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Use backticks not vertical bars to denote variables in comments for /sdk
Bug: webrtc:12338 Change-Id: Ifaad29ccb63b0f2f3aeefb77dae061ebc7f87e6c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227024 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34561}
This commit is contained in:
Artem Titov
WebRTC LUCI CQ
parent
f0671921a1
commit
d7ac581045
87 changed files with 235 additions and 235 deletions
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@ -158,7 +158,7 @@ public class Camera1Enumerator implements CameraEnumerator {
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return ranges;
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}
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// Returns the camera index for camera with name |deviceName|, or throws IllegalArgumentException
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// Returns the camera index for camera with name `deviceName`, or throws IllegalArgumentException
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// if no such camera can be found.
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static int getCameraIndex(String deviceName) {
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Logging.d(TAG, "getCameraIndex: " + deviceName);
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@ -152,24 +152,24 @@ public class CameraEnumerationAndroid {
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}
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}
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// Prefer a fps range with an upper bound close to |framerate|. Also prefer a fps range with a low
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// Prefer a fps range with an upper bound close to `framerate`. Also prefer a fps range with a low
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// lower bound, to allow the framerate to fluctuate based on lightning conditions.
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public static CaptureFormat.FramerateRange getClosestSupportedFramerateRange(
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List<CaptureFormat.FramerateRange> supportedFramerates, final int requestedFps) {
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return Collections.min(
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supportedFramerates, new ClosestComparator<CaptureFormat.FramerateRange>() {
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// Progressive penalty if the upper bound is further away than |MAX_FPS_DIFF_THRESHOLD|
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// Progressive penalty if the upper bound is further away than `MAX_FPS_DIFF_THRESHOLD`
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// from requested.
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private static final int MAX_FPS_DIFF_THRESHOLD = 5000;
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private static final int MAX_FPS_LOW_DIFF_WEIGHT = 1;
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private static final int MAX_FPS_HIGH_DIFF_WEIGHT = 3;
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// Progressive penalty if the lower bound is bigger than |MIN_FPS_THRESHOLD|.
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// Progressive penalty if the lower bound is bigger than `MIN_FPS_THRESHOLD`.
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private static final int MIN_FPS_THRESHOLD = 8000;
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private static final int MIN_FPS_LOW_VALUE_WEIGHT = 1;
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private static final int MIN_FPS_HIGH_VALUE_WEIGHT = 4;
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// Use one weight for small |value| less than |threshold|, and another weight above.
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// Use one weight for small `value` less than `threshold`, and another weight above.
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private int progressivePenalty(int value, int threshold, int lowWeight, int highWeight) {
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return (value < threshold) ? value * lowWeight
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: threshold * lowWeight + (value - threshold) * highWeight;
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@ -48,7 +48,7 @@ public interface CameraVideoCapturer extends VideoCapturer {
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* The callback may be called on an arbitrary thread.
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*/
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public interface CameraSwitchHandler {
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// Invoked on success. |isFrontCamera| is true if the new camera is front facing.
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// Invoked on success. `isFrontCamera` is true if the new camera is front facing.
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void onCameraSwitchDone(boolean isFrontCamera);
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// Invoked on failure, e.g. camera is stopped or only one camera available.
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@ -63,7 +63,7 @@ public class DataChannel {
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public final ByteBuffer data;
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/**
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* Indicates whether |data| contains UTF-8 text or "binary data"
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* Indicates whether `data` contains UTF-8 text or "binary data"
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* (i.e. anything else).
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*/
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public final boolean binary;
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@ -110,7 +110,7 @@ public class DataChannel {
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this.nativeDataChannel = nativeDataChannel;
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}
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/** Register |observer|, replacing any previously-registered observer. */
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/** Register `observer`, replacing any previously-registered observer. */
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public void registerObserver(Observer observer) {
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checkDataChannelExists();
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if (nativeObserver != 0) {
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@ -157,7 +157,7 @@ public class DataChannel {
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nativeClose();
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}
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/** Send |data| to the remote peer; return success. */
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/** Send `data` to the remote peer; return success. */
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public boolean send(Buffer buffer) {
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checkDataChannelExists();
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// TODO(fischman): this could be cleverer about avoiding copies if the
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@ -146,8 +146,8 @@ public interface EglBase {
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}
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/**
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* Create a new context with the specified config attributes, sharing data with |sharedContext|.
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* If |sharedContext| is null, a root context is created. This function will try to create an EGL
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* Create a new context with the specified config attributes, sharing data with `sharedContext`.
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* If `sharedContext` is null, a root context is created. This function will try to create an EGL
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* 1.4 context if possible, and an EGL 1.0 context otherwise.
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*/
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public static EglBase create(@Nullable Context sharedContext, int[] configAttributes) {
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@ -171,7 +171,7 @@ public interface EglBase {
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}
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/**
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* Helper function for creating a plain context, sharing data with |sharedContext|. This function
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* Helper function for creating a plain context, sharing data with `sharedContext`. This function
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* will try to create an EGL 1.4 context if possible, and an EGL 1.0 context otherwise.
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*/
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public static EglBase create(Context sharedContext) {
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@ -111,8 +111,8 @@ public class EglRenderer implements VideoSink {
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protected final String name;
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// |renderThreadHandler| is a handler for communicating with |renderThread|, and is synchronized
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// on |handlerLock|.
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// `renderThreadHandler` is a handler for communicating with `renderThread`, and is synchronized
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// on `handlerLock`.
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private final Object handlerLock = new Object();
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@Nullable private Handler renderThreadHandler;
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@ -136,11 +136,11 @@ public class EglRenderer implements VideoSink {
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private boolean usePresentationTimeStamp;
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private final Matrix drawMatrix = new Matrix();
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// Pending frame to render. Serves as a queue with size 1. Synchronized on |frameLock|.
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// Pending frame to render. Serves as a queue with size 1. Synchronized on `frameLock`.
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private final Object frameLock = new Object();
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@Nullable private VideoFrame pendingFrame;
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// These variables are synchronized on |layoutLock|.
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// These variables are synchronized on `layoutLock`.
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private final Object layoutLock = new Object();
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private float layoutAspectRatio;
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// If true, mirrors the video stream horizontally.
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@ -148,7 +148,7 @@ public class EglRenderer implements VideoSink {
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// If true, mirrors the video stream vertically.
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private boolean mirrorVertically;
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// These variables are synchronized on |statisticsLock|.
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// These variables are synchronized on `statisticsLock`.
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private final Object statisticsLock = new Object();
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// Total number of video frames received in renderFrame() call.
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private int framesReceived;
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@ -198,9 +198,9 @@ public class EglRenderer implements VideoSink {
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}
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/**
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* Initialize this class, sharing resources with |sharedContext|. The custom |drawer| will be used
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* Initialize this class, sharing resources with `sharedContext`. The custom `drawer` will be used
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* for drawing frames on the EGLSurface. This class is responsible for calling release() on
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* |drawer|. It is allowed to call init() to reinitialize the renderer after a previous
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* `drawer`. It is allowed to call init() to reinitialize the renderer after a previous
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* init()/release() cycle. If usePresentationTimeStamp is true, eglPresentationTimeANDROID will be
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* set with the frame timestamps, which specifies desired presentation time and might be useful
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* for e.g. syncing audio and video.
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@ -592,10 +592,10 @@ public class EglRenderer implements VideoSink {
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}
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/**
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* Renders and releases |pendingFrame|.
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* Renders and releases `pendingFrame`.
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*/
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private void renderFrameOnRenderThread() {
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// Fetch and render |pendingFrame|.
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// Fetch and render `pendingFrame`.
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final VideoFrame frame;
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synchronized (frameLock) {
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if (pendingFrame == null) {
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@ -78,16 +78,16 @@ public class GlShader {
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}
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/**
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* Enable and upload a vertex array for attribute |label|. The vertex data is specified in
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* |buffer| with |dimension| number of components per vertex.
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* Enable and upload a vertex array for attribute `label`. The vertex data is specified in
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* `buffer` with `dimension` number of components per vertex.
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*/
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public void setVertexAttribArray(String label, int dimension, FloatBuffer buffer) {
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setVertexAttribArray(label, dimension, 0 /* stride */, buffer);
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}
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/**
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* Enable and upload a vertex array for attribute |label|. The vertex data is specified in
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* |buffer| with |dimension| number of components per vertex and specified |stride|.
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* Enable and upload a vertex array for attribute `label`. The vertex data is specified in
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* `buffer` with `dimension` number of components per vertex and specified `stride`.
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*/
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public void setVertexAttribArray(String label, int dimension, int stride, FloatBuffer buffer) {
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if (program == -1) {
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@ -18,12 +18,12 @@ import java.util.Map;
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// Rtc histograms can be queried through the API, getAndReset().
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// The returned map holds the name of a histogram and its samples.
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//
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// Example of |map| with one histogram:
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// |name|: "WebRTC.Video.InputFramesPerSecond"
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// |min|: 1
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// |max|: 100
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// |bucketCount|: 50
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// |samples|: [30]:1
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// Example of `map` with one histogram:
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// `name`: "WebRTC.Video.InputFramesPerSecond"
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// `min`: 1
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// `max`: 100
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// `bucketCount`: 50
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// `samples`: [30]:1
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//
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// Most histograms are not updated frequently (e.g. most video metrics are an
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// average over the call and recorded when a stream is removed).
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@ -98,9 +98,9 @@ public interface NetworkChangeDetector {
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/**
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* Called when network preference change for a (list of) connection type(s). (e.g WIFI) is
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* |NOT_PREFERRED| or |NEUTRAL|.
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* `NOT_PREFERRED` or `NEUTRAL`.
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*
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* <p>note: |types| is a list of ConnectionTypes, so that all cellular types can be modified in
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* <p>note: `types` is a list of ConnectionTypes, so that all cellular types can be modified in
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* one call.
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*/
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public void onNetworkPreference(List<ConnectionType> types, @NetworkPreference int preference);
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@ -172,7 +172,7 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver implements Netwo
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}
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/**
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* Returns connection type and status information about |network|.
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* Returns connection type and status information about `network`.
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* Only callable on Lollipop and newer releases.
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*/
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@SuppressLint("NewApi")
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@ -186,9 +186,9 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver implements Netwo
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return new NetworkState(false, -1, -1, -1, -1);
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}
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// The general logic of handling a VPN in this method is as follows. getNetworkInfo will
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// return the info of the network with the same id as in |network| when it is registered via
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// ConnectivityManager.registerNetworkAgent in Android. |networkInfo| may or may not indicate
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// the type TYPE_VPN if |network| is a VPN. To reliably detect the VPN interface, we need to
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// return the info of the network with the same id as in `network` when it is registered via
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// ConnectivityManager.registerNetworkAgent in Android. `networkInfo` may or may not indicate
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// the type TYPE_VPN if `network` is a VPN. To reliably detect the VPN interface, we need to
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// query the network capability as below in the case when networkInfo.getType() is not
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// TYPE_VPN. On the other hand when networkInfo.getType() is TYPE_VPN, the only solution so
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// far to obtain the underlying network information is to query the active network interface.
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@ -198,7 +198,7 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver implements Netwo
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// getActiveNetworkInfo may thus give the wrong interface information, and one should note
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// that getActiveNetworkInfo would return the default network interface if the VPN does not
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// specify its underlying networks in the implementation. Therefore, we need further compare
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// |network| to the active network. If they are not the same network, we will have to fall
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// `network` to the active network. If they are not the same network, we will have to fall
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// back to report an unknown network.
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if (networkInfo.getType() != ConnectivityManager.TYPE_VPN) {
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@ -209,15 +209,15 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver implements Netwo
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|| !networkCapabilities.hasTransport(NetworkCapabilities.TRANSPORT_VPN)) {
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return getNetworkState(networkInfo);
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}
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// When |network| is in fact a VPN after querying its capability but |networkInfo| is not of
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// type TYPE_VPN, |networkInfo| contains the info for the underlying network, and we return
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// When `network` is in fact a VPN after querying its capability but `networkInfo` is not of
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// type TYPE_VPN, `networkInfo` contains the info for the underlying network, and we return
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// a NetworkState constructed from it.
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return new NetworkState(networkInfo.isConnected(), ConnectivityManager.TYPE_VPN, -1,
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networkInfo.getType(), networkInfo.getSubtype());
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}
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// When |networkInfo| is of type TYPE_VPN, which implies |network| is a VPN, we return the
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// NetworkState of the active network via getActiveNetworkInfo(), if |network| is the active
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// When `networkInfo` is of type TYPE_VPN, which implies `network` is a VPN, we return the
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// NetworkState of the active network via getActiveNetworkInfo(), if `network` is the active
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// network that supports the VPN. Otherwise, NetworkState of an unknown network with type -1
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// will be returned.
