Remove RTCRtpContributingSource dictionary (mdn#30079)

* Move RTCRtpContributingSource info into RTCRtpReceiver.getContributingSources() * Delete RTCRtpContributingSource, redirecting to RTCRtpReceiver.getContributingSources * Remove RTCRtpContributingSource from API top level dictionary list * Minor typo * RTCRtpReceiver.getSynchronizationSources() - add info about returned object
DaddynIssac7 · Nov 9, 2023 · 9716100 · 9716100
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diff --git a/files/en-us/_redirects.txt b/files/en-us/_redirects.txt
@@ -9525,6 +9525,11 @@
 /en-US/docs/Web/API/RTCPeerConnectionIceEvent.candidate	/en-US/docs/Web/API/RTCPeerConnectionIceEvent/candidate
 /en-US/docs/Web/API/RTCRtpCapabilities	/en-US/docs/Web/API/RTCRtpSender/getCapabilities_static
 /en-US/docs/Web/API/RTCRtpCodecCapability	/en-US/docs/Web/API/RTCRtpSender/getCapabilities_static
+/en-US/docs/Web/API/RTCRtpContributingSource	/en-US/docs/Web/API/RTCRtpReceiver/getContributingSources
+/en-US/docs/Web/API/RTCRtpContributingSource/audioLevel	/en-US/docs/Web/API/RTCRtpReceiver/getContributingSources
+/en-US/docs/Web/API/RTCRtpContributingSource/rtpTimestamp	/en-US/docs/Web/API/RTCRtpReceiver/getContributingSources
+/en-US/docs/Web/API/RTCRtpContributingSource/source	/en-US/docs/Web/API/RTCRtpReceiver/getContributingSources
+/en-US/docs/Web/API/RTCRtpContributingSource/timestamp	/en-US/docs/Web/API/RTCRtpReceiver/getContributingSources
 /en-US/docs/Web/API/RTCRtpParameters	/en-US/docs/Web/API/RTCRtpSender/setParameters
 /en-US/docs/Web/API/RTCRtpReceiveParameters	/en-US/docs/Web/API/RTCRtpReceiver/getParameters
 /en-US/docs/Web/API/RTCRtpReceiver/getCapabilities	/en-US/docs/Web/API/RTCRtpReceiver/getCapabilities_static

diff --git a/files/en-us/_wikihistory.json b/files/en-us/_wikihistory.json
@@ -55024,26 +55024,6 @@
     "modified": "2019-05-20T18:28:16.993Z",
     "contributors": ["Sheppy"]
   },
-  "Web/API/RTCRtpContributingSource": {
-    "modified": "2020-10-15T21:55:35.352Z",
-    "contributors": ["Sheppy", "fscholz", "Onkar316", "jpmedley"]
-  },
-  "Web/API/RTCRtpContributingSource/audioLevel": {
-    "modified": "2020-10-15T21:55:34.869Z",
-    "contributors": ["fscholz", "Sheppy", "jpmedley"]
-  },
-  "Web/API/RTCRtpContributingSource/rtpTimestamp": {
-    "modified": "2020-10-15T22:25:34.965Z",
-    "contributors": ["Sheppy"]
-  },
-  "Web/API/RTCRtpContributingSource/source": {
-    "modified": "2020-10-15T21:55:40.516Z",
-    "contributors": ["fscholz", "Sheppy", "jpmedley"]
-  },
-  "Web/API/RTCRtpContributingSource/timestamp": {
-    "modified": "2020-10-15T21:55:38.833Z",
-    "contributors": ["sideshowbarker", "fscholz", "Sheppy", "jpmedley"]
-  },
   "Web/API/RTCRtpEncodingParameters": {
     "modified": "2020-10-15T22:01:53.269Z",
     "contributors": ["Sheppy"]

diff --git a/files/en-us/mozilla/firefox/releases/72/index.md b/files/en-us/mozilla/firefox/releases/72/index.md
@@ -91,7 +91,7 @@ _No changes._
 #### Media, Web Audio, and WebRTC
 
 - The {{domxref("MediaDevices.getDisplayMedia()")}} method can now only be called in response to a user gesture such as a [`click`](/en-US/docs/Web/API/Element/click_event) event ([Firefox bug 1580944](https://bugzil.la/1580944)).
-- The {{domxref("RTCRtpContributingSource")}} dictionary can now include the {{domxref("RTCRtpContributingSource.rtpTimestamp", "rtpTimestamp")}} property, which is a source-generated time at which the packet's media was generated or sampled ([Firefox bug 1583867](https://bugzil.la/1583867)).
+- The `RTCRtpContributingSource` dictionary can now include the `rtpTimestamp` property, which is a source-generated time at which the packet's media was generated or sampled ([Firefox bug 1583867](https://bugzil.la/1583867)).
 
