namespace Tizen.Multimedia.Remoting
{
/// <summary>
- /// Represents a camera source.
+ /// Represents a media source that makes video data using camera device internally.
/// </summary>
/// <remarks>The camera privilege(http://tizen.org/privilege/camera) is required.</remarks>
/// <seealso cref="WebRTC.AddSource"/>
public MediaCameraSource() : base(MediaType.Video) {}
/// <summary>
- /// Gets or sets the camera device id.
+ /// Gets or sets the camera device id that identifies each camera device.
/// </summary>
/// <value>A value that specifies the camera device id.</value>
/// <exception cref="InvalidOperationException">MediaSource is not attached yet.</exception>
namespace Tizen.Multimedia.Remoting
{
/// <summary>
- /// Represents a audio, video custom source.
+ /// Represents a media source that makes audio, video data internally.
/// </summary>
+ /// <remarks>
+ /// This supports the product infrastructure and is not intended to be used directly from application code.
+ /// </remarks>
/// <seealso cref="WebRTC.AddSource"/>
/// <seealso cref="WebRTC.AddSources"/>
/// <since_tizen> 9 </since_tizen>
namespace Tizen.Multimedia.Remoting
{
/// <summary>
- /// Represents a media source with contents read from a file.
+ /// Represents a media source that reads media data from a file.
/// </summary>
/// <remarks>
/// Depending on where the source file is located either the media storage privilege (http://tizen.org/privilege/mediastorage) is required or<br/>
namespace Tizen.Multimedia.Remoting
{
/// <summary>
- /// Represents a microphone source.
+ /// Represents a media source that makes audio data using microphone.
/// </summary>
/// <remarks>The recorder privilege(http://tizen.org/privilege/recorder) is required.</remarks>
/// <seealso cref="WebRTC.AddSource"/>
namespace Tizen.Multimedia.Remoting
{
- /// <summary>Represents a null source.</summary>
+ /// <summary>
+ /// Represents a media source that only receives media streams from peer.
+ /// </summary>
/// <remarks>
/// If you add this source, WebRTC only receives media stream.<br/>
/// <see cref="TransceiverDirection"/> is set <see cref="TransceiverDirection.RecvOnly"/> by default.
}
/// <summary>
- /// Retrieves the supported transceiver codecs.
+ /// Retrieves the supported transceiver codecs for receiving media stream.
/// </summary>
/// <param name="type">The media type.</param>
/// <returns>The supported transceiver codecs.</returns>
}
/// <summary>
- /// Gets the source ID.
+ /// Gets the source ID of media packet.
/// </summary>
/// <value>The source ID.</value>
/// <since_tizen> 9 </since_tizen>
public uint SourceId { get; }
/// <summary>
- /// Gets the media packet buffer status.
+ /// Gets the media packet buffer status that indicates underrun or overflow.
/// </summary>
/// <value>The media packet buffer status.</value>
/// <since_tizen> 9 </since_tizen>
namespace Tizen.Multimedia.Remoting
{
/// <summary>
- /// Represents a media packet source.
+ /// Represents a media source that wraps media data into MediaPacket.
/// </summary>
/// <seealso cref="WebRTC.AddSource"/>
/// <seealso cref="WebRTC.AddSources"/>
namespace Tizen.Multimedia.Remoting
{
/// <summary>
- /// Provides means to configure properties and handle events for <see cref="MediaPacketSource"/>.
+ /// Provides means to configure properties and handle event for <see cref="MediaPacketSource"/>.
/// </summary>
/// <seealso cref="MediaPacketSource"/>
/// <since_tizen> 9 </since_tizen>
}
/// <summary>
- /// Occurs when the buffer underruns or overflows.
+ /// Occurs when the status of buffer is underruns or overflows.
/// </summary>
/// <remarks>The event handler will be executed on an internal thread.</remarks>
/// <since_tizen> 9 </since_tizen>
namespace Tizen.Multimedia.Remoting
{
/// <summary>
- /// Represents a screen source.
+ /// Represents a media source to share screen data.
/// </summary>
/// <remarks>
/// MediaScreenSource is not allowed to be used by third-party applications due to the security reasons. (Since Tizen 7.0)
namespace Tizen.Multimedia.Remoting
{
/// <summary>
- /// Provides the ability to control audio/video track.
+ /// Provides the ability to control audio or video track from peer.
/// </summary>
/// <since_tizen> 9 </since_tizen>
public sealed class MediaStreamTrack : IDisplayable<WebRTCErrorCode>
}
/// <summary>
- /// Gets the the of media stream track.
+ /// Gets the type of media stream track.
/// </summary>
/// <value><see cref="MediaType"/></value>
/// <since_tizen> 9 </since_tizen>
}
/// <summary>
- /// Gets or sets the display to show remote video.
+ /// Gets or sets the display to show video data from peer.
/// </summary>
/// <value>A <see cref="Multimedia.Display"/> that specifies the display.</value>
/// <remarks>
namespace Tizen.Multimedia.Remoting
{
/// <summary>
- /// Represents an audio or a video test source.
+ /// Represents a media source that makes test audio or video stream.