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//
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@ -169,9 +169,9 @@ public class PeerConnection {
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public final String password;
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public final TlsCertPolicy tlsCertPolicy;
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// If the URIs in |urls| only contain IP addresses, this field can be used
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// If the URIs in `urls` only contain IP addresses, this field can be used
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// to indicate the hostname, which may be necessary for TLS (using the SNI
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// extension). If |urls| itself contains the hostname, this isn't
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// extension). If `urls` itself contains the hostname, this isn't
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// necessary.
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public final String hostname;
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@ -1106,7 +1106,7 @@ public class PeerConnection {
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* transceiver will cause future calls to CreateOffer to add a media description
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* for the corresponding transceiver.
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*
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* <p>The initial value of |mid| in the returned transceiver is null. Setting a
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* <p>The initial value of `mid` in the returned transceiver is null. Setting a
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* new session description may change it to a non-null value.
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*
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* <p>https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
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@ -133,7 +133,7 @@ public class PeerConnectionFactory {
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public static class Options {
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// Keep in sync with webrtc/rtc_base/network.h!
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//
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// These bit fields are defined for |networkIgnoreMask| below.
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// These bit fields are defined for `networkIgnoreMask` below.
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static final int ADAPTER_TYPE_UNKNOWN = 0;
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static final int ADAPTER_TYPE_ETHERNET = 1 << 0;
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static final int ADAPTER_TYPE_WIFI = 1 << 1;
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@ -14,8 +14,8 @@ import java.util.Map;
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/**
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* Java version of webrtc::RTCStats. Represents an RTCStats object, as
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* described in https://w3c.github.io/webrtc-stats/. The |id|, |timestampUs|
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* and |type| accessors have the same meaning for this class as for the
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* described in https://w3c.github.io/webrtc-stats/. The `id`, `timestampUs`
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* and `type` accessors have the same meaning for this class as for the
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* RTCStats dictionary. Each RTCStatsReport produced by getStats contains
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* multiple RTCStats objects; one for each underlying object (codec, stream,
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* transport, etc.) that was inspected to produce the stats.
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@ -123,9 +123,9 @@ public class RendererCommon {
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// clipped.
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// SCALE_ASPECT_BALANCED - Compromise between FIT and FILL. Video frame will fill as much as
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// possible of the view while maintaining aspect ratio, under the constraint that at least
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// |BALANCED_VISIBLE_FRACTION| of the frame content will be shown.
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// `BALANCED_VISIBLE_FRACTION` of the frame content will be shown.
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public static enum ScalingType { SCALE_ASPECT_FIT, SCALE_ASPECT_FILL, SCALE_ASPECT_BALANCED }
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// The minimum fraction of the frame content that will be shown for |SCALE_ASPECT_BALANCED|.
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// The minimum fraction of the frame content that will be shown for `SCALE_ASPECT_BALANCED`.
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// This limits excessive cropping when adjusting display size.
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private static float BALANCED_VISIBLE_FRACTION = 0.5625f;
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@ -209,7 +209,7 @@ public class RendererCommon {
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}
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/**
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* Move |matrix| transformation origin to (0.5, 0.5). This is the origin for texture coordinates
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* Move `matrix` transformation origin to (0.5, 0.5). This is the origin for texture coordinates
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* that are in the range 0 to 1.
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*/
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private static void adjustOrigin(float[] matrix) {
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@ -39,7 +39,7 @@ public class RtpSender {
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*
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* @param takeOwnership If true, the RtpSender takes ownership of the track
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* from the caller, and will auto-dispose of it when no
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* longer needed. |takeOwnership| should only be used if
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* longer needed. `takeOwnership` should only be used if
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* the caller owns the track; it is not appropriate when
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* the track is owned by, for example, another RtpSender
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* or a MediaStream.
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@ -42,9 +42,9 @@ public class SurfaceEglRenderer extends EglRenderer implements SurfaceHolder.Cal
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}
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/**
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* Initialize this class, sharing resources with |sharedContext|. The custom |drawer| will be used
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* Initialize this class, sharing resources with `sharedContext`. The custom `drawer` will be used
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* for drawing frames on the EGLSurface. This class is responsible for calling release() on
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* |drawer|. It is allowed to call init() to reinitialize the renderer after a previous
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* `drawer`. It is allowed to call init() to reinitialize the renderer after a previous
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* init()/release() cycle.
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*/
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public void init(final EglBase.Context sharedContext,
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@ -125,7 +125,7 @@ public class SurfaceEglRenderer extends EglRenderer implements SurfaceHolder.Cal
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logD("surfaceChanged: format: " + format + " size: " + width + "x" + height);
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}
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// Update frame dimensions and report any changes to |rendererEvents|.
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// Update frame dimensions and report any changes to `rendererEvents`.
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private void updateFrameDimensionsAndReportEvents(VideoFrame frame) {
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synchronized (layoutLock) {
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if (isRenderingPaused) {
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@ -48,7 +48,7 @@ public class SurfaceTextureHelper {
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private static final String TAG = "SurfaceTextureHelper";
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/**
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* Construct a new SurfaceTextureHelper sharing OpenGL resources with |sharedContext|. A dedicated
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* Construct a new SurfaceTextureHelper sharing OpenGL resources with `sharedContext`. A dedicated
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* thread and handler is created for handling the SurfaceTexture. May return null if EGL fails to
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* initialize a pixel buffer surface and make it current. If alignTimestamps is true, the frame
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* timestamps will be aligned to rtc::TimeNanos(). If frame timestamps are aligned to
|
||||
|
|
@ -66,7 +66,7 @@ public class SurfaceTextureHelper {
|
|||
// The onFrameAvailable() callback will be executed on the SurfaceTexture ctor thread. See:
|
||||
// http://grepcode.com/file/repository.grepcode.com/java/ext/com.google.android/android/5.1.1_r1/android/graphics/SurfaceTexture.java#195.
|
||||
// Therefore, in order to control the callback thread on API lvl < 21, the SurfaceTextureHelper
|
||||
// is constructed on the |handler| thread.
|
||||
// is constructed on the `handler` thread.
|
||||
return ThreadUtils.invokeAtFrontUninterruptibly(handler, new Callable<SurfaceTextureHelper>() {
|
||||
@Nullable
|
||||
@Override
|
||||
|
|
@ -147,7 +147,7 @@ public class SurfaceTextureHelper {
|
|||
@Nullable private final TimestampAligner timestampAligner;
|
||||
private final FrameRefMonitor frameRefMonitor;
|
||||
|
||||
// These variables are only accessed from the |handler| thread.
|
||||
// These variables are only accessed from the `handler` thread.
|
||||
@Nullable private VideoSink listener;
|
||||
// The possible states of this class.
|
||||
private boolean hasPendingTexture;
|
||||
|
|
@ -156,7 +156,7 @@ public class SurfaceTextureHelper {
|
|||
private int frameRotation;
|
||||
private int textureWidth;
|
||||
private int textureHeight;
|
||||
// |pendingListener| is set in setListener() and the runnable is posted to the handler thread.
|
||||
// `pendingListener` is set in setListener() and the runnable is posted to the handler thread.
|
||||
// setListener() is not allowed to be called again before stopListening(), so this is thread safe.
|
||||
@Nullable private VideoSink pendingListener;
|
||||
final Runnable setListenerRunnable = new Runnable() {
|
||||
|
|
@ -223,7 +223,7 @@ public class SurfaceTextureHelper {
|
|||
}
|
||||
|
||||
/**
|
||||
* Start to stream textures to the given |listener|. If you need to change listener, you need to
|
||||
* Start to stream textures to the given `listener`. If you need to change listener, you need to
|
||||
* call stopListening() first.
|
||||
*/
|
||||
public void startListening(final VideoSink listener) {
|
||||
|
|
@ -331,7 +331,7 @@ public class SurfaceTextureHelper {
|
|||
}
|
||||
|
||||
/**
|
||||
* Posts to the correct thread to convert |textureBuffer| to I420.
|
||||
* Posts to the correct thread to convert `textureBuffer` to I420.
|
||||
*
|
||||
* @deprecated Use toI420() instead.
|
||||
*/
|
||||
|
|
|
|||
|
|
@ -64,7 +64,7 @@ public class SurfaceViewRenderer extends SurfaceView
|
|||
}
|
||||
|
||||
/**
|
||||
* Initialize this class, sharing resources with |sharedContext|. It is allowed to call init() to
|
||||
* Initialize this class, sharing resources with `sharedContext`. It is allowed to call init() to
|
||||
* reinitialize the renderer after a previous init()/release() cycle.
|
||||
*/
|
||||
public void init(EglBase.Context sharedContext, RendererCommon.RendererEvents rendererEvents) {
|
||||
|
|
@ -72,9 +72,9 @@ public class SurfaceViewRenderer extends SurfaceView
|
|||
}
|
||||
|
||||
/**
|
||||
* Initialize this class, sharing resources with |sharedContext|. The custom |drawer| will be used
|
||||
* Initialize this class, sharing resources with `sharedContext`. The custom `drawer` will be used
|
||||
* for drawing frames on the EGLSurface. This class is responsible for calling release() on
|
||||
* |drawer|. It is allowed to call init() to reinitialize the renderer after a previous
|
||||
* `drawer`. It is allowed to call init() to reinitialize the renderer after a previous
|
||||
* init()/release() cycle.
|
||||
*/
|
||||
public void init(final EglBase.Context sharedContext,
|
||||
|
|
|
|||
|
|
@ -31,7 +31,7 @@ public class TimestampAligner {
|
|||
|
||||
/**
|
||||
* Translates camera timestamps to the same timescale as is used by rtc::TimeNanos().
|
||||
* |cameraTimeNs| is assumed to be accurate, but with an unknown epoch and clock drift. Returns
|
||||
* `cameraTimeNs` is assumed to be accurate, but with an unknown epoch and clock drift. Returns
|
||||
* the translated timestamp.
|
||||
*/
|
||||
public long translateTimestamp(long cameraTimeNs) {
|
||||
|
|
|
|||
|
|
@ -238,7 +238,7 @@ public interface VideoEncoder {
|
|||
|
||||
public interface Callback {
|
||||
/**
|
||||
* Old encoders assume that the byte buffer held by |frame| is not accessed after the call to
|
||||
* Old encoders assume that the byte buffer held by `frame` is not accessed after the call to
|
||||
* this method returns. If the pipeline downstream needs to hold on to the buffer, it then has
|
||||
* to make its own copy. We want to move to a model where no copying is needed, and instead use
|
||||
* retain()/release() to signal to the encoder when it is safe to reuse the buffer.
|
||||
|
|
|
|||
|
|
@ -60,8 +60,8 @@ public class VideoFrame implements RefCounted {
|
|||
@Override @CalledByNative("Buffer") void release();
|
||||
|
||||
/**
|
||||
* Crops a region defined by |cropx|, |cropY|, |cropWidth| and |cropHeight|. Scales it to size
|
||||
* |scaleWidth| x |scaleHeight|.
|
||||
* Crops a region defined by `cropx`, `cropY`, `cropWidth` and `cropHeight`. Scales it to size
|
||||
* `scaleWidth` x `scaleHeight`.
|
||||
*/
|
||||
@CalledByNative("Buffer")
|
||||
Buffer cropAndScale(
|
||||
|
|
|
|||
|
|
@ -61,7 +61,7 @@ public class VideoFrameDrawer {
|
|||
@Nullable private int[] yuvTextures;
|
||||
|
||||
/**
|
||||
* Upload |planes| into OpenGL textures, taking stride into consideration.
|
||||
* Upload `planes` into OpenGL textures, taking stride into consideration.
|
||||
*
|
||||
* @return Array of three texture indices corresponding to Y-, U-, and V-plane respectively.
|
||||
*/
|
||||
|
|
@ -145,8 +145,8 @@ public class VideoFrameDrawer {
|
|||
private int renderWidth;
|
||||
private int renderHeight;
|
||||
|
||||
// Calculate the frame size after |renderMatrix| is applied. Stores the output in member variables
|
||||
// |renderWidth| and |renderHeight| to avoid allocations since this function is called for every
|
||||
// Calculate the frame size after `renderMatrix` is applied. Stores the output in member variables
|
||||
// `renderWidth` and `renderHeight` to avoid allocations since this function is called for every
|
||||
// frame.
|
||||
private void calculateTransformedRenderSize(
|
||||
int frameWidth, int frameHeight, @Nullable Matrix renderMatrix) {
|
||||
|
|
@ -155,7 +155,7 @@ public class VideoFrameDrawer {
|
|||
renderHeight = frameHeight;
|
||||
return;
|
||||
}
|
||||
// Transform the texture coordinates (in the range [0, 1]) according to |renderMatrix|.
|
||||
// Transform the texture coordinates (in the range [0, 1]) according to `renderMatrix`.
|
||||
renderMatrix.mapPoints(dstPoints, srcPoints);
|
||||
|
||||
// Multiply with the width and height to get the positions in terms of pixels.
