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Changeset 266052 in webkit


Ignore:
Timestamp:
Aug 24, 2020, 2:26:58 AM (6 years ago)
Author:
commit-queue@webkit.org
Message:

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):

Location:
trunk
Files:
7 edited

Legend:

Unmodified
Added
Removed

  • trunk/LayoutTests/imported/w3c/ChangeLog

    r266038 r266052  
     12020-08-24  Justin Uberti  <justin@uberti.name>
     2
     3        RTCRtpSynchronizationSource.rtpTimestamp is not present
     4        https://bugs.webkit.org/show_bug.cgi?id=215722
     5
     6        Reviewed by Youenn Fablet.
     7
     8        Updated expectations file to indicate that tests checking for .rtpTimestamp now pass.
     9
     10        * LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt:
     11
    1122020-08-22  Emilio Cobos Álvarez  <emilio@crisal.io>
    213

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

    r264202 r266052  
    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

  • trunk/Source/WebCore/ChangeLog

    r266051 r266052  
     12020-08-24  Justin Uberti  <justin@uberti.name>
     2
     3        RTCRtpSynchronizationSource.rtpTimestamp is not present
     4        https://bugs.webkit.org/show_bug.cgi?id=215722
     5
     6        Reviewed by Youenn Fablet.
     7
     8        Updated expected results in LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https-expected.txt.
     9
     10        * Modules/mediastream/RTCRtpContributingSource.idl:
     11        * Modules/mediastream/RTCRtpContributingSource.idl:
     12        * Modules/mediastream/RTCRtpSynchronizationSource.idl:
     13        Minor modification to ensure JSRTCRtpSynchronizationSource.cpp gets regenerated.
     14        * Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp:
     15        (WebCore::fillRTCRtpContributingSource):
     16
    1172020-08-23  Wenson Hsieh  <wenson_hsieh@apple.com>
    218

  • trunk/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.h

    r239427 r266052  
    3232struct RTCRtpContributingSource {
    3333    double timestamp;
     34    unsigned long rtpTimestamp;
    3435    unsigned long source;
    3536    Optional<double> audioLevel;

  • trunk/Source/WebCore/Modules/mediastream/RTCRtpContributingSource.idl

    r236147 r266052  
    3232] dictionary RTCRtpContributingSource {
    3333    required DOMHighResTimeStamp timestamp;
     34    required unsigned long rtpTimestamp;
    3435    required unsigned long source;
    3536    double audioLevel;

  • trunk/Source/WebCore/Modules/mediastream/RTCRtpSynchronizationSource.idl

    r236147 r266052  
    2929    Conditional=WEB_RTC,
    3030    EnabledAtRuntime=PeerConnection,
    31     JSGenerateToJSObject,
     31    JSGenerateToJSObject
    3232] dictionary RTCRtpSynchronizationSource : RTCRtpContributingSource {
    3333    boolean voiceActivityFlag;

  • trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp

    r258493 r266052  
    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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