/// </summary>
/// <seealso cref="WebRTC.AddSource"/>
/// <seealso cref="WebRTC.AddSources"/>
#endregion
/// <summary>
- /// Starts the WebRTC.
+ /// Starts the WebRTC with specific media source.
/// </summary>
/// <remarks>
/// The WebRTC must be in the <see cref="WebRTCState.Idle"/> state.<br/>
}
/// <summary>
- /// Starts the WebRTC asynchronously.
+ /// Starts the WebRTC asynchronously with specific media source.
/// </summary>
/// <remarks>
/// The WebRTC must be in the <see cref="WebRTCState.Idle"/> state.<br/>
}
/// <summary>
- /// Creates SDP offer asynchronously to start a new WebRTC connection to a remote peer.
+ /// Creates SDP(Session Description Protocol) offer asynchronously to start a new WebRTC connection to a remote peer.
/// </summary>
/// <remarks>
/// The WebRTC must be in the <see cref="WebRTCState.Negotiating"/> or <see cref="WebRTCState.Playing"/>(Since API Level 12)
}
/// <summary>
- /// Creates SDP answer asynchronously with option to an offer received from a remote peer.
+ /// Creates SDP(Session Description Protocol) answer asynchronously with option to an offer received from a remote peer.
/// </summary>
/// <remarks>
/// The WebRTC must be in the <see cref="WebRTCState.Negotiating"/> or <see cref="WebRTCState.Playing"/>(Since API Level 12)
}
/// <summary>
- /// Sets the session description of the remote peer's current offer or answer.
+ /// Sets the offer or answer session description from the current remote peer.
/// </summary>
/// <remarks>
/// The WebRTC must be in the <see cref="WebRTCState.Negotiating"/> or <see cref="WebRTCState.Playing"/>(Since API Level 12)
}
/// <summary>
- /// Gets the session description of the remote peer's current offer or answer.
+ /// Gets the offer or answer session description from the current remote peer.
/// </summary>
/// <value>The remote session description string</value>
/// <exception cref="ObjectDisposedException">The WebRTC has already been disposed.</exception>
}
/// <summary>
- /// Adds media source.
+ /// Adds media source to the current WebRTC.
/// </summary>
/// <remarks>
/// This method does not throw state exception anymore(Since API Level 12). It can be called in any state.<br/>
}
/// <summary>
- /// Adds media sources.
+ /// Adds media sources from the current WebRTC.
/// </summary>
/// <remarks>
/// This method does not throw state exception anymore(Since API Level 12). It can be called in any state.<br/>
}
/// <summary>
- /// Removes media source.
+ /// Removes media source from the current WebRTC.
/// </summary>
/// <remarks>
/// This method does not throw state exception anymore(Since API Level 12). It can be called in any state.<br/>
}
/// <summary>
- /// Removes media sources.
+ /// Removes media sources from the current WebRTC.
/// </summary>
/// <remarks>
/// This method does not throw state exception anymore(Since API Level 12). It can be called in any state.<br/>
}
/// <summary>
- /// Sets a turn server.
+ /// Sets a turn server for signalling with remote peer which cannot be connected directly.
/// </summary>
/// <exception cref="ArgumentNullException">The <paramref name="turnServer"/> is null.</exception>
/// <exception cref="ObjectDisposedException">The WebRTC has already been disposed.</exception>
}
/// <summary>
- /// Sets turn servers.
+ /// Sets turn servers for signalling with remote peer which cannot be connected directly.
/// </summary>
/// <exception cref="ArgumentNullException">The one of <paramref name="turnServers"/> is null.</exception>
/// <exception cref="ObjectDisposedException">The WebRTC has already been disposed.</exception>
}
/// <summary>
- /// Gets the error.
+ /// Gets the error of data channel.
/// </summary>
/// <value>The error.</value>
/// <since_tizen> 9 </since_tizen>
namespace Tizen.Multimedia.Remoting
{
/// <summary>
- /// Specifies errors.
+ /// Specifies WebRTC errors.
/// </summary>
/// <seealso cref="WebRTC.ErrorOccurred"/>
/// <seealso cref="WebRTCErrorOccurredEventArgs"/>
}
/// <summary>
- /// Specifies the display type.
+ /// Specifies the display mode that indicates the way of displaying video data.
/// </summary>
/// <since_tizen> 9 </since_tizen>
public enum WebRTCDisplayMode
}
/// <summary>
- /// Specifies the bundle policy.
+ /// Specifies the bundle policy that affects which media tracks are negotiated if the remote endpoint is not bundle-aware,
+ /// and what ICE candidates are gathered.
/// </summary>
/// <remarks>
/// The details of bundle policy enum is described in https://www.w3.org/TR/webrtc/#rtcbundlepolicy-enum.
}
/// <summary>
- /// Gets the track information.
+ /// Gets the information of media stream track from remote peer.
/// </summary>
/// <value>The media type.</value>
/// <since_tizen> 9 </since_tizen>
}
/// <summary>
- /// Gets the media type.
+ /// Gets the media stream track from reemote peer.
/// </summary>
/// <value>The media type.</value>
/// <since_tizen> 9 </since_tizen>
projects / platform / core / csapi / tizenfx.git / commitdiff