|
||||
|
|
|
|||
|
|
@ -153,7 +153,7 @@ public final class YuvConverter {
|
|||
// +----+----+
|
||||
//
|
||||
// In memory, we use the same stride for all of Y, U and V. The
|
||||
// U data starts at offset |height| * |stride| from the Y data,
|
||||
// U data starts at offset `height` * `stride` from the Y data,
|
||||
// and the V data starts at at offset |stride/2| from the U
|
||||
// data, with rows of U and V data alternating.
|
||||
//
|
||||
|
|
@ -161,12 +161,12 @@ public final class YuvConverter {
|
|||
// a single byte per pixel (EGL10.EGL_COLOR_BUFFER_TYPE,
|
||||
// EGL10.EGL_LUMINANCE_BUFFER,), but that seems to be
|
||||
// unsupported by devices. So do the following hack: Allocate an
|
||||
// RGBA buffer, of width |stride|/4. To render each of these
|
||||
// RGBA buffer, of width `stride`/4. To render each of these
|
||||
// large pixels, sample the texture at 4 different x coordinates
|
||||
// and store the results in the four components.
|
||||
//
|
||||
// Since the V data needs to start on a boundary of such a
|
||||
// larger pixel, it is not sufficient that |stride| is even, it
|
||||
// larger pixel, it is not sufficient that `stride` is even, it
|
||||
// has to be a multiple of 8 pixels.
|
||||
final int frameWidth = preparedBuffer.getWidth();
|
||||
final int frameHeight = preparedBuffer.getHeight();
|
||||
|
|
|
|||
|
|
@ -541,7 +541,7 @@ class CameraVideoCapturerTestFixtures {
|
|||
capturerInstance.capturer.stopCapture();
|
||||
capturerInstance.observer.releaseFrame();
|
||||
|
||||
// We can't change |capturer| at this point, but we should not crash.
|
||||
// We can't change `capturer` at this point, but we should not crash.
|
||||
capturerInstance.capturer.switchCamera(null /* switchEventsHandler */);
|
||||
capturerInstance.capturer.changeCaptureFormat(DEFAULT_WIDTH, DEFAULT_HEIGHT, DEFAULT_FPS);
|
||||
|
||||
|
|
|
|||
|
|
@ -145,7 +145,7 @@ public class PeerConnectionTest {
|
|||
// TODO(fischman) MOAR test ideas:
|
||||
// - Test that PC.removeStream() works; requires a second
|
||||
// createOffer/createAnswer dance.
|
||||
// - audit each place that uses |constraints| for specifying non-trivial
|
||||
// - audit each place that uses `constraints` for specifying non-trivial
|
||||
// constraints (and ensure they're honored).
|
||||
// - test error cases
|
||||
// - ensure reasonable coverage of jni code is achieved. Coverage is
|
||||
|
|
|
|||
|
|
@ -123,8 +123,8 @@ public class SurfaceTextureHelperTest {
|
|||
surfaceTextureHelper.startListening(listener);
|
||||
surfaceTextureHelper.setTextureSize(width, height);
|
||||
|
||||
// Create resources for stubbing an OES texture producer. |eglOesBase| has the SurfaceTexture in
|
||||
// |surfaceTextureHelper| as the target EGLSurface.
|
||||
// Create resources for stubbing an OES texture producer. `eglOesBase` has the SurfaceTexture in
|
||||
// `surfaceTextureHelper` as the target EGLSurface.
|
||||
final EglBase eglOesBase = EglBase.create(eglBase.getEglBaseContext(), EglBase.CONFIG_PLAIN);
|
||||
eglOesBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
|
||||
assertEquals(eglOesBase.surfaceWidth(), width);
|
||||
|
|
@ -191,8 +191,8 @@ public class SurfaceTextureHelperTest {
|
|||
surfaceTextureHelper.startListening(listener);
|
||||
surfaceTextureHelper.setTextureSize(width, height);
|
||||
|
||||
// Create resources for stubbing an OES texture producer. |eglOesBase| has the SurfaceTexture in
|
||||
// |surfaceTextureHelper| as the target EGLSurface.
|
||||
// Create resources for stubbing an OES texture producer. `eglOesBase` has the SurfaceTexture in
|
||||
// `surfaceTextureHelper` as the target EGLSurface.
|
||||
final EglBase eglOesBase = EglBase.create(eglBase.getEglBaseContext(), EglBase.CONFIG_PLAIN);
|
||||
eglOesBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
|
||||
assertEquals(eglOesBase.surfaceWidth(), width);
|
||||
|
|
@ -410,7 +410,7 @@ public class SurfaceTextureHelperTest {
|
|||
eglBase.swapBuffers();
|
||||
listener1.waitForTextureBuffer().release();
|
||||
|
||||
// Stop listening - |listener1| should not receive any textures after this.
|
||||
// Stop listening - `listener1` should not receive any textures after this.
|
||||
surfaceTextureHelper.stopListening();
|
||||
|
||||
// Connect different listener.
|
||||
|
|
@ -423,7 +423,7 @@ public class SurfaceTextureHelperTest {
|
|||
GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
|
||||
eglBase.swapBuffers();
|
||||
|
||||
// Check that |listener2| received the frame, and not |listener1|.
|
||||
// Check that `listener2` received the frame, and not `listener1`.
|
||||
listener2.waitForTextureBuffer().release();
|
||||
listener1.assertNoFrameIsDelivered(/* waitPeriodMs= */ 1);
|
||||
|
||||
|
|
@ -446,8 +446,8 @@ public class SurfaceTextureHelperTest {
|
|||
surfaceTextureHelper.startListening(listener);
|
||||
surfaceTextureHelper.setTextureSize(width, height);
|
||||
|
||||
// Create resources for stubbing an OES texture producer. |eglBase| has the SurfaceTexture in
|
||||
// |surfaceTextureHelper| as the target EGLSurface.
|
||||
// Create resources for stubbing an OES texture producer. `eglBase` has the SurfaceTexture in
|
||||
// `surfaceTextureHelper` as the target EGLSurface.
|
||||
|
||||
eglBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
|
||||
assertEquals(eglBase.surfaceWidth(), width);
|
||||
|
|
|
|||
|
|
@ -153,7 +153,7 @@ public class VideoFrameBufferTest {
|
|||
}
|
||||
|
||||
/**
|
||||
* Create an RGB texture buffer available in |eglContext| with the same pixel content as the given
|
||||
* Create an RGB texture buffer available in `eglContext` with the same pixel content as the given
|
||||
* I420 buffer.
|
||||
*/
|
||||
public static VideoFrame.TextureBuffer createRgbTextureBuffer(
|
||||
|
|
@ -191,7 +191,7 @@ public class VideoFrameBufferTest {
|
|||
}
|
||||
|
||||
/**
|
||||
* Create an OES texture buffer available in |eglContext| with the same pixel content as the given
|
||||
* Create an OES texture buffer available in `eglContext` with the same pixel content as the given
|
||||
* I420 buffer.
|
||||
*/
|
||||
public static VideoFrame.TextureBuffer createOesTextureBuffer(
|
||||
|
|
|
|||
|
|
@ -18,7 +18,7 @@
|
|||
#include "sdk/android/native_api/jni/java_types.h"
|
||||
#include "sdk/android/native_api/jni/scoped_java_ref.h"
|
||||
|
||||
// Abort the process if |jni| has a Java exception pending. This macros uses the
|
||||
// Abort the process if `jni` has a Java exception pending. This macros uses the
|
||||
// comma operator to execute ExceptionDescribe and ExceptionClear ignoring their
|
||||
// return values and sending "" to the error stream.
|
||||
#define CHECK_EXCEPTION(jni) \
|
||||
|
|
|
|||
|
|
@ -30,7 +30,7 @@
|
|||
#include "rtc_base/checks.h"
|
||||
#include "sdk/android/native_api/jni/scoped_java_ref.h"
|
||||
|
||||
// Abort the process if |jni| has a Java exception pending.
|
||||
// Abort the process if `jni` has a Java exception pending.
|
||||
// This macros uses the comma operator to execute ExceptionDescribe
|
||||
// and ExceptionClear ignoring their return values and sending ""
|
||||
// to the error stream.
|
||||
|
|
@ -110,7 +110,7 @@ class Iterable {
|
|||
RTC_DISALLOW_COPY_AND_ASSIGN(Iterable);
|
||||
};
|
||||
|
||||
// Returns true if |obj| == null in Java.
|
||||
// Returns true if `obj` == null in Java.
|
||||
bool IsNull(JNIEnv* jni, const JavaRef<jobject>& obj);
|
||||
|
||||
// Returns the name of a Java enum.
|
||||
|
|
@ -319,7 +319,7 @@ ScopedJavaLocalRef<jobject> NativeToJavaStringMap(JNIEnv* env,
|
|||
return builder.GetJavaMap();
|
||||
}
|
||||
|
||||
// Return a |jlong| that will correctly convert back to |ptr|. This is needed
|
||||
// Return a `jlong` that will correctly convert back to `ptr`. This is needed
|
||||
// because the alternative (of silently passing a 32-bit pointer to a vararg
|
||||
// function expecting a 64-bit param) picks up garbage in the high 32 bits.
|
||||
jlong NativeToJavaPointer(void* ptr);
|
||||
|
|
|
|||
|
|
@ -74,7 +74,7 @@ class JavaRef : public JavaRef<jobject> {
|
|||
template <typename T>
|
||||
class JavaParamRef : public JavaRef<T> {
|
||||
public:
|
||||
// Assumes that |obj| is a parameter passed to a JNI method from Java.
|
||||
// Assumes that `obj` is a parameter passed to a JNI method from Java.
|
||||
// Does not assume ownership as parameters should not be deleted.
|
||||
explicit JavaParamRef(T obj) : JavaRef<T>(obj) {}
|
||||
JavaParamRef(JNIEnv*, T obj) : JavaRef<T>(obj) {}
|
||||
|
|
@ -112,7 +112,7 @@ class ScopedJavaLocalRef : public JavaRef<T> {
|
|||
Reset(other.obj(), OwnershipPolicy::RETAIN);
|
||||
}
|
||||
|
||||
// Assumes that |obj| is a reference to a Java object and takes
|
||||
// Assumes that `obj` is a reference to a Java object and takes
|
||||
// ownership of this reference. This should preferably not be used
|
||||
// outside of JNI helper functions.
|
||||
ScopedJavaLocalRef(JNIEnv* env, T obj) : JavaRef<T>(obj), env_(env) {}
|
||||
|
|
|
|||
|
|
@ -20,7 +20,7 @@
|
|||
|
||||
namespace webrtc {
|
||||
|
||||
// Creates java PeerConnectionFactory with specified |pcf|.
|
||||
// Creates java PeerConnectionFactory with specified `pcf`.
|
||||
jobject NativeToJavaPeerConnectionFactory(
|
||||
JNIEnv* jni,
|
||||
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf,
|
||||
|
|
|
|||
|
|
@ -99,7 +99,7 @@ ABSL_CONST_INIT GlobalMutex g_signal_handler_lock(absl::kConstInit);
|
|||
SignalHandlerOutputState* volatile g_signal_handler_output_state;
|
||||
|
||||
// This function is called iteratively for each stack trace element and stores
|
||||
// the element in the array from |unwind_output_state|.
|
||||
// the element in the array from `unwind_output_state`.
|
||||
_Unwind_Reason_Code UnwindBacktrace(struct _Unwind_Context* unwind_context,
|
||||
void* unwind_output_state) {
|
||||
SignalHandlerOutputState* const output_state =
|
||||
|
|
@ -136,7 +136,7 @@ void SignalHandler(int signum, siginfo_t* info, void* ptr) {
|
|||
|
||||
// Temporarily change the signal handler to a function that records a raw stack
|
||||
// trace and interrupt the given tid. This function will block until the output
|
||||
// thread stack trace has been stored in |params|. The return value is an error
|
||||
// thread stack trace has been stored in `params`. The return value is an error
|
||||
// string on failure and null on success.
|
||||
const char* CaptureRawStacktrace(int pid,
|
||||
int tid,
|
||||
|
|
@ -206,8 +206,8 @@ std::vector<StackTraceElement> FormatStackTrace(
|
|||
std::vector<StackTraceElement> GetStackTrace(int tid) {
|
||||
// Only a thread itself can unwind its stack, so we will interrupt the given
|
||||
// tid with a custom signal handler in order to unwind its stack. The stack
|
||||
// will be recorded to |params| through the use of the global pointer
|
||||
// |g_signal_handler_param|.
|
||||
// will be recorded to `params` through the use of the global pointer
|
||||
// `g_signal_handler_param`.
