { SetupMockForSetupSendCodec(expect_set_encoder_call); SetupMockForCallEncoder(); - // Use ISAC as default codec so as to prevent unnecessary |channel_proxy_| + // Use ISAC as default codec so as to prevent unnecessary `channel_proxy_` // calls from the default ctor behavior. stream_config_.send_codec_spec = AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat); @@ -336,7 +336,7 @@ struct ConfigHelper { ::testing::NiceMock rtp_rtcp_; ::testing::NiceMock limit_observer_; BitrateAllocator bitrate_allocator_; - // |worker_queue| is defined last to ensure all pending tasks are cancelled + // `worker_queue` is defined last to ensure all pending tasks are cancelled // and deleted before any other members. TaskQueueForTest worker_queue_; std::unique_ptr audio_encoder_; diff --git a/audio/audio_transport_impl.cc b/audio/audio_transport_impl.cc index 8710ced9b7..2a80ea893d 100644 --- a/audio/audio_transport_impl.cc +++ b/audio/audio_transport_impl.cc @@ -64,8 +64,8 @@ void ProcessCaptureFrame(uint32_t delay_ms, } } -// Resample audio in |frame| to given sample rate preserving the -// channel count and place the result in |destination|. +// Resample audio in `frame` to given sample rate preserving the +// channel count and place the result in `destination`. int Resample(const AudioFrame& frame, const int destination_sample_rate, PushResampler* resampler, diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc index 57269cd193..3ca3b51bb1 100644 --- a/audio/channel_receive.cc +++ b/audio/channel_receive.cc @@ -429,8 +429,8 @@ AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo( } // Measure audio level (0-9) - // TODO(henrik.lundin) Use the |muted| information here too. - // TODO(deadbeef): Use RmsLevel for |_outputAudioLevel| (see + // TODO(henrik.lundin) Use the `muted` information here too. + // TODO(deadbeef): Use RmsLevel for `_outputAudioLevel` (see // https://crbug.com/webrtc/7517). _outputAudioLevel.ComputeLevel(*audio_frame, kAudioSampleDurationSeconds); @@ -454,10 +454,10 @@ AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo( // Compute ntp time. audio_frame->ntp_time_ms_ = ntp_estimator_.Estimate(audio_frame->timestamp_); - // |ntp_time_ms_| won't be valid until at least 2 RTCP SRs are received. + // `ntp_time_ms_` won't be valid until at least 2 RTCP SRs are received. if (audio_frame->ntp_time_ms_ > 0) { - // Compute |capture_start_ntp_time_ms_| so that - // |capture_start_ntp_time_ms_| + |elapsed_time_ms_| == |ntp_time_ms_| + // Compute `capture_start_ntp_time_ms_` so that + // `capture_start_ntp_time_ms_` + `elapsed_time_ms_` == `ntp_time_ms_` capture_start_ntp_time_ms_ = audio_frame->ntp_time_ms_ - audio_frame->elapsed_time_ms_; } diff --git a/audio/channel_receive_frame_transformer_delegate.h b/audio/channel_receive_frame_transformer_delegate.h index f59834d24e..04ad7c4695 100644 --- a/audio/channel_receive_frame_transformer_delegate.h +++ b/audio/channel_receive_frame_transformer_delegate.h @@ -23,7 +23,7 @@ namespace webrtc { // Delegates calls to FrameTransformerInterface to transform frames, and to // ChannelReceive to receive the transformed frames using the -// |receive_frame_callback_| on the |channel_receive_thread_|. +// `receive_frame_callback_` on the `channel_receive_thread_`. class ChannelReceiveFrameTransformerDelegate : public TransformedFrameCallback { public: using ReceiveFrameCallback = @@ -34,12 +34,12 @@ class ChannelReceiveFrameTransformerDelegate : public TransformedFrameCallback { rtc::scoped_refptr frame_transformer, TaskQueueBase* channel_receive_thread); - // Registers |this| as callback for |frame_transformer_|, to get the + // Registers `this` as callback for `frame_transformer_`, to get the // transformed frames. void Init(); - // Unregisters and releases the |frame_transformer_| reference, and resets - // |receive_frame_callback_| on |channel_receive_thread_|. Called from + // Unregisters and releases the `frame_transformer_` reference, and resets + // `receive_frame_callback_` on `channel_receive_thread_`. Called from // ChannelReceive destructor to prevent running the callback on a dangling // channel. void Reset(); @@ -55,7 +55,7 @@ class ChannelReceiveFrameTransformerDelegate : public TransformedFrameCallback { std::unique_ptr frame) override; // Delegates the call to ChannelReceive::OnReceivedPayloadData on the - // |channel_receive_thread_|, by