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Changeset 269793 in webkit


Ignore:
Timestamp:
Nov 13, 2020, 2:16:12 PM (6 years ago)
Author:
Russell Epstein
Message:

Cherry-pick r266052. rdar://problem/71378190

RTCRtpSynchronizationSource.rtpTimestamp is not present
https://bugs.webkit.org/show_bug.cgi?id=215722

Patch by Justin Uberti <justin@uberti.name> on 2020-08-24
Reviewed by Youenn Fablet.

LayoutTests/imported/w3c:

Updated expectations file to indicate that tests checking for .rtpTimestamp now pass.

  • LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt:

Source/WebCore:

Updated expected results in LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt.

  • Modules/mediastream/RTCRtpContributingSource.idl:
  • Modules/mediastream/RTCRtpContributingSource.idl:
  • Modules/mediastream/RTCRtpSynchronizationSource.idl: Minor modification to ensure JSRTCRtpSynchronizationSource.cpp gets regenerated.
  • Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp: (WebCore::fillRTCRtpContributingSource):

git-svn-id: https://svn.webkit.org/repository/webkit/trunk@266052 268f45cc-cd09-0410-ab3c-d52691b4dbfc

Location:
branches/safari-610-branch
Files:
7 edited

Legend:

Unmodified
Added
Removed
  • branches/safari-610-branch/LayoutTests/imported/w3c/ChangeLog

    r268100 r269793  
     12020-11-13  Russell Epstein  <repstein@apple.com>
     2
     3        Cherry-pick r266052. rdar://problem/71378190
     4
     5    RTCRtpSynchronizationSource.rtpTimestamp is not present
     6    https://bugs.webkit.org/show_bug.cgi?id=215722
     7   
     8    Patch by Justin Uberti <justin@uberti.name> on 2020-08-24
     9    Reviewed by Youenn Fablet.
     10   
     11    LayoutTests/imported/w3c:
     12   
     13    Updated expectations file to indicate that tests checking for .rtpTimestamp now pass.
     14   
     15    * LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt:
     16   
     17    Source/WebCore:
     18   
     19    Updated expected results in LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt.
     20   
     21    * Modules/mediastream/RTCRtpContributingSource.idl:
     22    * Modules/mediastream/RTCRtpContributingSource.idl:
     23    * Modules/mediastream/RTCRtpSynchronizationSource.idl:
     24    Minor modification to ensure JSRTCRtpSynchronizationSource.cpp gets regenerated.
     25    * Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp:
     26    (WebCore::fillRTCRtpContributingSource):
     27   
     28    git-svn-id: https://svn.webkit.org/repository/webkit/trunk@266052 268f45cc-cd09-0410-ab3c-d52691b4dbfc
     29
     30    2020-08-24  Justin Uberti  <justin@uberti.name>
     31
     32            RTCRtpSynchronizationSource.rtpTimestamp is not present
     33            https://bugs.webkit.org/show_bug.cgi?id=215722
     34
     35            Reviewed by Youenn Fablet.
     36
     37            Updated expectations file to indicate that tests checking for .rtpTimestamp now pass.
     38
     39            * LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt:
     40
    1412020-10-06  Alan Coon  <alancoon@apple.com>
    242
  • branches/safari-610-branch/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt

    r264202 r269793  
    22PASS [audio] getSynchronizationSources() eventually returns a non-empty list
    33PASS [audio] RTCRtpSynchronizationSource.timestamp is a number
    4 FAIL [audio] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1] assert_equals: expected "number" but got "undefined"
     4PASS [audio] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1]
    55PASS [audio] getSynchronizationSources() does not contain SSRCs older than 10 seconds
    66FAIL [audio] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now() assert_true: expected true got false
     
    88PASS [video] getSynchronizationSources() eventually returns a non-empty list
    99PASS [video] RTCRtpSynchronizationSource.timestamp is a number
    10 FAIL [video] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1] assert_equals: expected "number" but got "undefined"
     10PASS [video] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1]
    1111PASS [video] getSynchronizationSources() does not contain SSRCs older than 10 seconds
    1212FAIL [video] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now() assert_true: expected true got false
  • branches/safari-610-branch/Source/WebCore/ChangeLog

    r269411 r269793  
     12020-11-13  Russell Epstein  <repstein@apple.com>
     2
     3        Cherry-pick r266052. rdar://problem/71378190
     4
     5    RTCRtpSynchronizationSource.rtpTimestamp is not present
     6    https://bugs.webkit.org/show_bug.cgi?id=215722
     7   
     8    Patch by Justin Uberti <justin@uberti.name> on 2020-08-24
     9    Reviewed by Youenn Fablet.
     10   
     11    LayoutTests/imported/w3c:
     12   
     13    Updated expectations file to indicate that tests checking for .rtpTimestamp now pass.
     14   
     15    * LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt:
     16   
     17    Source/WebCore:
     18   
     19    Updated expected results in LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt.
     20   
     21    * Modules/mediastream/RTCRtpContributingSource.idl:
     22    * Modules/mediastream/RTCRtpContributingSource.idl:
     23    * Modules/mediastream/RTCRtpSynchronizationSource.idl:
     24    Minor modification to ensure JSRTCRtpSynchronizationSource.cpp gets regenerated.
     25    * Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp:
     26    (WebCore::fillRTCRtpContributingSource):
     27   
     28    git-svn-id: https://svn.webkit.org/repository/webkit/trunk@266052 268f45cc-cd09-0410-ab3c-d52691b4dbfc
     29
     30    2020-08-24  Justin Uberti  <justin@uberti.name>
     31
     32            RTCRtpSynchronizationSource.rtpTimestamp is not present
     33            https://bugs.webkit.org/show_bug.cgi?id=215722
     34
     35            Reviewed by Youenn Fablet.
     36
     37            Updated expected results in LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt.
     38
     39            * Modules/mediastream/RTCRtpContributingSource.idl:
     40            * Modules/mediastream/RTCRtpContributingSource.idl:
     41            * Modules/mediastream/RTCRtpSynchronizationSource.idl:
     42            Minor modification to ensure JSRTCRtpSynchronizationSource.cpp gets regenerated.
     43            * Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp:
     44            (WebCore::fillRTCRtpContributingSource):
     45
    1462020-11-04  Kocsen Chung  <kocsen_chung@apple.com>
    247
  • branches/safari-610-branch/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.h

    r239427 r269793  
    3232struct RTCRtpContributingSource {
    3333    double timestamp;
     34    unsigned long rtpTimestamp;
    3435    unsigned long source;
    3536    Optional<double> audioLevel;
  • branches/safari-610-branch/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.idl

    r236147 r269793  
    3232] dictionary RTCRtpContributingSource {
    3333    required DOMHighResTimeStamp timestamp;
     34    required unsigned long rtpTimestamp;
    3435    required unsigned long source;
    3536    double audioLevel;
  • branches/safari-610-branch/Source/WebCore/Modules/mediastream/RTCRtpSynchronizationSource.idl

    r236147 r269793  
    2929    Conditional=WEB_RTC,
    3030    EnabledAtRuntime=PeerConnection,
    31     JSGenerateToJSObject,
     31    JSGenerateToJSObject
    3232] dictionary RTCRtpSynchronizationSource : RTCRtpContributingSource {
    3333    boolean voiceActivityFlag;
  • branches/safari-610-branch/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp

    r258493 r269793  
    4242{
    4343    source.timestamp = rtcSource.timestamp_ms();
     44    source.rtpTimestamp = rtcSource.rtp_timestamp();
    4445    source.source = rtcSource.source_id();
    4546    if (rtcSource.audio_level())
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