|
||||
SignalHandlerOutputState params;
|
||||
|
||||
const char* error_string = CaptureRawStacktrace(getpid(), tid, ¶ms);
|
||||
|
|
|
|||
|
|
@ -65,7 +65,7 @@ static const int kFilePlayTimeInSec = 5;
|
|||
static const size_t kBitsPerSample = 16;
|
||||
static const size_t kBytesPerSample = kBitsPerSample / 8;
|
||||
// Run the full-duplex test during this time (unit is in seconds).
|
||||
// Note that first |kNumIgnoreFirstCallbacks| are ignored.
|
||||
// Note that first `kNumIgnoreFirstCallbacks` are ignored.
|
||||
static const int kFullDuplexTimeInSec = 5;
|
||||
// Wait for the callback sequence to stabilize by ignoring this amount of the
|
||||
// initial callbacks (avoids initial FIFO access).
|
||||
|
|
@ -124,7 +124,7 @@ class FileAudioStream : public AudioStreamInterface {
|
|||
void Write(const void* source, size_t num_frames) override {}
|
||||
|
||||
// Read samples from file stored in memory (at construction) and copy
|
||||
// |num_frames| (<=> 10ms) to the |destination| byte buffer.
|
||||
// `num_frames` (<=> 10ms) to the `destination` byte buffer.
|
||||
void Read(void* destination, size_t num_frames) override {
|
||||
memcpy(destination, static_cast<int16_t*>(&file_[file_pos_]),
|
||||
num_frames * sizeof(int16_t));
|
||||
|
|
@ -168,7 +168,7 @@ class FifoAudioStream : public AudioStreamInterface {
|
|||
|
||||
~FifoAudioStream() { Flush(); }
|
||||
|
||||
// Allocate new memory, copy |num_frames| samples from |source| into memory
|
||||
// Allocate new memory, copy `num_frames` samples from `source` into memory
|
||||
// and add pointer to the memory location to end of the list.
|
||||
// Increases the size of the FIFO by one element.
|
||||
void Write(const void* source, size_t num_frames) override {
|
||||
|
|
@ -189,8 +189,8 @@ class FifoAudioStream : public AudioStreamInterface {
|
|||
total_written_elements_ += size;
|
||||
}
|
||||
|
||||
// Read pointer to data buffer from front of list, copy |num_frames| of stored
|
||||
// data into |destination| and delete the utilized memory allocation.
|
||||
// Read pointer to data buffer from front of list, copy `num_frames` of stored
|
||||
// data into `destination` and delete the utilized memory allocation.
|
||||
// Decreases the size of the FIFO by one element.
|
||||
void Read(void* destination, size_t num_frames) override {
|
||||
ASSERT_EQ(num_frames, frames_per_buffer_);
|
||||
|
|
@ -248,7 +248,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
|
|||
rec_count_(0),
|
||||
pulse_time_(0) {}
|
||||
|
||||
// Insert periodic impulses in first two samples of |destination|.
|
||||
// Insert periodic impulses in first two samples of `destination`.
|
||||
void Read(void* destination, size_t num_frames) override {
|
||||
ASSERT_EQ(num_frames, frames_per_buffer_);
|
||||
if (play_count_ == 0) {
|
||||
|
|
@ -269,14 +269,14 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
|
|||
}
|
||||
}
|
||||
|
||||
// Detect received impulses in |source|, derive time between transmission and
|
||||
// Detect received impulses in `source`, derive time between transmission and
|
||||
// detection and add the calculated delay to list of latencies.
|
||||
void Write(const void* source, size_t num_frames) override {
|
||||
ASSERT_EQ(num_frames, frames_per_buffer_);
|
||||
rec_count_++;
|
||||
if (pulse_time_ == 0) {
|
||||
// Avoid detection of new impulse response until a new impulse has
|
||||
// been transmitted (sets |pulse_time_| to value larger than zero).
|
||||
// been transmitted (sets `pulse_time_` to value larger than zero).
|
||||
return;
|
||||
}
|
||||
const int16_t* ptr16 = static_cast<const int16_t*>(source);
|
||||
|
|
@ -295,7 +295,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
|
|||
// Total latency is the difference between transmit time and detection
|
||||
// tome plus the extra delay within the buffer in which we detected the
|
||||
// received impulse. It is transmitted at sample 0 but can be received
|
||||
// at sample N where N > 0. The term |extra_delay| accounts for N and it
|
||||
// at sample N where N > 0. The term `extra_delay` accounts for N and it
|
||||
// is a value between 0 and 10ms.
|
||||
latencies_.push_back(now_time - pulse_time_ + extra_delay);
|
||||
pulse_time_ = 0;
|
||||
|
|
|
|||
|
|
@ -514,7 +514,7 @@ class AndroidVideoDecoder implements VideoDecoder, VideoSink {
|
|||
throw new AssertionError("Stride is not divisible by two: " + stride);
|
||||
}
|
||||
|
||||
// Note that the case with odd |sliceHeight| is handled in a special way.
|
||||
// Note that the case with odd `sliceHeight` is handled in a special way.
|
||||
// The chroma height contained in the payload is rounded down instead of
|
||||
// up, making it one row less than what we expect in WebRTC. Therefore, we
|
||||
// have to duplicate the last chroma rows for this case. Also, the offset
|
||||
|
|
|
|||
|
|
@ -133,7 +133,7 @@ class Camera1Session implements CameraSession {
|
|||
|
||||
private static CaptureFormat findClosestCaptureFormat(
|
||||
android.hardware.Camera.Parameters parameters, int width, int height, int framerate) {
|
||||
// Find closest supported format for |width| x |height| @ |framerate|.
|
||||
// Find closest supported format for `width` x `height` @ `framerate`.
|
||||
final List<CaptureFormat.FramerateRange> supportedFramerates =
|
||||
Camera1Enumerator.convertFramerates(parameters.getSupportedPreviewFpsRange());
|
||||
Logging.d(TAG, "Available fps ranges: " + supportedFramerates);
|
||||
|
|
|
|||
|
|
@ -69,7 +69,7 @@ class EglBase14Impl implements EglBase14 {
|
|||
}
|
||||
|
||||
// Create a new context with the specified config type, sharing data with sharedContext.
|
||||
// |sharedContext| may be null.
|
||||
// `sharedContext` may be null.
|
||||
public EglBase14Impl(EGLContext sharedContext, int[] configAttributes) {
|
||||
eglDisplay = getEglDisplay();
|
||||
eglConfig = getEglConfig(eglDisplay, configAttributes);
|
||||
|
|
|
|||
|
|
@ -22,7 +22,7 @@ import org.webrtc.Logging;
|
|||
// This class wraps control of three different platform effects. Supported
|
||||
// effects are: AcousticEchoCanceler (AEC) and NoiseSuppressor (NS).
|
||||
// Calling enable() will active all effects that are
|
||||
// supported by the device if the corresponding |shouldEnableXXX| member is set.
|
||||
// supported by the device if the corresponding `shouldEnableXXX` member is set.
|
||||
class WebRtcAudioEffects {
|
||||
private static final boolean DEBUG = false;
|
||||
|
||||
|
|
@ -71,7 +71,7 @@ class WebRtcAudioEffects {
|
|||
}
|
||||
|
||||
// Call this method to enable or disable the platform AEC. It modifies
|
||||
// |shouldEnableAec| which is used in enable() where the actual state
|
||||
// `shouldEnableAec` which is used in enable() where the actual state
|
||||
// of the AEC effect is modified. Returns true if HW AEC is supported and
|
||||
// false otherwise.
|
||||
public boolean setAEC(boolean enable) {
|
||||
|
|
@ -90,7 +90,7 @@ class WebRtcAudioEffects {
|
|||
}
|
||||
|
||||
// Call this method to enable or disable the platform NS. It modifies
|
||||
// |shouldEnableNs| which is used in enable() where the actual state
|
||||
// `shouldEnableNs` which is used in enable() where the actual state
|
||||
// of the NS effect is modified. Returns true if HW NS is supported and
|
||||
// false otherwise.
|
||||
public boolean setNS(boolean enable) {
|
||||
|
|
@ -180,7 +180,7 @@ class WebRtcAudioEffects {
|
|||
}
|
||||
}
|
||||
|
||||
// Returns true for effect types in |type| that are of "VoIP" types:
|
||||
// Returns true for effect types in `type` that are of "VoIP" types:
|
||||
// Acoustic Echo Canceler (AEC) or Automatic Gain Control (AGC) or
|
||||
// Noise Suppressor (NS). Note that, an extra check for support is needed
|
||||
// in each comparison since some devices includes effects in the
|
||||
|
|
@ -217,7 +217,7 @@ class WebRtcAudioEffects {
|
|||
}
|
||||
|
||||
// Returns true if an effect of the specified type is available. Functionally
|
||||
// equivalent to (NoiseSuppressor|AutomaticGainControl|...).isAvailable(), but
|
||||
// equivalent to (NoiseSuppressor`AutomaticGainControl`...).isAvailable(), but
|
||||
// faster as it avoids the expensive OS call to enumerate effects.
|
||||
private static boolean isEffectTypeAvailable(UUID effectType, UUID blockListedUuid) {
|
||||
Descriptor[] effects = getAvailableEffects();
|
||||
|
|
|
|||
|
|
@ -237,7 +237,7 @@ class WebRtcAudioRecord {
|
|||
|
||||
// Returns true if verifyAudioConfig() succeeds. This value is set after a specific delay when
|
||||
// startRecording() has been called. Hence, should preferably be called in combination with
|
||||
// stopRecording() to ensure that it has been set properly. |isAudioConfigVerified| is
|
||||
// stopRecording() to ensure that it has been set properly. `isAudioConfigVerified` is
|
||||
// enabled in WebRtcAudioRecord to ensure that the returned value is valid.
|
||||
@CalledByNative
|
||||
boolean isAudioSourceMatchingRecordingSession() {
|
||||
|
|
@ -491,7 +491,7 @@ class WebRtcAudioRecord {
|
|||
long nativeAudioRecordJni, ByteBuffer byteBuffer);
|
||||
private native void nativeDataIsRecorded(long nativeAudioRecordJni, int bytes);
|
||||
|
||||
// Sets all recorded samples to zero if |mute| is true, i.e., ensures that
|
||||
// Sets all recorded samples to zero if `mute` is true, i.e., ensures that
|
||||
// the microphone is muted.
|
||||
public void setMicrophoneMute(boolean mute) {
|
||||
Logging.w(TAG, "setMicrophoneMute(" + mute + ")");
|
||||
|
|
|
|||
|
|
@ -76,7 +76,7 @@ class WebRtcAudioTrack {
|
|||
private @Nullable AudioTrackThread audioThread;
|
||||
private final VolumeLogger volumeLogger;
|
||||
|
||||
// Samples to be played are replaced by zeros if |speakerMute| is set to true.
|
||||
// Samples to be played are replaced by zeros if `speakerMute` is set to true.
|
||||
// Can be used to ensure that the speaker is fully muted.
|
||||
private volatile boolean speakerMute;
|
||||
private byte[] emptyBytes;
|
||||
|
|
@ -218,9 +218,9 @@ class WebRtcAudioTrack {
|
|||
Logging.d(TAG, "minBufferSizeInBytes: " + minBufferSizeInBytes);
|
||||
// For the streaming mode, data must be written to the audio sink in
|
||||
// chunks of size (given by byteBuffer.capacity()) less than or equal
|
||||
// to the total buffer size |minBufferSizeInBytes|. But, we have seen
|
||||
// to the total buffer size `minBufferSizeInBytes`. But, we have seen
|
||||
// reports of "getMinBufferSize(): error querying hardware". Hence, it
|
||||
// can happen that |minBufferSizeInBytes| contains an invalid value.
|
||||
// can happen that `minBufferSizeInBytes` contains an invalid value.
|
||||
if (minBufferSizeInBytes < byteBuffer.capacity()) {
|
||||
reportWebRtcAudioTrackInitError("AudioTrack.getMinBufferSize returns an invalid value.");
|
||||
return -1;
|
||||
|
|
@ -559,7 +559,7 @@ class WebRtcAudioTrack {
|
|||
long nativeAudioTrackJni, ByteBuffer byteBuffer);
|
||||
private static native void nativeGetPlayoutData(long nativeAudioTrackJni, int bytes);
|
||||
|
||||
// Sets all samples to be played out to zero if |mute| is true, i.e.,
|
||||
// Sets all samples to be played out to zero if `mute` is true, i.e.,
|
||||
// ensures that the speaker is muted.
|
||||
public void setSpeakerMute(boolean mute) {
|
||||
Logging.w(TAG, "setSpeakerMute(" + mute + ")");
|
||||
|
|
|
|||
|
|
@ -31,14 +31,14 @@ static ScopedJavaLocalRef<jobject> JNI_Metrics_GetAndReset(JNIEnv* jni) {
|
|||
std::map<std::string, std::unique_ptr<metrics::SampleInfo>> histograms;
|
||||
metrics::GetAndReset(&histograms);
|
||||
for (const auto& kv : histograms) {
|
||||
// Create and add samples to |HistogramInfo|.
