Use backticks not vertical bars to denote variables in comments for /audio
Bug: webrtc:12338
Change-Id: Ief89269aa39d0cb6749a1c6cc995ce8830ca327f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226942
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34564}
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index e0b15dc..b407508 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -189,7 +189,7 @@
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<AudioSendStream::TargetAudioBitrateConstraints>
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 357e080..db42efc 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -172,7 +172,7 @@
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 @@
::testing::NiceMock<MockRtpRtcpInterface> rtp_rtcp_;
::testing::NiceMock<MockLimitObserver> 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<AudioEncoder> audio_encoder_;
diff --git a/audio/audio_transport_impl.cc b/audio/audio_transport_impl.cc
index 8710ced..2a80ea8 100644
--- a/audio/audio_transport_impl.cc
+++ b/audio/audio_transport_impl.cc
@@ -64,8 +64,8 @@
}
}
-// 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<int16_t>* resampler,
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 57269cd..3ca3b51b 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -429,8 +429,8 @@
}
// 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 @@
// 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 f59834d..04ad7c4 100644
--- a/audio/channel_receive_frame_transformer_delegate.h
+++ b/audio/channel_receive_frame_transformer_delegate.h
@@ -23,7 +23,7 @@
// 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 @@
rtc::scoped_refptr<FrameTransformerInterface> 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 @@
std::unique_ptr<TransformableFrameInterface> 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<TransformableFrameInterface> frame) const;
protected:
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 67391af..663b947 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -98,12 +98,12 @@
std::unique_ptr<AudioFrame> 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 9b7eb33..6d9f0a8 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 @@
rtc::scoped_refptr<FrameTransformerInterface> 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 @@
void OnTransformedFrame(
std::unique_ptr<TransformableFrameInterface> 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<TransformableFrameInterface> frame) const;
protected:
diff --git a/audio/null_audio_poller.cc b/audio/null_audio_poller.cc
index 22f575d..de2c5ca 100644
--- a/audio/null_audio_poller.cc
+++ b/audio/null_audio_poller.cc
@@ -47,7 +47,7 @@
// 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 a45270b..bd8da76 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<int16_t>* 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 d2155a6..a80476e 100644
--- a/audio/remix_resample_unittest.cc
+++ b/audio/remix_resample_unittest.cc
@@ -43,7 +43,7 @@
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 @@
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 @@
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 @@
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 e13a09b..8f3f37a 100644
--- a/audio/utility/audio_frame_operations.cc
+++ b/audio/utility/audio_frame_operations.cc
@@ -222,14 +222,14 @@
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 2f1540b..7e954df 100644
--- a/audio/utility/audio_frame_operations.h
+++ b/audio/utility/audio_frame_operations.h
@@ -24,40 +24,40 @@
// 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 @@
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 @@
// 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 8867a3e..0f1e663 100644
--- a/audio/utility/channel_mixer.cc
+++ b/audio/utility/channel_mixer.cc
@@ -90,7 +90,7 @@
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 8b6b7f5..2dea8eb 100644
--- a/audio/utility/channel_mixer.h
+++ b/audio/utility/channel_mixer.h
@@ -38,8 +38,8 @@
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 @@
// 1D array used as temporary storage during the transformation.
std::unique_ptr<int16_t[]> 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 4baff8b..1244653 100644
--- a/audio/utility/channel_mixing_matrix.cc
+++ b/audio/utility/channel_mixing_matrix.cc
@@ -274,7 +274,7 @@
// 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 @@
}
}
- // 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 7aef47b..ee00860 100644
--- a/audio/utility/channel_mixing_matrix.h
+++ b/audio/utility/channel_mixing_matrix.h
@@ -29,7 +29,7 @@
// 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 @@
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 b4a50ee..4650d19 100644
--- a/audio/voip/audio_channel.cc
+++ b/audio/voip/audio_channel.cc
@@ -75,7 +75,7 @@
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 a39c7e2..989e5bd 100644
--- a/audio/voip/audio_egress.h
+++ b/audio/voip/audio_egress.h
@@ -52,7 +52,7 @@
// 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 @@
// 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 @@
// 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 fd66379..8df1c59 100644
--- a/audio/voip/voip_core.cc
+++ b/audio/voip/voip_core.cc
@@ -55,7 +55,7 @@
}
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 @@
}
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 359e072..4393935 100644
--- a/audio/voip/voip_core.h
+++ b/audio/voip/voip_core.h
@@ -53,7 +53,7 @@
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<AudioEncoderFactory> encoder_factory,
@@ -128,7 +128,7 @@
// 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<AudioChannel> GetChannel(ChannelId channel_id);
@@ -144,15 +144,15 @@
std::unique_ptr<TaskQueueFactory> 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<AudioProcessing> 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<AudioMixer> 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<AudioTransportImpl> audio_transport_;
// Synchronization is handled internally by AudioDeviceModule.