calling |receive_frame_callback_|. + // `channel_receive_thread_`, by calling `receive_frame_callback_`. void ReceiveFrame(std::unique_ptr frame) const; protected: diff --git a/audio/channel_send.h b/audio/channel_send.h index 67391af956..663b947036 100644 --- a/audio/channel_send.h +++ b/audio/channel_send.h @@ -98,12 +98,12 @@ class ChannelSendInterface { std::unique_ptr audio_frame) = 0; virtual RtpRtcpInterface* GetRtpRtcp() const = 0; - // In RTP we currently rely on RTCP packets (|ReceivedRTCPPacket|) to inform + // In RTP we currently rely on RTCP packets (`ReceivedRTCPPacket`) to inform // about RTT. // In media transport we rely on the TargetTransferRateObserver instead. // In other words, if you are using RTP, you should expect - // |ReceivedRTCPPacket| to be called, if you are using media transport, - // |OnTargetTransferRate| will be called. + // `ReceivedRTCPPacket` to be called, if you are using media transport, + // `OnTargetTransferRate` will be called. // // In future, RTP media will move to the media transport implementation and // these conditions will be removed. diff --git a/audio/channel_send_frame_transformer_delegate.h b/audio/channel_send_frame_transformer_delegate.h index 9b7eb33b5c..6d9f0a8613 100644 --- a/audio/channel_send_frame_transformer_delegate.h +++ b/audio/channel_send_frame_transformer_delegate.h @@ -23,8 +23,8 @@ namespace webrtc { // Delegates calls to FrameTransformerInterface to transform frames, and to -// ChannelSend to send the transformed frames using |send_frame_callback_| on -// the |encoder_queue_|. +// ChannelSend to send the transformed frames using `send_frame_callback_` on +// the `encoder_queue_`. // OnTransformedFrame() can be called from any thread, the delegate ensures // thread-safe access to the ChannelSend callback. class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback { @@ -40,12 +40,12 @@ class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback { rtc::scoped_refptr frame_transformer, rtc::TaskQueue* encoder_queue); - // Registers |this| as callback for |frame_transformer_|, to get the + // Registers `this` as callback for `frame_transformer_`, to get the // transformed frames. void Init(); - // Unregisters and releases the |frame_transformer_| reference, and resets - // |send_frame_callback_| under lock. Called from ChannelSend destructor to + // Unregisters and releases the `frame_transformer_` reference, and resets + // `send_frame_callback_` under lock. Called from ChannelSend destructor to // prevent running the callback on a dangling channel. void Reset(); @@ -64,8 +64,8 @@ class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback { void OnTransformedFrame( std::unique_ptr frame) override; - // Delegates the call to ChannelSend::SendRtpAudio on the |encoder_queue_|, - // by calling |send_audio_callback_|. + // Delegates the call to ChannelSend::SendRtpAudio on the `encoder_queue_`, + // by calling `send_audio_callback_`. void SendFrame(std::unique_ptr frame) const; protected: diff --git a/audio/null_audio_poller.cc b/audio/null_audio_poller.cc index 22f575d8bb..de2c5cabec 100644 --- a/audio/null_audio_poller.cc +++ b/audio/null_audio_poller.cc @@ -47,7 +47,7 @@ void NullAudioPoller::OnMessage(rtc::Message* msg) { // Buffer to hold the audio samples. int16_t buffer[kNumSamples * kNumChannels]; - // Output variables from |NeedMorePlayData|. + // Output variables from `NeedMorePlayData`. size_t n_samples; int64_t elapsed_time_ms; int64_t ntp_time_ms; diff --git a/audio/remix_resample.h b/audio/remix_resample.h index a45270b39a..bd8da76c6a 100644 --- a/audio/remix_resample.h +++ b/audio/remix_resample.h @@ -17,19 +17,19 @@ namespace webrtc { namespace voe { -// Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame| +// Upmix or downmix and resample the audio to `dst_frame`. Expects `dst_frame` // to have its sample rate and channels members set to the desired values. -// Updates the |samples_per_channel_| member accordingly. +// Updates the `samples_per_channel_` member accordingly. // -// This version has an AudioFrame |src_frame| as input and sets the output -// |timestamp_|, |elapsed_time_ms_| and |ntp_time_ms_| members equals to the +// This version has an AudioFrame `src_frame` as input and sets the output +// `timestamp_`, `elapsed_time_ms_` and `ntp_time_ms_` members equals to the // input ones. void RemixAndResample(const AudioFrame& src_frame, PushResampler* resampler, AudioFrame* dst_frame); -// This version has a pointer to the samples |src_data| as input and receives -// |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as +// This version has a pointer to the samples `src_data` as input and receives +// `samples_per_channel`, `num_channels` and `sample_rate_hz` of the data as // parameters. void RemixAndResample(const int16_t* src_data, size_t samples_per_channel, diff --git a/audio/remix_resample_unittest.cc b/audio/remix_resample_unittest.cc index d2155a64f0..a80476e3f1 100644 --- a/audio/remix_resample_unittest.cc +++ b/audio/remix_resample_unittest.cc @@ -43,7 +43,7 @@ class UtilityTest : public ::testing::Test { AudioFrame golden_frame_; }; -// Sets the signal value to increase by |data| with every sample. Floats are +// Sets the signal value to increase by `data` with every sample. Floats are // used so non-integer values result in rounding error, but not an accumulating // error. void SetMonoFrame(float data, int sample_rate_hz, AudioFrame* frame) { @@ -62,7 +62,7 @@ void SetMonoFrame(float data, AudioFrame* frame) { SetMonoFrame(data, frame->sample_rate_hz_, frame); } -// Sets the signal value to increase by |left| and |right| with every sample in +// Sets the signal value to increase by `left` and `right` with every sample in // each channel respectively. void SetStereoFrame(float left, float right, @@ -84,7 +84,7 @@ void SetStereoFrame(float left, float right, AudioFrame* frame) { SetStereoFrame(left, right, frame->sample_rate_hz_, frame); } -// Sets the signal value to increase by |ch1|, |ch2|, |ch3|, |ch4| with every +// Sets the signal value to increase by `ch1`, `ch2`, `ch3`, `ch4` with every // sample in each channel respectively. void SetQuadFrame(float ch1, float ch2, @@ -111,8 +111,8 @@ void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) { EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_); } -// Computes the best SNR based on the error between |ref_frame| and -// |test_frame|. It allows for up to a |max_delay| in samples between the +// Computes the best SNR based on the error between `ref_frame` and +// `test_frame`. It allows for up to a `max_delay` in samples between the // signals to compensate for the resampling delay. float ComputeSNR(const AudioFrame& ref_frame, const AudioFrame& test_frame, diff --git a/audio/utility/audio_frame_operations.cc b/audio/utility/audio_frame_operations.cc index e13a09bace..8f3f37a037 100644 --- a/audio/utility/audio_frame_operations.cc +++ b/audio/utility/audio_frame_operations.cc @@ -222,14 +222,14 @@ void AudioFrameOperations::Mute(AudioFrame* frame, size_t end = count; float start_g = 0.0f; if (current_frame_muted) { - // Fade out the last |count| samples of frame. + // Fade out the last `count` samples of frame. RTC_DCHECK(!previous_frame_muted); start = frame->samples_per_channel_ - count; end = frame->samples_per_channel_; start_g = 1.0f; inc = -inc; } else { - // Fade in the first |count| samples of frame. + // Fade in the first `count` samples of frame. RTC_DCHECK(previous_frame_muted); } diff --git a/audio/utility/audio_frame_operations.h b/audio/utility/audio_frame_operations.h index 2f1540bcf5..7e954dfde9 100644 --- a/audio/utility/audio_frame_operations.h +++ b/audio/utility/audio_frame_operations.h @@ -24,40 +24,40 @@ namespace webrtc { // than a class. class AudioFrameOperations { public: - // Add samples in |frame_to_add| with samples in |result_frame| - // putting the results in |results_frame|. The fields - // |vad_activity_| and |speech_type_| of the result frame are - // updated. If |result_frame| is empty (|samples_per_channel_|==0), - // the samples in |frame_to_add| are added to it. The number of + // Add samples in `frame_to_add` with samples in `result_frame` + // putting the results in `results_frame`. The fields + // `vad_activity_` and `speech_type_` of the result frame are + // updated. If `result_frame` is empty (`samples_per_channel_`==0), + // the samples in `frame_to_add` are added to it. The number of // channels and number of samples per channel must match except when - // |result_frame| is empty. + // `result_frame` is empty. static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame); // |frame.num_channels_| will be updated. This version checks