|
||||
// Create and add samples to `HistogramInfo`.
|
||||
ScopedJavaLocalRef<jobject> j_info = Java_HistogramInfo_Constructor(
|
||||
jni, kv.second->min, kv.second->max,
|
||||
static_cast<int>(kv.second->bucket_count));
|
||||
for (const auto& sample : kv.second->samples) {
|
||||
Java_HistogramInfo_addSample(jni, j_info, sample.first, sample.second);
|
||||
}
|
||||
// Add |HistogramInfo| to |Metrics|.
|
||||
// Add `HistogramInfo` to `Metrics`.
|
||||
ScopedJavaLocalRef<jstring> j_name = NativeToJavaString(jni, kv.first);
|
||||
Java_Metrics_add(jni, j_metrics, j_name, j_info);
|
||||
}
|
||||
|
|
|
|||
|
|
@ -376,7 +376,7 @@ rtc::NetworkBindingResult AndroidNetworkMonitor::BindSocketToNetwork(
|
|||
rv = lollipopSetNetworkForSocket(*network_handle, socket_fd);
|
||||
}
|
||||
|
||||
// If |network| has since disconnected, |rv| will be ENONET. Surface this as
|
||||
// If `network` has since disconnected, `rv` will be ENONET. Surface this as
|
||||
// ERR_NETWORK_CHANGED, rather than MapSystemError(ENONET) which gives back
|
||||
// the less descriptive ERR_FAILED.
|
||||
if (rv == 0) {
|
||||
|
|
|
|||
|
|
@ -76,7 +76,7 @@ class AndroidNetworkMonitor : public rtc::NetworkMonitorInterface {
|
|||
void Start() override;
|
||||
void Stop() override;
|
||||
|
||||
// Does |this| NetworkMonitorInterface implement BindSocketToNetwork?
|
||||
// Does `this` NetworkMonitorInterface implement BindSocketToNetwork?
|
||||
// Only Android returns true.
|
||||
virtual bool SupportsBindSocketToNetwork() const override { return true; }
|
||||
|
||||
|
|
|
|||
|
|
@ -200,7 +200,7 @@ aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data,
|
|||
}
|
||||
|
||||
// Read audio data from the WebRTC source using the FineAudioBuffer object
|
||||
// and write that data into |audio_data| to be played out by AAudio.
|
||||
// and write that data into `audio_data` to be played out by AAudio.
|
||||
// Prime output with zeros during a short initial phase to avoid distortion.
|
||||
// TODO(henrika): do more work to figure out of if the initial forced silence
|
||||
// period is really needed.
|
||||
|
|
|
|||
|
|
@ -81,8 +81,8 @@ class AAudioPlayer final : public AudioOutput,
|
|||
protected:
|
||||
// AAudioObserverInterface implementation.
|
||||
|
||||
// For an output stream, this function should render and write |num_frames|
|
||||
// of data in the streams current data format to the |audio_data| buffer.
|
||||
// For an output stream, this function should render and write `num_frames`
|
||||
// of data in the streams current data format to the `audio_data` buffer.
|
||||
// Called on a real-time thread owned by AAudio.
|
||||
aaudio_data_callback_result_t OnDataCallback(void* audio_data,
|
||||
int32_t num_frames) override;
|
||||
|
|
|
|||
|
|
@ -157,7 +157,7 @@ void AAudioRecorder::OnErrorCallback(aaudio_result_t error) {
|
|||
}
|
||||
}
|
||||
|
||||
// Read and process |num_frames| of data from the |audio_data| buffer.
|
||||
// Read and process `num_frames` of data from the `audio_data` buffer.
|
||||
// TODO(henrika): possibly add trace here to be included in systrace.
|
||||
// See https://developer.android.com/studio/profile/systrace-commandline.html.
|
||||
aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
|
||||
|
|
@ -191,7 +191,7 @@ aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
|
|||
RTC_DLOG(INFO) << "input latency: " << latency_millis_
|
||||
<< ", num_frames: " << num_frames;
|
||||
}
|
||||
// Copy recorded audio in |audio_data| to the WebRTC sink using the
|
||||
// Copy recorded audio in `audio_data` to the WebRTC sink using the
|
||||
// FineAudioBuffer object.
|
||||
fine_audio_buffer_->DeliverRecordedData(
|
||||
rtc::MakeArrayView(static_cast<const int16_t*>(audio_data),
|
||||
|
|
|
|||
|
|
@ -72,8 +72,8 @@ class AAudioRecorder : public AudioInput,
|
|||
protected:
|
||||
// AAudioObserverInterface implementation.
|
||||
|
||||
// For an input stream, this function should read |num_frames| of recorded
|
||||
// data, in the stream's current data format, from the |audio_data| buffer.
|
||||
// For an input stream, this function should read `num_frames` of recorded
|
||||
// data, in the stream's current data format, from the `audio_data` buffer.
|
||||
// Called on a real-time thread owned by AAudio.
|
||||
aaudio_data_callback_result_t OnDataCallback(void* audio_data,
|
||||
int32_t num_frames) override;
|
||||
|
|
|
|||
|
|
@ -253,8 +253,8 @@ void AudioRecordJni::DataIsRecorded(JNIEnv* env,
|
|||
audio_device_buffer_->SetRecordedBuffer(direct_buffer_address_,
|
||||
frames_per_buffer_);
|
||||
// We provide one (combined) fixed delay estimate for the APM and use the
|
||||
// |playDelayMs| parameter only. Components like the AEC only sees the sum
|
||||
// of |playDelayMs| and |recDelayMs|, hence the distributions does not matter.
|
||||
// `playDelayMs` parameter only. Components like the AEC only sees the sum
|
||||
// of `playDelayMs` and `recDelayMs`, hence the distributions does not matter.
|
||||
audio_device_buffer_->SetVQEData(total_delay_ms_, 0);
|
||||
if (audio_device_buffer_->DeliverRecordedData() == -1) {
|
||||
RTC_LOG(INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";
|
||||
|
|
|
|||
|
|
@ -74,8 +74,8 @@ class AudioRecordJni : public AudioInput {
|
|||
int32_t EnableBuiltInNS(bool enable) override;
|
||||
|
||||
// Called from Java side so we can cache the address of the Java-manged
|
||||
// |byte_buffer| in |direct_buffer_address_|. The size of the buffer
|
||||
// is also stored in |direct_buffer_capacity_in_bytes_|.
|
||||
// `byte_buffer` in `direct_buffer_address_`. The size of the buffer
|
||||
// is also stored in `direct_buffer_capacity_in_bytes_`.
|
||||
// This method will be called by the WebRtcAudioRecord constructor, i.e.,
|
||||
// on the same thread that this object is created on.
|
||||
void CacheDirectBufferAddress(JNIEnv* env,
|
||||
|
|
@ -83,8 +83,8 @@ class AudioRecordJni : public AudioInput {
|
|||
const JavaParamRef<jobject>& byte_buffer);
|
||||
|
||||
// Called periodically by the Java based WebRtcAudioRecord object when
|
||||
// recording has started. Each call indicates that there are |length| new
|
||||
// bytes recorded in the memory area |direct_buffer_address_| and it is
|
||||
// recording has started. Each call indicates that there are `length` new
|
||||
// bytes recorded in the memory area `direct_buffer_address_` and it is
|
||||
// now time to send these to the consumer.
|
||||
// This method is called on a high-priority thread from Java. The name of
|
||||
// the thread is 'AudioRecordThread'.
|
||||
|
|
@ -111,10 +111,10 @@ class AudioRecordJni : public AudioInput {
|
|||
// possible values. See audio_common.h for details.
|
||||
const int total_delay_ms_;
|
||||
|
||||
// Cached copy of address to direct audio buffer owned by |j_audio_record_|.
|
||||
// Cached copy of address to direct audio buffer owned by `j_audio_record_`.
|
||||
void* direct_buffer_address_;
|
||||
|
||||
// Number of bytes in the direct audio buffer owned by |j_audio_record_|.
|
||||
// Number of bytes in the direct audio buffer owned by `j_audio_record_`.
|
||||
size_t direct_buffer_capacity_in_bytes_;
|
||||
|
||||
// Number audio frames per audio buffer. Each audio frame corresponds to
|
||||
|
|
|
|||
|
|
@ -71,14 +71,14 @@ class AudioTrackJni : public AudioOutput {
|
|||
void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
|
||||
|
||||
// Called from Java side so we can cache the address of the Java-manged
|
||||
// |byte_buffer| in |direct_buffer_address_|. The size of the buffer
|
||||
// is also stored in |direct_buffer_capacity_in_bytes_|.
|
||||
// `byte_buffer` in `direct_buffer_address_`. The size of the buffer
|
||||
// is also stored in `direct_buffer_capacity_in_bytes_`.
|
||||
// Called on the same thread as the creating thread.
|
||||
void CacheDirectBufferAddress(JNIEnv* env,
|
||||
const JavaParamRef<jobject>& byte_buffer);
|
||||
// Called periodically by the Java based WebRtcAudioTrack object when
|
||||
// playout has started. Each call indicates that |length| new bytes should
|
||||
// be written to the memory area |direct_buffer_address_| for playout.
|
||||
// playout has started. Each call indicates that `length` new bytes should
|
||||
// be written to the memory area `direct_buffer_address_` for playout.
|
||||
// This method is called on a high-priority thread from Java. The name of
|
||||
// the thread is 'AudioTrackThread'.
|
||||
void GetPlayoutData(JNIEnv* env, size_t length);
|
||||
|
|
@ -99,10 +99,10 @@ class AudioTrackJni : public AudioOutput {
|
|||
// AudioManager.
|
||||
const AudioParameters audio_parameters_;
|
||||
|
||||
// Cached copy of address to direct audio buffer owned by |j_audio_track_|.
|
||||
// Cached copy of address to direct audio buffer owned by `j_audio_track_`.
|
||||
void* direct_buffer_address_;
|
||||
|
||||
// Number of bytes in the direct audio buffer owned by |j_audio_track_|.
|
||||
// Number of bytes in the direct audio buffer owned by `j_audio_track_`.
|
||||
size_t direct_buffer_capacity_in_bytes_;
|
||||
|
||||
// Number of audio frames per audio buffer. Each audio frame corresponds to
|
||||
|
|
|
|||
|
|
@ -95,7 +95,7 @@ class OpenSLESPlayer : public AudioOutput {
|
|||
// Reads audio data in PCM format using the AudioDeviceBuffer.
|
||||
// Can be called both on the main thread (during Start()) and from the
|
||||
// internal audio thread while output streaming is active.
|
||||
// If the |silence| flag is set, the audio is filled with zeros instead of
|
||||
// If the `silence` flag is set, the audio is filled with zeros instead of
|
||||
// asking the WebRTC layer for real audio data. This procedure is also known
|
||||
// as audio priming.
|
||||
void EnqueuePlayoutData(bool silence);
|
||||
|
|
@ -106,7 +106,7 @@ class OpenSLESPlayer : public AudioOutput {
|
|||
|
||||
// Obtaines the SL Engine Interface from the existing global Engine object.
|
||||
// The interface exposes creation methods of all the OpenSL ES object types.
|
||||
// This method defines the |engine_| member variable.
|
||||
// This method defines the `engine_` member variable.
|
||||
bool ObtainEngineInterface();
|
||||
|
||||
// Creates/destroys the output mix object.
|
||||
|
|
|
|||
|
|
@ -88,7 +88,7 @@ class OpenSLESRecorder : public AudioInput {
|
|||
private:
|
||||
// Obtaines the SL Engine Interface from the existing global Engine object.
|
||||
// The interface exposes creation methods of all the OpenSL ES object types.
|
||||
// This method defines the |engine_| member variable.
|
||||
// This method defines the `engine_` member variable.
|
||||
bool ObtainEngineInterface();
|
||||
|
||||
// Creates/destroys the audio recorder and the simple-buffer queue object.
|
||||
|
|
@ -109,7 +109,7 @@ class OpenSLESRecorder : public AudioInput {
|
|||
// Wraps calls to SLAndroidSimpleBufferQueueState::Enqueue() and it can be
|
||||
// called both on the main thread (but before recording has started) and from
|
||||
// the internal audio thread while input streaming is active. It uses
|
||||
// |simple_buffer_queue_| but no lock is needed since the initial calls from
|
||||
// `simple_buffer_queue_` but no lock is needed since the initial calls from
|
||||
// the main thread and the native callback thread are mutually exclusive.
|
||||
bool EnqueueAudioBuffer();
|
||||
|
||||
|
|
|
|||
|
|
@ -15,7 +15,7 @@
|
|||
|
||||
namespace webrtc {
|
||||
|
||||
// If |atomic_class_id| set, it'll return immediately. Otherwise, it will look
|
||||
// If `atomic_class_id` set, it'll return immediately. Otherwise, it will look
|
||||
// up the class and store it. If there's a race, we take care to only store one
|
||||
// global reference (and the duplicated effort will happen only once).
