Use backticks not vertical bars to denote variables in comments for /modules/audio_processing
Bug: webrtc:12338
Change-Id: I85bff694dd2ead83c939c4d1945eff82e1296001
No-Presubmit: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227161
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34690}
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 64b1b5d..047776b 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -53,7 +53,7 @@
class CustomProcessing;
// Use to enable experimental gain control (AGC). At startup the experimental
-// AGC moves the microphone volume up to |startup_min_volume| if the current
+// AGC moves the microphone volume up to `startup_min_volume` if the current
// microphone volume is set too low. The value is clamped to its operating range
// [12, 255]. Here, 255 maps to 100%.
//
@@ -99,8 +99,8 @@
//
// APM operates on two audio streams on a frame-by-frame basis. Frames of the
// primary stream, on which all processing is applied, are passed to
-// |ProcessStream()|. Frames of the reverse direction stream are passed to
-// |ProcessReverseStream()|. On the client-side, this will typically be the
+// `ProcessStream()`. Frames of the reverse direction stream are passed to
+// `ProcessReverseStream()`. On the client-side, this will typically be the
// near-end (capture) and far-end (render) streams, respectively. APM should be
// placed in the signal chain as close to the audio hardware abstraction layer
// (HAL) as possible.
@@ -264,7 +264,7 @@
bool enabled = false;
} transient_suppression;
- // Enables reporting of |voice_detected| in webrtc::AudioProcessingStats.
+ // Enables reporting of `voice_detected` in webrtc::AudioProcessingStats.
struct VoiceDetection {
bool enabled = false;
} voice_detection;
@@ -377,7 +377,7 @@
// Enables the next generation AGC functionality. This feature replaces the
// standard methods of gain control in the previous AGC. Enabling this
// submodule enables an adaptive digital AGC followed by a limiter. By
- // setting |fixed_gain_db|, the limiter can be turned into a compressor that
+ // setting `fixed_gain_db`, the limiter can be turned into a compressor that
// first applies a fixed gain. The adaptive digital AGC can be turned off by
// setting |adaptive_digital_mode=false|.
struct RTC_EXPORT GainController2 {
@@ -425,7 +425,7 @@
bool enabled = true;
} residual_echo_detector;
- // Enables reporting of |output_rms_dbfs| in webrtc::AudioProcessingStats.
+ // Enables reporting of `output_rms_dbfs` in webrtc::AudioProcessingStats.
struct LevelEstimation {
bool enabled = false;
} level_estimation;
@@ -501,7 +501,7 @@
}
// Creates a runtime setting to notify play-out (aka render) volume changes.
- // |volume| is the unnormalized volume, the maximum of which
+ // `volume` is the unnormalized volume, the maximum of which
static RuntimeSetting CreatePlayoutVolumeChange(int volume) {
return {Type::kPlayoutVolumeChange, volume};
}
@@ -562,13 +562,13 @@
//
// It is also not necessary to call if the audio parameters (sample
// rate and number of channels) have changed. Passing updated parameters
- // directly to |ProcessStream()| and |ProcessReverseStream()| is permissible.
+ // directly to `ProcessStream()` and `ProcessReverseStream()` is permissible.
// If the parameters are known at init-time though, they may be provided.
// TODO(webrtc:5298): Change to return void.
virtual int Initialize() = 0;
// The int16 interfaces require:
- // - only |NativeRate|s be used
+ // - only `NativeRate`s be used
// - that the input, output and reverse rates must match
// - that |processing_config.output_stream()| matches
// |processing_config.input_stream()|.
@@ -616,7 +616,7 @@
virtual bool PostRuntimeSetting(RuntimeSetting setting) = 0;
// Accepts and produces a 10 ms frame interleaved 16 bit integer audio as
- // specified in |input_config| and |output_config|. |src| and |dest| may use
+ // specified in `input_config` and `output_config`. `src` and `dest` may use
// the same memory, if desired.
virtual int ProcessStream(const int16_t* const src,
const StreamConfig& input_config,
@@ -624,35 +624,35 @@
int16_t* const dest) = 0;
// Accepts deinterleaved float audio with the range [-1, 1]. Each element of
- // |src| points to a channel buffer, arranged according to |input_stream|. At
- // output, the channels will be arranged according to |output_stream| in
- // |dest|.
+ // `src` points to a channel buffer, arranged according to `input_stream`. At
+ // output, the channels will be arranged according to `output_stream` in
+ // `dest`.
//
- // The output must have one channel or as many channels as the input. |src|
- // and |dest| may use the same memory, if desired.
+ // The output must have one channel or as many channels as the input. `src`
+ // and `dest` may use the same memory, if desired.
virtual int ProcessStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) = 0;
// Accepts and produces a 10 ms frame of interleaved 16 bit integer audio for
- // the reverse direction audio stream as specified in |input_config| and
- // |output_config|. |src| and |dest| may use the same memory, if desired.
+ // the reverse direction audio stream as specified in `input_config` and
+ // `output_config`. `src` and `dest` may use the same memory, if desired.
virtual int ProcessReverseStream(const int16_t* const src,
const StreamConfig& input_config,
const StreamConfig& output_config,
int16_t* const dest) = 0;
// Accepts deinterleaved float audio with the range [-1, 1]. Each element of
- // |data| points to a channel buffer, arranged according to |reverse_config|.
+ // `data` points to a channel buffer, arranged according to `reverse_config`.
virtual int ProcessReverseStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) = 0;
// Accepts deinterleaved float audio with the range [-1, 1]. Each element
- // of |data| points to a channel buffer, arranged according to
- // |reverse_config|.
+ // of `data` points to a channel buffer, arranged according to
+ // `reverse_config`.
virtual int AnalyzeReverseStream(const float* const* data,
const StreamConfig& reverse_config) = 0;
@@ -675,7 +675,7 @@
// This must be called if and only if echo processing is enabled.
//
- // Sets the |delay| in ms between ProcessReverseStream() receiving a far-end
+ // Sets the `delay` in ms between ProcessReverseStream() receiving a far-end
// frame and ProcessStream() receiving a near-end frame containing the
// corresponding echo. On the client-side this can be expressed as
// delay = (t_render - t_analyze) + (t_process - t_capture)
@@ -695,10 +695,10 @@
// Creates and attaches an webrtc::AecDump for recording debugging
// information.
- // The |worker_queue| may not be null and must outlive the created
+ // The `worker_queue` may not be null and must outlive the created
// AecDump instance. |max_log_size_bytes == -1| means the log size
- // will be unlimited. |handle| may not be null. The AecDump takes
- // responsibility for |handle| and closes it in the destructor. A
+ // will be unlimited. `handle` may not be null. The AecDump takes
+ // responsibility for `handle` and closes it in the destructor. A
// return value of true indicates that the file has been
// sucessfully opened, while a value of false indicates that
// opening the file failed.
@@ -726,7 +726,7 @@
// Get audio processing statistics.
virtual AudioProcessingStats GetStatistics() = 0;
- // TODO(webrtc:5298) Deprecated variant. The |has_remote_tracks| argument
+ // TODO(webrtc:5298) Deprecated variant. 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 AudioProcessing, because they only make sense if there is at least