for sufficient - // buffer size and that |num_channels_| is mono. Use UpmixChannels + // buffer size and that `num_channels_` is mono. Use UpmixChannels // instead. TODO(bugs.webrtc.org/8649): remove. ABSL_DEPRECATED("bugs.webrtc.org/8649") static int MonoToStereo(AudioFrame* frame); // |frame.num_channels_| will be updated. This version checks that - // |num_channels_| is stereo. Use DownmixChannels + // `num_channels_` is stereo. Use DownmixChannels // instead. TODO(bugs.webrtc.org/8649): remove. ABSL_DEPRECATED("bugs.webrtc.org/8649") static int StereoToMono(AudioFrame* frame); - // Downmixes 4 channels |src_audio| to stereo |dst_audio|. This is an in-place - // operation, meaning |src_audio| and |dst_audio| may point to the same + // Downmixes 4 channels `src_audio` to stereo `dst_audio`. This is an in-place + // operation, meaning `src_audio` and `dst_audio` may point to the same // buffer. static void QuadToStereo(const int16_t* src_audio, size_t samples_per_channel, int16_t* dst_audio); // |frame.num_channels_| will be updated. This version checks that - // |num_channels_| is 4 channels. + // `num_channels_` is 4 channels. static int QuadToStereo(AudioFrame* frame); - // Downmixes |src_channels| |src_audio| to |dst_channels| |dst_audio|. - // This is an in-place operation, meaning |src_audio| and |dst_audio| + // Downmixes `src_channels` `src_audio` to `dst_channels` `dst_audio`. + // This is an in-place operation, meaning `src_audio` and `dst_audio` // may point to the same buffer. Supported channel combinations are // Stereo to Mono, Quad to Mono, and Quad to Stereo. static void DownmixChannels(const int16_t* src_audio, @@ -67,26 +67,26 @@ class AudioFrameOperations { int16_t* dst_audio); // |frame.num_channels_| will be updated. This version checks that - // |num_channels_| and |dst_channels| are valid and performs relevant downmix. + // `num_channels_` and `dst_channels` are valid and performs relevant downmix. // Supported channel combinations are N channels to Mono, and Quad to Stereo. static void DownmixChannels(size_t dst_channels, AudioFrame* frame); // |frame.num_channels_| will be updated. This version checks that - // |num_channels_| and |dst_channels| are valid and performs relevant + // `num_channels_` and `dst_channels` are valid and performs relevant // downmix. Supported channel combinations are Mono to N // channels. The single channel is replicated. static void UpmixChannels(size_t target_number_of_channels, AudioFrame* frame); - // Swap the left and right channels of |frame|. Fails silently if |frame| is + // Swap the left and right channels of `frame`. Fails silently if `frame` is // not stereo. static void SwapStereoChannels(AudioFrame* frame); - // Conditionally zero out contents of |frame| for implementing audio mute: - // |previous_frame_muted| && |current_frame_muted| - Zero out whole frame. - // |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start. - // !|previous_frame_muted| && |current_frame_muted| - Fade-out at frame end. - // !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched. + // Conditionally zero out contents of `frame` for implementing audio mute: + // `previous_frame_muted` && `current_frame_muted` - Zero out whole frame. + // `previous_frame_muted` && !`current_frame_muted` - Fade-in at frame start. + // !`previous_frame_muted` && `current_frame_muted` - Fade-out at frame end. + // !`previous_frame_muted` && !`current_frame_muted` - Leave frame untouched. static void Mute(AudioFrame* frame, bool previous_frame_muted, bool current_frame_muted); @@ -94,7 +94,7 @@ class AudioFrameOperations { // Zero out contents of frame. static void Mute(AudioFrame* frame); - // Halve samples in |frame|. + // Halve samples in `frame`. static void ApplyHalfGain(AudioFrame* frame); static int Scale(float left, float right, AudioFrame* frame); diff --git a/audio/utility/channel_mixer.cc b/audio/utility/channel_mixer.cc index 8867a3eed4..0f1e663873 100644 --- a/audio/utility/channel_mixer.cc +++ b/audio/utility/channel_mixer.cc @@ -90,7 +90,7 @@ void ChannelMixer::Transform(AudioFrame* frame) { frame->num_channels_ = output_channels_; frame->channel_layout_ = output_layout_; - // Copy the output result to the audio frame in |frame|. + // Copy the output result to the audio frame in `frame`. memcpy( frame->mutable_data(), out_audio, sizeof(int16_t) * frame->samples_per_channel() * frame->num_channels()); diff --git a/audio/utility/channel_mixer.h b/audio/utility/channel_mixer.h index 8b6b7f517d..2dea8eb45b 100644 --- a/audio/utility/channel_mixer.h +++ b/audio/utility/channel_mixer.h @@ -38,8 +38,8 @@ class ChannelMixer { ChannelMixer(ChannelLayout input_layout, ChannelLayout output_layout); ~ChannelMixer(); - // Transforms all input channels corresponding to the selected |input_layout| - // to the number of channels in the selected |output_layout|. + // Transforms all input channels corresponding to the selected `input_layout` + // to the number of channels in the selected `output_layout`. // Example usage (downmix from stereo to mono): // // ChannelMixer mixer(CHANNEL_LAYOUT_STEREO, CHANNEL_LAYOUT_MONO); @@ -69,11 +69,11 @@ class ChannelMixer { // 1D array used as temporary storage during the transformation. std::unique_ptr audio_vector_; - // Number of elements allocated for |audio_vector_|. + // Number of elements allocated for `audio_vector_`. size_t audio_vector_size_ = 0; // Optimization case for when we can simply remap the input channels to output - // channels, i.e., when all scaling factors in |matrix_| equals 1.0. + // channels, i.e., when all scaling factors in `matrix_` equals 1.0. bool remapping_; // Delete the copy constructor and assignment operator. diff --git a/audio/utility/channel_mixing_matrix.cc b/audio/utility/channel_mixing_matrix.cc index 4baff8bfba..1244653f63 100644 --- a/audio/utility/channel_mixing_matrix.cc +++ b/audio/utility/channel_mixing_matrix.cc @@ -274,7 +274,7 @@ bool ChannelMixingMatrix::CreateTransformationMatrix( // All channels should now be accounted for. RTC_DCHECK(unaccounted_inputs_.empty()); - // See if the output |matrix_| is simply a remapping matrix. If each input + // See if the output `matrix_` is simply a remapping matrix. If each input // channel maps to a single output channel we can simply remap. Doing this // programmatically is less fragile than logic checks on channel mappings. for (int output_ch = 0; output_ch < output_channels_; ++output_ch) { @@ -287,7 +287,7 @@ bool ChannelMixingMatrix::CreateTransformationMatrix( } } - // If we've gotten here, |matrix_| is simply a remapping. + // If we've gotten here, `matrix_` is simply a remapping. return true; } diff --git a/audio/utility/channel_mixing_matrix.h b/audio/utility/channel_mixing_matrix.h index 7aef47b3b2..ee00860846 100644 --- a/audio/utility/channel_mixing_matrix.h +++ b/audio/utility/channel_mixing_matrix.h @@ -29,7 +29,7 @@ class ChannelMixingMatrix { // Create the transformation matrix of input channels to output channels. // Updates the empty matrix with the transformation, and returns true // if the transformation is just a remapping of channels (no mixing). - // The size of |matrix| is |output_channels| x |input_channels|, i.e., the + // The size of `matrix` is `output_channels` x `input_channels`, i.e., the // number of rows equals the number of output channels and the number of // columns corresponds to the number of input channels. // This file is derived from Chromium's media/base/channel_mixing_matrix.h. @@ -55,14 +55,14 @@ class ChannelMixingMatrix { void AccountFor(Channels ch); bool IsUnaccounted(Channels ch) const; - // Helper methods for checking if |ch| exists in either |input_layout_| or - // |output_layout_| respectively. + // Helper methods for checking if `ch` exists in either `input_layout_` or + // `output_layout_` respectively. bool HasInputChannel(Channels ch) const; bool HasOutputChannel(Channels ch) const; - // Helper methods for updating |matrix_| with the proper value for - // mixing |input_ch| into |output_ch|. MixWithoutAccounting() does not - // remove the channel from |unaccounted_inputs_|. + // Helper methods for updating `matrix_` with the proper value for + // mixing `input_ch` into `output_ch`. MixWithoutAccounting() does not + // remove the channel from `unaccounted_inputs_`. void Mix(Channels input_ch, Channels output_ch, float scale); void MixWithoutAccounting(Channels input_ch, Channels output_ch, float scale); diff --git a/audio/voip/audio_channel.cc b/audio/voip/audio_channel.cc index b4a50eec12..4650d195f2 100644 --- a/audio/voip/audio_channel.cc +++ b/audio/voip/audio_channel.cc @@ -75,7 +75,7 @@ AudioChannel::~AudioChannel() { audio_mixer_->RemoveSource(ingress_.get()); - // TODO(bugs.webrtc.org/11581): unclear