|
||||
jclass LazyGetClass(JNIEnv* env,
|
||||
|
|
@ -29,18 +29,18 @@ jclass LazyGetClass(JNIEnv* env,
|
|||
jclass cas_result = nullptr;
|
||||
if (std::atomic_compare_exchange_strong(atomic_class_id, &cas_result,
|
||||
clazz.obj())) {
|
||||
// We sucessfully stored |clazz| in |atomic_class_id|, so we are
|
||||
// We sucessfully stored `clazz` in `atomic_class_id`, so we are
|
||||
// intentionally leaking the global ref since it's now stored there.
|
||||
return clazz.Release();
|
||||
} else {
|
||||
// Some other thread came before us and stored a global pointer in
|
||||
// |atomic_class_id|. Relase our global ref and return the ref from the
|
||||
// `atomic_class_id`. Relase our global ref and return the ref from the
|
||||
// other thread.
|
||||
return cas_result;
|
||||
}
|
||||
}
|
||||
|
||||
// If |atomic_method_id| set, it'll return immediately. Otherwise, it will look
|
||||
// If `atomic_method_id` set, it'll return immediately. Otherwise, it will look
|
||||
// up the method id and store it. If there's a race, it's ok since the values
|
||||
// are the same (and the duplicated effort will happen only once).
|
||||
template <MethodID::Type type>
|
||||
|
|
|
|||
|
|
@ -44,11 +44,11 @@
|
|||
|
||||
namespace webrtc {
|
||||
|
||||
// This function will initialize |atomic_class_id| to contain a global ref to
|
||||
// This function will initialize `atomic_class_id` to contain a global ref to
|
||||
// the given class, and will return that ref on subsequent calls. The caller is
|
||||
// responsible to zero-initialize |atomic_class_id|. It's fine to
|
||||
// responsible to zero-initialize `atomic_class_id`. It's fine to
|
||||
// simultaneously call this on multiple threads referencing the same
|
||||
// |atomic_method_id|.
|
||||
// `atomic_method_id`.
|
||||
jclass LazyGetClass(JNIEnv* env,
|
||||
const char* class_name,
|
||||
std::atomic<jclass>* atomic_class_id);
|
||||
|
|
@ -61,11 +61,11 @@ class MethodID {
|
|||
TYPE_INSTANCE,
|
||||
};
|
||||
|
||||
// This function will initialize |atomic_method_id| to contain a ref to
|
||||
// This function will initialize `atomic_method_id` to contain a ref to
|
||||
// the given method, and will return that ref on subsequent calls. The caller
|
||||
// is responsible to zero-initialize |atomic_method_id|. It's fine to
|
||||
// is responsible to zero-initialize `atomic_method_id`. It's fine to
|
||||
// simultaneously call this on multiple threads referencing the same
|
||||
// |atomic_method_id|.
|
||||
// `atomic_method_id`.
|
||||
template <Type type>
|
||||
static jmethodID LazyGet(JNIEnv* env,
|
||||
jclass clazz,
|
||||
|
|
@ -151,7 +151,7 @@ struct BASE_EXPORT JniJavaCallContextChecked {
|
|||
const char* jni_signature,
|
||||
std::atomic<jmethodID>* atomic_method_id) {
|
||||
base.Init<type>(env, clazz, method_name, jni_signature, atomic_method_id);
|
||||
// Reset |pc| to correct caller.
|
||||
// Reset `pc` to correct caller.
|
||||
base.pc = reinterpret_cast<uintptr_t>(__builtin_return_address(0));
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -27,7 +27,7 @@ static JavaVM* g_jvm = nullptr;
|
|||
|
||||
static pthread_once_t g_jni_ptr_once = PTHREAD_ONCE_INIT;
|
||||
|
||||
// Key for per-thread JNIEnv* data. Non-NULL in threads attached to |g_jvm| by
|
||||
// Key for per-thread JNIEnv* data. Non-NULL in threads attached to `g_jvm` by
|
||||
// AttachCurrentThreadIfNeeded(), NULL in unattached threads and threads that
|
||||
// were attached by the JVM because of a Java->native call.
|
||||
static pthread_key_t g_jni_ptr;
|
||||
|
|
@ -48,7 +48,7 @@ JNIEnv* GetEnv() {
|
|||
}
|
||||
|
||||
static void ThreadDestructor(void* prev_jni_ptr) {
|
||||
// This function only runs on threads where |g_jni_ptr| is non-NULL, meaning
|
||||
// This function only runs on threads where `g_jni_ptr` is non-NULL, meaning
|
||||
// we were responsible for originally attaching the thread, so are responsible
|
||||
// for detaching it now. However, because some JVM implementations (notably
|
||||
// Oracle's http://goo.gl/eHApYT) also use the pthread_key_create mechanism,
|
||||
|
|
@ -102,7 +102,7 @@ static std::string GetThreadName() {
|
|||
return std::string(name);
|
||||
}
|
||||
|
||||
// Return a |JNIEnv*| usable on this thread. Attaches to |g_jvm| if necessary.
|
||||
// Return a |JNIEnv*| usable on this thread. Attaches to `g_jvm` if necessary.
|
||||
JNIEnv* AttachCurrentThreadIfNeeded() {
|
||||
JNIEnv* jni = GetEnv();
|
||||
if (jni)
|
||||
|
|
|
|||
|
|
@ -23,7 +23,7 @@ JNIEnv* GetEnv();
|
|||
|
||||
JavaVM* GetJVM();
|
||||
|
||||
// Return a |JNIEnv*| usable on this thread. Attaches to |g_jvm| if necessary.
|
||||
// Return a |JNIEnv*| usable on this thread. Attaches to `g_jvm` if necessary.
|
||||
JNIEnv* AttachCurrentThreadIfNeeded();
|
||||
|
||||
} // namespace jni
|
||||
|
|
|
|||
|
|
@ -49,7 +49,7 @@ JavaMediaStream::JavaMediaStream(
|
|||
observer_->SignalVideoTrackAdded.connect(
|
||||
this, &JavaMediaStream::OnVideoTrackAddedToStream);
|
||||
|
||||
// |j_media_stream| holds one reference. Corresponding Release() is in
|
||||
// `j_media_stream` holds one reference. Corresponding Release() is in
|
||||
// MediaStream_free, triggered by MediaStream.dispose().
|
||||
media_stream.release();
|
||||
}
|
||||
|
|
|
|||
|
|
@ -499,7 +499,7 @@ static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetLocalDescription(
|
|||
const JavaParamRef<jobject>& j_pc) {
|
||||
PeerConnectionInterface* pc = ExtractNativePC(jni, j_pc);
|
||||
// It's only safe to operate on SessionDescriptionInterface on the
|
||||
// signaling thread, but |jni| may only be used on the current thread, so we
|
||||
// signaling thread, but `jni` may only be used on the current thread, so we
|
||||
// must do this odd dance.
|
||||
std::string sdp;
|
||||
std::string type;
|
||||
|
|
@ -518,7 +518,7 @@ static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetRemoteDescription(
|
|||
const JavaParamRef<jobject>& j_pc) {
|
||||
PeerConnectionInterface* pc = ExtractNativePC(jni, j_pc);
|
||||
// It's only safe to operate on SessionDescriptionInterface on the
|
||||
// signaling thread, but |jni| may only be used on the current thread, so we
|
||||
// signaling thread, but `jni` may only be used on the current thread, so we
|
||||
// must do this odd dance.
|
||||
std::string sdp;
|
||||
std::string type;
|
||||
|
|
|
|||
|
|
@ -242,9 +242,9 @@ static void JNI_PeerConnectionFactory_ShutdownInternalTracer(JNIEnv* jni) {
|
|||
}
|
||||
|
||||
// Following parameters are optional:
|
||||
// |audio_device_module|, |jencoder_factory|, |jdecoder_factory|,
|
||||
// |audio_processor|, |fec_controller_factory|,
|
||||
// |network_state_predictor_factory|, |neteq_factory|.
|
||||
// `audio_device_module`, `jencoder_factory`, `jdecoder_factory`,
|
||||
// `audio_processor`, `fec_controller_factory`,
|
||||
// `network_state_predictor_factory`, `neteq_factory`.
|
||||
ScopedJavaLocalRef<jobject> CreatePeerConnectionFactoryForJava(
|
||||
JNIEnv* jni,
|
||||
const JavaParamRef<jobject>& jcontext,
|
||||
|
|
|
|||
|
|
@ -18,7 +18,7 @@
|
|||
namespace webrtc {
|
||||
namespace jni {
|
||||
|
||||
// Creates java PeerConnectionFactory with specified |pcf|.
|
||||
// Creates java PeerConnectionFactory with specified `pcf`.
|
||||
jobject NativeToJavaPeerConnectionFactory(
|
||||
JNIEnv* jni,
|
||||
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf,
|
||||
|
|
|
|||
|
|
@ -23,7 +23,7 @@ ScopedJavaLocalRef<jobject> NativeToJavaRtpReceiver(
|
|||
JNIEnv* env,
|
||||
rtc::scoped_refptr<RtpReceiverInterface> receiver);
|
||||
|
||||
// Takes ownership of the passed |j_receiver| and stores it as a global
|
||||
// Takes ownership of the passed `j_receiver` and stores it as a global
|
||||
// reference. Will call dispose() in the dtor.
|
||||
class JavaRtpReceiverGlobalOwner {
|
||||
public:
|
||||
|
|
|
|||
|
|
@ -27,7 +27,7 @@ ScopedJavaLocalRef<jobject> NativeToJavaRtpTransceiver(
|
|||
JNIEnv* env,
|
||||
rtc::scoped_refptr<RtpTransceiverInterface> transceiver);
|
||||
|
||||
// This takes ownership of the of the |j_transceiver| and stores it as a global
|
||||
// This takes ownership of the of the `j_transceiver` and stores it as a global
|
||||
// reference. This calls the Java Transceiver's dispose() method with the dtor.
|
||||
class JavaRtpTransceiverGlobalOwner {
|
||||
public:
|
||||
|
|
|
|||
|
|
@ -267,7 +267,7 @@ void VideoEncoderWrapper::OnEncodedFrame(
|
|||
frame_extra_infos_.pop_front();
|
||||
}
|
||||
|
||||
// This is a bit subtle. The |frame| variable from the lambda capture is
|
||||
// This is a bit subtle. The `frame` variable from the lambda capture is
|
||||
// const. Which implies that (i) we need to make a copy to be able to
|
||||
// write to the metadata, and (ii) we should avoid using the .data()
|
||||
// method (including implicit conversion to ArrayView) on the non-const
|
||||
|
|
|
|||
|
|
@ -41,8 +41,8 @@ class AndroidVideoBuffer : public VideoFrameBuffer {
|
|||
|
||||
const ScopedJavaGlobalRef<jobject>& video_frame_buffer() const;
|
||||
|
||||
// Crops a region defined by |crop_x|, |crop_y|, |crop_width| and
|
||||
// |crop_height|. Scales it to size |scale_width| x |scale_height|.
|
||||
// Crops a region defined by `crop_x`, `crop_y`, `crop_width` and
|
||||
// `crop_height`. Scales it to size `scale_width` x `scale_height`.
|
||||
rtc::scoped_refptr<VideoFrameBuffer> CropAndScale(int crop_x,
|
||||
int crop_y,
|
||||
int crop_width,
|
||||
|
|
|
|||
|
|
@ -17,8 +17,8 @@ namespace webrtc {
|
|||
namespace {
|
||||
|
||||
// Find the highest-priority instance of the T-valued constraint named by
|
||||
// |key| and return its value as |value|. |constraints| can be null.
|
||||
// If |mandatory_constraints| is non-null, it is incremented if the key appears
|
||||
// `key` and return its value as `value`. `constraints` can be null.
|
||||
// If `mandatory_constraints` is non-null, it is incremented if the key appears
|
||||
// among the mandatory constraints.
|
||||
// Returns true if the key was found and has a valid value for type T.
|
||||
// If the key appears multiple times as an optional constraint, appearances
|
||||
|
|
@ -135,8 +135,8 @@ const char MediaConstraints::kRawPacketizationForVideoEnabled[] =
|
|||
|
||||
const char MediaConstraints::kNumSimulcastLayers[] = "googNumSimulcastLayers";
|
||||
|
||||
// Set |value| to the value associated with the first appearance of |key|, or
|
||||
// return false if |key| is not found.
|
||||
// Set `value` to the value associated with the first appearance of `key`, or
|
||||
// return false if `key` is not found.