 #### Removals
 

diff --git a/files/en-us/web/api/rtcrtpcontributingsource/audiolevel/index.md b/files/en-us/web/api/rtcrtpcontributingsource/audiolevel/index.md
diff --git a/files/en-us/web/api/rtcrtpcontributingsource/index.md b/files/en-us/web/api/rtcrtpcontributingsource/index.md
diff --git a/files/en-us/web/api/rtcrtpcontributingsource/rtptimestamp/index.md b/files/en-us/web/api/rtcrtpcontributingsource/rtptimestamp/index.md
diff --git a/files/en-us/web/api/rtcrtpcontributingsource/source/index.md b/files/en-us/web/api/rtcrtpcontributingsource/source/index.md
diff --git a/files/en-us/web/api/rtcrtpcontributingsource/timestamp/index.md b/files/en-us/web/api/rtcrtpcontributingsource/timestamp/index.md
diff --git a/files/en-us/web/api/rtcrtpreceiver/getcontributingsources/index.md b/files/en-us/web/api/rtcrtpreceiver/getcontributingsources/index.md
@@ -8,11 +8,7 @@ browser-compat: api.RTCRtpReceiver.getContributingSources
 
 {{APIRef("WebRTC API")}}
 
-The **`getContributingSources()`**
-method of the {{domxref("RTCRtpReceiver")}} interface returns an array of
-{{domxref("RTCRtpContributingSource")}} instances, each corresponding to one CSRC
-(contributing source) identifier received by the current `RTCRtpReceiver`
-in the last ten seconds.
+The **`getContributingSources()`** method of the {{domxref("RTCRtpReceiver")}} interface returns an array of objects, each corresponding to one CSRC (contributing source) identifier received by the current `RTCRtpReceiver` in the last ten seconds.
 
 ## Syntax
 
@@ -26,9 +22,32 @@ None.
 
 ### Return value
 
-An array of {{domxref("RTCRtpContributingSource")}} instances. Each instance describes
-one of the contributing sources that provided data to the incoming stream in the past
-ten seconds.
+An array of objects, each describing one of the contributing sources that provided data to the incoming stream in the past ten seconds.
+These objects contain the following properties:
+
+- `audioLevel` {{optional_inline}}
+
+  - : A floating-point value between 0.0 and 1.0 specifying the audio level contained in the last RTP packet played from the contributing source.
+
+    The value is on a linear scale and is defined in units of dBov, or decibels (overload).
+    This is the amplitude relative to the point at which clipping of the audio begins to occur.
+    A value of 1.0 represents 0 dBov (maximum volume), a value of 0.0 represents silence, and a value of 0.5 represents approximately 6 dB SPL (decibels of sound pressure level) change in the sound pressure level from 0 dBov.
+
+    If this property is not present then no volume level was provided by the source.
+
+- `rtpTimestamp` {{optional_inline}}
+
+  - : The RTP timestamp (an integer {{domxref("DOMHighResTimeStamp")}}) of the media.
+    This source-generated timestamp indicates the time at which the media in this packet, scheduled for play out at the time indicated by `timestamp`, was initially sampled or generated.
+    It may be useful for sequencing and synchronization purposes.
+
+- `source` {{optional_inline}}
+
+  - : A positive integer value specifying the CSRC identifier of the contributing source.
+    This uniquely identifies the source of the particular stream RTP packets.
+
+- `timestamp` {{optional_inline}}
+  - : A {{domxref("DOMHighResTimeStamp")}} indicating the most recent time at which a frame originating from this source was delivered to the receiver's {{domxref("MediaStreamTrack")}}
 
 ## Specifications
 

diff --git a/files/en-us/web/api/rtcrtpreceiver/getsynchronizationsources/index.md b/files/en-us/web/api/rtcrtpreceiver/getsynchronizationsources/index.md
@@ -8,11 +8,7 @@ browser-compat: api.RTCRtpReceiver.getSynchronizationSources
 
 {{APIRef("WebRTC API")}}
 
-The **`getSynchronizationSources()`**
-method of the {{domxref("RTCRtpReceiver")}} interface returns an array of
-{{domxref("RTCRtpContributingSource")}} instances, each corresponding to one SSRC
-(synchronization source) identifier received by the current
-`RTCRtpReceiver` in the last ten seconds.
+The **`getSynchronizationSources()`** method of the {{domxref("RTCRtpReceiver")}} interface returns an array of objects, each corresponding to one SSRC (synchronization source) identifier received by the current `RTCRtpReceiver` in the last ten seconds.
 
 ## Syntax
 
@@ -26,16 +22,32 @@ None.
 