if we still need to clear |egress_| + // TODO(bugs.webrtc.org/11581): unclear if we still need to clear `egress_` // here. egress_.reset(); ingress_.reset(); diff --git a/audio/voip/audio_egress.h b/audio/voip/audio_egress.h index a39c7e225a..989e5bda59 100644 --- a/audio/voip/audio_egress.h +++ b/audio/voip/audio_egress.h @@ -52,7 +52,7 @@ class AudioEgress : public AudioSender, public AudioPacketizationCallback { // Set the encoder format and payload type for AudioCodingModule. // It's possible to change the encoder type during its active usage. - // |payload_type| must be the type that is negotiated with peer through + // `payload_type` must be the type that is negotiated with peer through // offer/answer. void SetEncoder(int payload_type, const SdpAudioFormat& encoder_format, @@ -84,7 +84,7 @@ class AudioEgress : public AudioSender, public AudioPacketizationCallback { // 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. // This will return true when requested dtmf event is successfully scheduled // otherwise false when the dtmf queue reached maximum of 20 events. @@ -139,7 +139,7 @@ class AudioEgress : public AudioSender, public AudioPacketizationCallback { // newly received audio frame from AudioTransport. uint32_t frame_rtp_timestamp_ = 0; - // Flag to track mute state from caller. |previously_muted_| is used to + // Flag to track mute state from caller. `previously_muted_` is used to // track previous state as part of input to AudioFrameOperations::Mute // to implement fading effect when (un)mute is invoked. bool mute_ = false; diff --git a/audio/voip/voip_core.cc b/audio/voip/voip_core.cc index fd66379f4a..8df1c594aa 100644 --- a/audio/voip/voip_core.cc +++ b/audio/voip/voip_core.cc @@ -55,7 +55,7 @@ VoipCore::VoipCore(rtc::scoped_refptr encoder_factory, } bool VoipCore::InitializeIfNeeded() { - // |audio_device_module_| internally owns a lock and the whole logic here + // `audio_device_module_` internally owns a lock and the whole logic here // needs to be executed atomically once using another lock in VoipCore. // Further changes in this method will need to make sure that no deadlock is // introduced in the future. @@ -178,7 +178,7 @@ VoipResult VoipCore::ReleaseChannel(ChannelId channel_id) { } if (no_channels_after_release) { - // TODO(bugs.webrtc.org/11581): unclear if we still need to clear |channel| + // TODO(bugs.webrtc.org/11581): unclear if we still need to clear `channel` // here. channel = nullptr; diff --git a/audio/voip/voip_core.h b/audio/voip/voip_core.h index 359e07272d..439393585c 100644 --- a/audio/voip/voip_core.h +++ b/audio/voip/voip_core.h @@ -53,7 +53,7 @@ class VoipCore : public VoipEngine, public VoipVolumeControl { public: // Construct VoipCore with provided arguments. - // ProcessThread implementation can be injected by |process_thread| + // ProcessThread implementation can be injected by `process_thread` // (mainly for testing purpose) and when set to nullptr, default // implementation will be used. VoipCore(rtc::scoped_refptr encoder_factory, @@ -128,7 +128,7 @@ class VoipCore : public VoipEngine, // mode. Therefore it would be better to delay the logic as late as possible. bool InitializeIfNeeded(); - // Fetches the corresponding AudioChannel assigned with given |channel|. + // Fetches the corresponding AudioChannel assigned with given `channel`. // Returns nullptr if not found. rtc::scoped_refptr GetChannel(ChannelId channel_id); @@ -144,15 +144,15 @@ class VoipCore : public VoipEngine, std::unique_ptr task_queue_factory_; // Synchronization is handled internally by AudioProcessing. - // Must be placed before |audio_device_module_| for proper destruction. + // Must be placed before `audio_device_module_` for proper destruction. rtc::scoped_refptr audio_processing_; // Synchronization is handled internally by AudioMixer. - // Must be placed before |audio_device_module_| for proper destruction. + // Must be placed before `audio_device_module_` for proper destruction. rtc::scoped_refptr audio_mixer_; // Synchronization is handled internally by AudioTransportImpl. - // Must be placed before |audio_device_module_| for proper destruction. + // Must be placed before `audio_device_module_` for proper destruction. std::unique_ptr audio_transport_; // Synchronization is handled internally by AudioDeviceModule.