|
||||
bool MediaConstraints::Constraints::FindFirst(const std::string& key,
|
||||
std::string* value) const {
|
||||
for (Constraints::const_iterator iter = begin(); iter != end(); ++iter) {
|
||||
|
|
@ -209,7 +209,7 @@ void CopyConstraintsIntoAudioOptions(const MediaConstraints* constraints,
|
|||
ConstraintToOptional<std::string>(
|
||||
constraints, MediaConstraints::kAudioNetworkAdaptorConfig,
|
||||
&options->audio_network_adaptor_config);
|
||||
// When |kAudioNetworkAdaptorConfig| is defined, it both means that audio
|
||||
// When `kAudioNetworkAdaptorConfig` is defined, it both means that audio
|
||||
// network adaptor is desired, and provides the config string.
|
||||
if (options->audio_network_adaptor_config) {
|
||||
options->audio_network_adaptor = true;
|
||||
|
|
|
|||
|
|
@ -20,7 +20,7 @@ RTC_OBJC_EXPORT
|
|||
|
||||
- (instancetype)init NS_UNAVAILABLE;
|
||||
|
||||
// Sets the volume for the RTCMediaSource. |volume| is a gain value in the range
|
||||
// Sets the volume for the RTCMediaSource. `volume` is a gain value in the range
|
||||
// [0, 10].
|
||||
// Temporary fix to be able to modify volume of remote audio tracks.
|
||||
// TODO(kthelgason): Property stays here temporarily until a proper volume-api
|
||||
|
|
|
|||
|
|
@ -84,7 +84,7 @@ RTC_OBJC_EXPORT
|
|||
@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCertificate) * certificate;
|
||||
|
||||
/** Which candidates the ICE agent is allowed to use. The W3C calls it
|
||||
* |iceTransportPolicy|, while in C++ it is called |type|. */
|
||||
* `iceTransportPolicy`, while in C++ it is called `type`. */
|
||||
@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
|
||||
|
||||
/** The media-bundling policy to use when gathering ICE candidates. */
|
||||
|
|
@ -144,7 +144,7 @@ RTC_OBJC_EXPORT
|
|||
*/
|
||||
@property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed;
|
||||
|
||||
/* This flag is only effective when |continualGatheringPolicy| is
|
||||
/* This flag is only effective when `continualGatheringPolicy` is
|
||||
* RTCContinualGatheringPolicyGatherContinually.
|
||||
*
|
||||
* If YES, after the ICE transport type is changed such that new types of
|
||||
|
|
|
|||
|
|
@ -21,13 +21,13 @@ RTC_OBJC_EXPORT
|
|||
/** NSData representation of the underlying buffer. */
|
||||
@property(nonatomic, readonly) NSData *data;
|
||||
|
||||
/** Indicates whether |data| contains UTF-8 or binary data. */
|
||||
/** Indicates whether `data` contains UTF-8 or binary data. */
|
||||
@property(nonatomic, readonly) BOOL isBinary;
|
||||
|
||||
- (instancetype)init NS_UNAVAILABLE;
|
||||
|
||||
/**
|
||||
* Initialize an RTCDataBuffer from NSData. |isBinary| indicates whether |data|
|
||||
* Initialize an RTCDataBuffer from NSData. `isBinary` indicates whether `data`
|
||||
* contains UTF-8 or binary data.
|
||||
*/
|
||||
- (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary;
|
||||
|
|
@ -47,7 +47,7 @@ RTC_OBJC_EXPORT
|
|||
didReceiveMessageWithBuffer:(RTC_OBJC_TYPE(RTCDataBuffer) *)buffer;
|
||||
|
||||
@optional
|
||||
/** The data channel's |bufferedAmount| changed. */
|
||||
/** The data channel's `bufferedAmount` changed. */
|
||||
- (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
|
||||
didChangeBufferedAmount:(uint64_t)amount;
|
||||
|
||||
|
|
@ -124,7 +124,7 @@ RTC_OBJC_EXPORT
|
|||
/** Closes the data channel. */
|
||||
- (void)close;
|
||||
|
||||
/** Attempt to send |data| on this data channel's underlying data transport. */
|
||||
/** Attempt to send `data` on this data channel's underlying data transport. */
|
||||
- (BOOL)sendData:(RTC_OBJC_TYPE(RTCDataBuffer) *)data;
|
||||
|
||||
@end
|
||||
|
|
|
|||
|
|
@ -43,7 +43,7 @@ RTC_OBJC_EXPORT
|
|||
// kRTCFileLoggerTypeCall.
|
||||
@property(nonatomic, readonly) RTCFileLoggerRotationType rotationType;
|
||||
|
||||
// Disables buffering disk writes. Should be set before |start|. Buffering
|
||||
// Disables buffering disk writes. Should be set before `start`. Buffering
|
||||
// is enabled by default for performance.
|
||||
@property(nonatomic, assign) BOOL shouldDisableBuffering;
|
||||
|
||||
|
|
|
|||
|
|
@ -37,9 +37,9 @@ RTC_OBJC_EXPORT
|
|||
@property(nonatomic, readonly) RTCTlsCertPolicy tlsCertPolicy;
|
||||
|
||||
/**
|
||||
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 necessary.
|
||||
extension). If `urls` itself contains the hostname, this isn't necessary.
|
||||
*/
|
||||
@property(nonatomic, readonly, nullable) NSString *hostname;
|
||||
|
||||
|
|
|
|||
|
|
@ -174,7 +174,7 @@ RTC_OBJC_EXPORT
|
|||
*/
|
||||
@property(nonatomic, weak, nullable) id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)> delegate;
|
||||
/** This property is not available with RTCSdpSemanticsUnifiedPlan. Please use
|
||||
* |senders| instead.
|
||||
* `senders` instead.
|
||||
*/
|
||||
@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *localStreams;
|
||||
@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * localDescription;
|
||||
|
|
@ -207,7 +207,7 @@ RTC_OBJC_EXPORT
|
|||
|
||||
- (instancetype)init NS_UNAVAILABLE;
|
||||
|
||||
/** Sets the PeerConnection's global configuration to |configuration|.
|
||||
/** Sets the PeerConnection's global configuration to `configuration`.
|
||||
* 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. Note that the BUNDLE and RTCP-multiplexing policies
|
||||
|
|
@ -243,7 +243,7 @@ RTC_OBJC_EXPORT
|
|||
|
||||
/** Add a new media stream track to be sent on this peer connection, and return
|
||||
* the newly created RTCRtpSender. The RTCRtpSender will be
|
||||
* associated with the streams specified in the |streamIds| list.
|
||||
* associated with the streams specified in the `streamIds` list.
|
||||
*
|
||||
* Errors: If an error occurs, returns nil. An error can occur if:
|
||||
* - A sender already exists for the track.
|
||||
|
|
@ -265,7 +265,7 @@ RTC_OBJC_EXPORT
|
|||
* 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 nil. Setting a
|
||||
* The initial value of `mid` in the returned transceiver is nil. Setting a
|
||||
* new session description may change it to a non-nil value.
|
||||
*
|
||||
* https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
|
||||
|
|
@ -325,7 +325,7 @@ RTC_OBJC_EXPORT
|
|||
|
||||
/** Limits the bandwidth allocated for all RTP streams sent by this
|
||||
* PeerConnection. Nil parameters will be unchanged. Setting
|
||||
* |currentBitrateBps| will force the available bitrate estimate to the given
|
||||
* `currentBitrateBps` will force the available bitrate estimate to the given
|
||||
* value. Returns YES if the parameters were successfully updated.
|
||||
*/
|
||||
- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
|
||||
|
|
@ -365,7 +365,7 @@ typedef void (^RTCStatisticsCompletionHandler)(RTC_OBJC_TYPE(RTCStatisticsReport
|
|||
@interface RTC_OBJC_TYPE (RTCPeerConnection)
|
||||
(Stats)
|
||||
|
||||
/** Gather stats for the given RTCMediaStreamTrack. If |mediaStreamTrack| is nil
|
||||
/** Gather stats for the given RTCMediaStreamTrack. If `mediaStreamTrack` is nil
|
||||
* statistics are gathered for all tracks.
|
||||
*/
|
||||
- (void)statsForTrack
|
||||
|
|
|
|||
|
|
@ -22,8 +22,8 @@ NS_ASSUME_NONNULL_BEGIN
|
|||
*/
|
||||
@property(nonatomic, readonly) int activationCount;
|
||||
|
||||
/** The number of times |beginWebRTCSession| was called without a balanced call
|
||||
* to |endWebRTCSession|.
|
||||
/** The number of times `beginWebRTCSession` was called without a balanced call
|
||||
* to `endWebRTCSession`.
|
||||
*/
|
||||
@property(nonatomic, readonly) int webRTCSessionCount;
|
||||
|
||||
|
|
@ -57,16 +57,16 @@ NS_ASSUME_NONNULL_BEGIN
|
|||
/** Configure the audio session for WebRTC. This call will fail if the session
|
||||
* is already configured. On other failures, we will attempt to restore the
|
||||
* previously used audio session configuration.
|
||||
* |lockForConfiguration| must be called first.
|
||||
* `lockForConfiguration` must be called first.
|
||||
* Successful calls to configureWebRTCSession must be matched by calls to
|
||||
* |unconfigureWebRTCSession|.
|
||||
* `unconfigureWebRTCSession`.
|
||||
*/
|
||||
- (BOOL)configureWebRTCSession:(NSError **)outError;
|
||||
|
||||
/** Unconfigures the session for WebRTC. This will attempt to restore the
|
||||
* audio session to the settings used before |configureWebRTCSession| was
|
||||
* audio session to the settings used before `configureWebRTCSession` was
|
||||
* called.
|
||||
* |lockForConfiguration| must be called first.
|
||||
* `lockForConfiguration` must be called first.
|
||||
*/
|
||||
- (BOOL)unconfigureWebRTCSession:(NSError **)outError;
|
||||
|
||||
|
|
|
|||
|
|
@ -209,9 +209,9 @@ RTC_OBJC_EXPORT
|
|||
/** Relinquishes exclusive access to the audio session. */
|
||||
- (void)unlockForConfiguration;
|
||||
|
||||
/** If |active|, activates the audio session if it isn't already active.
|
||||
/** If `active`, activates the audio session if it isn't already active.
|
||||
* Successful calls must be balanced with a setActive:NO when activation is no
|
||||
* longer required. If not |active|, deactivates the audio session if one is
|
||||
* longer required. If not `active`, deactivates the audio session if one is
|
||||
* active and this is the last balanced call. When deactivating, the
|
||||
* AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
|
||||
* AVAudioSession.
|
||||
|
|
@ -219,7 +219,7 @@ RTC_OBJC_EXPORT
|
|||
- (BOOL)setActive:(BOOL)active error:(NSError **)outError;
|
||||
|
||||
// The following methods are proxies for the associated methods on
|
||||
// AVAudioSession. |lockForConfiguration| must be called before using them
|
||||
// AVAudioSession. `lockForConfiguration` must be called before using them
|
||||
// otherwise they will fail with kRTCAudioSessionErrorLockRequired.
|
||||
|
||||
- (BOOL)setCategory:(NSString *)category
|
||||
|
|
@ -245,13 +245,13 @@ RTC_OBJC_EXPORT
|
|||
/** Applies the configuration to the current session. Attempts to set all
|
||||
* properties even if previous ones fail. Only the last error will be
|
||||
* returned.
|
||||
* |lockForConfiguration| must be called first.
|
||||
* `lockForConfiguration` must be called first.
|
||||
*/
|
||||
- (BOOL)setConfiguration : (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration error
|
||||
: (NSError **)outError;
|
||||
|
||||
/** Convenience method that calls both setConfiguration and setActive.
|
||||
* |lockForConfiguration| must be called first.
|
||||
* `lockForConfiguration` must be called first.
|
||||
*/
|
||||
- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration
|
||||
active:(BOOL)active
|
||||
|
|
|
|||
|
|
@ -23,7 +23,7 @@ class AudioSessionObserver;
|
|||
|
||||
- (instancetype)init NS_UNAVAILABLE;
|
||||
|
||||
/** |observer| is a raw pointer and should be kept alive
|
||||
/** `observer` is a raw pointer and should be kept alive
|
||||
* for this object's lifetime.
|
||||
*/
|
||||
- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer NS_DESIGNATED_INITIALIZER;
|
||||
|
|
|
|||
|
|
@ -165,7 +165,7 @@ typedef NS_ENUM(NSInteger, RTCFileVideoCapturerStatus) {
|
|||
int64_t presentationDifferenceRound = lroundf(presentationDifference * NSEC_PER_SEC);
|
||||
|
||||
__block dispatch_source_t timer = [self createStrictTimer];
|
||||
// Strict timer that will fire |presentationDifferenceRound| ns from now and never again.
|
||||
// Strict timer that will fire `presentationDifferenceRound` ns from now and never again.