 ### Return value
 
-An array of object describing one of the synchronization sources that provided data to the incoming stream in the past ten seconds. These objects contain the following properties:
+An array of objects, each describing one of the synchronization sources that provided data to the incoming stream in the past ten seconds.
+These objects contain the following properties:
 
-- {{domxref("RTCRtpContributingSource.audioLevel", "audioLevel")}} {{optional_inline}}
-  - : A double-precision floating-point value between 0.0 and 1.0 specifying the audio level (in dBov) contained in the last RTP packet played from this source.
-- {{domxref("RTCRtpContributingSource.rtpTimestamp", "rtpTimestamp")}} {{optional_inline}}
-  - : The RTP timestamp of the media played out at the time indicated by `timestamp`. This value is a source-generated time value which can be used to help with sequencing and synchronization.
-- {{domxref("RTCRtpContributingSource.source", "source")}} {{optional_inline}}
-  - : A 32-bit unsigned integer value specifying the CSRC identifier of the contributing source.
-- {{domxref("RTCRtpContributingSource.timestamp", "timestamp")}} {{optional_inline}}
-  - : A {{domxref("DOMHighResTimeStamp")}} indicating the most recent time at which a frame originating from this source was delivered to the receiver's {{domxref("MediaStreamTrack")}}.
+- `audioLevel`
+
+  - : A floating-point value between 0.0 and 1.0 specifying the audio level contained in the last RTP packet played from the synchronization source.
+
+    The value is on a linear scale and is defined in units of dBov, or decibels (overload).
+    This is the amplitude relative to the point at which clipping of the audio begins to occur.
+    A value of 1.0 represents 0 dBov (maximum volume), a value of 0.0 represents silence, and a value of 0.5 represents approximately 6 dB SPL (decibels of sound pressure level) change in the sound pressure level from 0 dBov.
+
+    This value is required and always present.
+
+- `rtpTimestamp` {{optional_inline}}
+
+  - : The RTP timestamp (an integer {{domxref("DOMHighResTimeStamp")}}) of the media.
+    This source-generated timestamp indicates the time at which the media in this packet, scheduled for play out at the time indicated by `timestamp`, was initially sampled or generated.
+    It may be useful for sequencing and synchronization purposes.
+
+- `source` {{optional_inline}}
+
+  - : A positive integer value specifying the SSRC identifier of the synchronization source.
+    This uniquely identifies the source of the particular stream RTP packets.
+
+- `timestamp` {{optional_inline}}
+  - : A {{domxref("DOMHighResTimeStamp")}} indicating the most recent time at which a frame originating from this source was delivered to the receiver's {{domxref("MediaStreamTrack")}}
 
 ## Specifications
 

diff --git a/files/en-us/web/api/rtcrtpreceiver/index.md b/files/en-us/web/api/rtcrtpreceiver/index.md
@@ -31,13 +31,13 @@ The **`RTCRtpReceiver`** interface of the [WebRTC API](/en-US/docs/Web/API/WebRT
 ## Instance methods
 
 - {{domxref("RTCRtpReceiver.getContributingSources()")}}
-  - : Returns an array of {{domxref("RTCRtpContributingSource")}} instances for each unique CSRC (contributing source) identifier received by the current `RTCRtpReceiver` in the last ten seconds.
+  - : Returns an array that contains an object for each unique CSRC (contributing source) identifier received by the current `RTCRtpReceiver` in the last ten seconds.
 - {{domxref("RTCRtpReceiver.getParameters()")}}
   - : Returns an object that contains information about how the RTC data is to be decoded.
 - {{domxref("RTCRtpReceiver.getStats()")}}
   - : Returns a {{jsxref("Promise")}} whose fulfillment handler receives a {{domxref("RTCStatsReport")}} which contains statistics about the incoming streams and their dependencies.
 - {{domxref("RTCRtpReceiver.getSynchronizationSources()")}}
-  - : Returns an array including one `RTCRtpSynchronizationSource` instance for each unique SSRC (synchronization source) identifier received by the current `RTCRtpReceiver` in the last ten seconds.
+  - : Returns an array that contains an object for each unique SSRC (synchronization source) identifier received by the current `RTCRtpReceiver` in the last ten seconds.
 
 ## Specifications
 

diff --git a/files/en-us/web/api/webrtc_api/index.md b/files/en-us/web/api/webrtc_api/index.md
@@ -63,11 +63,6 @@ These interfaces, dictionaries, and types are used to set up, open, and manage W
 - {{DOMxRef("RTCSctpTransport")}}
   - : Provides information which describes a Stream Control Transmission Protocol (**{{Glossary("SCTP")}}**) transport and also provides a way to access the underlying Datagram Transport Layer Security (**{{Glossary("DTLS")}}**) transport over which SCTP packets for all of an [`RTCPeerConnection`](/en-US/docs/Web/API/RTCPeerConnection)'s data channels are sent and received.
 
-#### Dictionaries
-
-- {{DOMxRef("RTCRtpContributingSource")}}
-  - : Contains information about a given contributing source (CSRC) including the most recent time a packet that the source contributed was played out.
-
 #### Events
 
 - {{domxref("RTCDataChannel.bufferedamountlow_event", "bufferedamountlow")}}