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

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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}
This commit is contained in:
Artem Titov
2021-07-28 20:50:03 +02:00
committed by WebRTC LUCI CQ
parent dc6801c618
commit 0b489303d2
102 changed files with 483 additions and 483 deletions
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50
modules/audio_processing/include/audio_processing.h
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@ -53,7 +53,7 @@ class CustomAudioAnalyzer;
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 @@ struct ExperimentalNs {
//
// 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 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
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 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
// 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 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
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 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
}
// 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 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
//
// 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 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
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 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
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 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
// 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 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
// 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 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
// 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