|
||||
dispatch_source_set_timer(timer,
|
||||
dispatch_time(DISPATCH_TIME_NOW, presentationDifferenceRound),
|
||||
DISPATCH_TIME_FOREVER,
|
||||
|
|
|
|||
|
|
@ -14,7 +14,7 @@
|
|||
|
||||
@interface RTCNetworkMonitor ()
|
||||
|
||||
/** |observer| is a raw pointer and should be kept alive
|
||||
/** `observer` is a raw pointer and should be kept alive
|
||||
* for this object's lifetime.
|
||||
*/
|
||||
- (instancetype)initWithObserver:(webrtc::NetworkMonitorObserver *)observer
|
||||
|
|
|
|||
|
|
@ -31,11 +31,11 @@
|
|||
// the method that will trigger the binding of the render
|
||||
// buffer. Because the standard behaviour of -[UIView setNeedsDisplay]
|
||||
// is disabled for the reasons above, the RTC_OBJC_TYPE(RTCEAGLVideoView) maintains
|
||||
// its own |isDirty| flag.
|
||||
// its own `isDirty` flag.
|
||||
|
||||
@interface RTC_OBJC_TYPE (RTCEAGLVideoView)
|
||||
()<GLKViewDelegate>
|
||||
// |videoFrame| is set when we receive a frame from a worker thread and is read
|
||||
// `videoFrame` is set when we receive a frame from a worker thread and is read
|
||||
// from the display link callback so atomicity is required.
|
||||
@property(atomic, strong) RTC_OBJC_TYPE(RTCVideoFrame) * videoFrame;
|
||||
@property(nonatomic, readonly) GLKView *glkView;
|
||||
|
|
@ -183,7 +183,7 @@
|
|||
// redrawn. This occurs on main thread.
|
||||
- (void)glkView:(GLKView *)view drawInRect:(CGRect)rect {
|
||||
// The renderer will draw the frame to the framebuffer corresponding to the
|
||||
// one used by |view|.
|
||||
// one used by `view`.
|
||||
RTC_OBJC_TYPE(RTCVideoFrame) *frame = self.videoFrame;
|
||||
if (!frame || frame.timeStampNs == _lastDrawnFrameTimeStampNs) {
|
||||
return;
|
||||
|
|
|
|||
|
|
@ -25,7 +25,7 @@
|
|||
|
||||
@interface RTC_OBJC_TYPE (RTCNSGLVideoView)
|
||||
()
|
||||
// |videoFrame| is set when we receive a frame from a worker thread and is read
|
||||
// `videoFrame` is set when we receive a frame from a worker thread and is read
|
||||
// from the display link callback so atomicity is required.
|
||||
@property(atomic, strong) RTC_OBJC_TYPE(RTCVideoFrame) *
|
||||
videoFrame;
|
||||
|
|
|
|||
|
|
@ -36,7 +36,7 @@ const char kRTCVertexShaderSource[] =
|
|||
" v_texcoord = texcoord;\n"
|
||||
"}\n";
|
||||
|
||||
// Compiles a shader of the given |type| with GLSL source |source| and returns
|
||||
// Compiles a shader of the given `type` with GLSL source `source` and returns
|
||||
// the shader handle or 0 on error.
|
||||
GLuint RTCCreateShader(GLenum type, const GLchar *source) {
|
||||
GLuint shader = glCreateShader(type);
|
||||
|
|
|
|||
|
|
@ -273,7 +273,7 @@ CFStringRef ExtractProfile(const webrtc::H264ProfileLevelId &profile_level_id) {
|
|||
}
|
||||
|
||||
// The function returns the max allowed sample rate (pixels per second) that
|
||||
// can be processed by given encoder with |profile_level_id|.
|
||||
// can be processed by given encoder with `profile_level_id`.
|
||||
// See https://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-H.264-201610-S!!PDF-E&type=items
|
||||
// for details.
|
||||
NSUInteger GetMaxSampleRate(const webrtc::H264ProfileLevelId &profile_level_id) {
|
||||
|
|
@ -723,7 +723,7 @@ NSUInteger GetMaxSampleRate(const webrtc::H264ProfileLevelId &profile_level_id)
|
|||
if (_compressionSession) {
|
||||
SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_AverageBitRate, bitrateBps);
|
||||
|
||||
// With zero |_maxAllowedFrameRate|, we fall back to automatic frame rate detection.
|
||||
// With zero `_maxAllowedFrameRate`, we fall back to automatic frame rate detection.
|
||||
if (_maxAllowedFrameRate > 0) {
|
||||
SetVTSessionProperty(
|
||||
_compressionSession, kVTCompressionPropertyKey_ExpectedFrameRate, frameRate);
|
||||
|
|
|
|||
|
|
@ -111,7 +111,7 @@ bool H264CMSampleBufferToAnnexBBuffer(CMSampleBufferRef avcc_sample_buffer,
|
|||
}
|
||||
size_t bytes_remaining = block_buffer_size;
|
||||
while (bytes_remaining > 0) {
|
||||
// The size type here must match |nalu_header_size|, we expect 4 bytes.
|
||||
// The size type here must match `nalu_header_size`, we expect 4 bytes.
|
||||
// Read the length of the next packet of data. Must convert from big endian
|
||||
// to host endian.
|
||||
RTC_DCHECK_GE(bytes_remaining, (size_t)nalu_header_size);
|
||||
|
|
|
|||
|
|
@ -26,7 +26,7 @@ namespace webrtc {
|
|||
|
||||
// Converts a sample buffer emitted from the VideoToolbox encoder into a buffer
|
||||
// suitable for RTP. The sample buffer is in avcc format whereas the rtp buffer
|
||||
// needs to be in Annex B format. Data is written directly to |annexb_buffer|.
|
||||
// needs to be in Annex B format. Data is written directly to `annexb_buffer`.
|
||||
bool H264CMSampleBufferToAnnexBBuffer(CMSampleBufferRef avcc_sample_buffer,
|
||||
bool is_keyframe,
|
||||
rtc::Buffer* annexb_buffer);
|
||||
|
|
@ -34,8 +34,8 @@ bool H264CMSampleBufferToAnnexBBuffer(CMSampleBufferRef avcc_sample_buffer,
|
|||
// Converts a buffer received from RTP into a sample buffer suitable for the
|
||||
// VideoToolbox decoder. The RTP buffer is in annex b format whereas the sample
|
||||
// buffer is in avcc format.
|
||||
// If |is_keyframe| is true then |video_format| is ignored since the format will
|
||||
// be read from the buffer. Otherwise |video_format| must be provided.
|
||||
// If `is_keyframe` is true then `video_format` is ignored since the format will
|
||||
// be read from the buffer. Otherwise `video_format` must be provided.
|
||||
// Caller is responsible for releasing the created sample buffer.
|
||||
bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
|
||||
size_t annexb_buffer_size,
|
||||
|
|
|
|||
|
|
@ -40,9 +40,9 @@ RTC_OBJC_EXPORT
|
|||
- (BOOL)requiresScalingToWidth:(int)width height:(int)height;
|
||||
- (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height;
|
||||
|
||||
/** The minimum size of the |tmpBuffer| must be the number of bytes returned from the
|
||||
/** The minimum size of the `tmpBuffer` must be the number of bytes returned from the
|
||||
* bufferSizeForCroppingAndScalingToWidth:height: method.
|
||||
* If that size is 0, the |tmpBuffer| may be nil.
|
||||
* If that size is 0, the `tmpBuffer` may be nil.
|
||||
*/
|
||||
- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
|
||||
withTempBuffer:(nullable uint8_t *)tmpBuffer;
|
||||
|
|
|
|||
|
|
@ -17,9 +17,9 @@
|
|||
|
||||
namespace webrtc {
|
||||
|
||||
// If |bypass_voice_processing| is true, WebRTC will attempt to disable hardware
|
||||
// If `bypass_voice_processing` is true, WebRTC will attempt to disable hardware
|
||||
// audio processing on iOS.
|
||||
// Warning: Setting |bypass_voice_processing| will have unpredictable
|
||||
// Warning: Setting `bypass_voice_processing` will have unpredictable
|
||||
// consequences for the audio path in the device. It is not advisable to use in
|
||||
// most scenarios.
|
||||
rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
|
||||
|
|
|
|||
|
|
@ -164,7 +164,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
|
|||
bool IsInterrupted();
|
||||
|
||||
private:
|
||||
// Called by the relevant AudioSessionObserver methods on |thread_|.
|
||||
// Called by the relevant AudioSessionObserver methods on `thread_`.
|
||||
void HandleInterruptionBegin();
|
||||
void HandleInterruptionEnd();
|
||||
void HandleValidRouteChange();
|
||||
|
|
@ -173,7 +173,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
|
|||
void HandlePlayoutGlitchDetected();
|
||||
void HandleOutputVolumeChange();
|
||||
|
||||
// Uses current |playout_parameters_| and |record_parameters_| to inform the
|
||||
// Uses current `playout_parameters_` and `record_parameters_` to inform the
|
||||
// audio device buffer (ADB) about our internal audio parameters.
|
||||
void UpdateAudioDeviceBuffer();
|
||||
|
||||
|
|
@ -181,7 +181,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
|
|||
// values may be different once the AVAudioSession has been activated.
|
||||
// This method asks for the current hardware parameters and takes actions
|
||||
// if they should differ from what we have asked for initially. It also
|
||||
// defines |playout_parameters_| and |record_parameters_|.
|
||||
// defines `playout_parameters_` and `record_parameters_`.
|
||||
void SetupAudioBuffersForActiveAudioSession();
|
||||
|
||||
// Creates the audio unit.
|
||||
|
|
|
|||
|
|
@ -386,7 +386,7 @@ OSStatus AudioDeviceIOS::OnDeliverRecordedData(AudioUnitRenderActionFlags* flags
|
|||
// Allocate AudioBuffers to be used as storage for the received audio.
|
||||
// The AudioBufferList structure works as a placeholder for the
|
||||
// AudioBuffer structure, which holds a pointer to the actual data buffer
|
||||
// in |record_audio_buffer_|. Recorded audio will be rendered into this memory
|
||||
// in `record_audio_buffer_`. Recorded audio will be rendered into this memory
|
||||
// at each input callback when calling AudioUnitRender().
|
||||
AudioBufferList audio_buffer_list;
|
||||
audio_buffer_list.mNumberBuffers = 1;
|
||||
|
|
@ -397,7 +397,7 @@ OSStatus AudioDeviceIOS::OnDeliverRecordedData(AudioUnitRenderActionFlags* flags
|
|||
audio_buffer->mData = reinterpret_cast<int8_t*>(record_audio_buffer_.data());
|
||||
|
||||
// Obtain the recorded audio samples by initiating a rendering cycle.
|
||||
// Since it happens on the input bus, the |io_data| parameter is a reference
|
||||
// Since it happens on the input bus, the `io_data` parameter is a reference
|
||||
// to the preallocated audio buffer list that the audio unit renders into.
|
||||
// We can make the audio unit provide a buffer instead in io_data, but we
|
||||
// currently just use our own.
|
||||
|
|
@ -467,7 +467,7 @@ OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
|
|||
|
||||
// Read decoded 16-bit PCM samples from WebRTC (using a size that matches
|
||||
// the native I/O audio unit) and copy the result to the audio buffer in the
|
||||
// |io_data| destination.
|
||||
// `io_data` destination.
|
||||
fine_audio_buffer_->GetPlayoutData(
|
||||
rtc::ArrayView<int16_t>(static_cast<int16_t*>(audio_buffer->mData), num_frames),
|
||||
kFixedPlayoutDelayEstimate);
|
||||
|
|
|
|||
|
|
@ -24,7 +24,7 @@ class NetworkMonitorObserver {
|
|||
public:
|
||||
// Called when a path update occurs, on network monitor dispatch queue.
|
||||
//
|
||||
// |adapter_type_by_name| is a map from interface name (i.e. "pdp_ip0") to
|
||||
// `adapter_type_by_name` is a map from interface name (i.e. "pdp_ip0") to
|
||||
// adapter type, for all available interfaces on the current path. If an
|
||||
// interface name isn't present it can be assumed to be unavailable.
|
||||
virtual void OnPathUpdate(
|
||||
|
|
|
|||
|
|
@ -118,7 +118,7 @@ static const NSUInteger kNumCallbacksPerSecond = 100;
|
|||
// Play out a test file during this time (unit is in seconds).
|
||||
static const NSUInteger kFilePlayTimeInSec = 15;
|
||||
// Run the full-duplex test during this time (unit is in seconds).
|
||||
// Note that first |kNumIgnoreFirstCallbacks| are ignored.
|
||||
// Note that first `kNumIgnoreFirstCallbacks` are ignored.
|
||||
static const NSUInteger kFullDuplexTimeInSec = 10;
|
||||
// Wait for the callback sequence to stabilize by ignoring this amount of the
|
||||
// initial callbacks (avoids initial FIFO access).
|
||||
|
|
|
|||
Loading…
Reference in a new issue