This reverts commit e20fbb00d0e0219b710da24664e81a10b12c703a.
Reason for revert: Breaks WebRTC roll to Chromium, see:
https://chromium-review.googlesource.com/c/chromium/src/+/6218060
Example of error: https://ci.chromium.org/ui/p/chrome/builders/ci/win-arm64-rel-ready/51821/test-results?sortby=&groupby=
Original change's description:
> Get DeviceScaleFactor for the captured monitor/screen
>
> Accesses DeviceScaleFactor using the windows API
> GetScaleFactorForMonitor and adds it to the DesktopFrame. In a follow-up
> CL, this value is propagated to
> DesktopCaptureDevice::Core::OnCaptureResult where it is added to the
> frame metadata.
>
> In a follow-up CL, add RegisterScaleChangeEvent to get notified whenever
> the device scale factor changes.
>
> Design doc: go/expose-captured-surface-resolution
>
> Bug: chromium:383946052
> Change-Id: I363af33c569419d95ddf31a0cc2f9cecf6fb0c7b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374344
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Mark Foltz <mfoltz@chromium.org>
> Commit-Queue: Palak Agarwal <agpalak@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43827}
Bug: chromium:383946052
Change-Id: I3065b278939ca0e686ee6da0f0721082bc0c99e8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375902
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43829}
Accesses DeviceScaleFactor using the windows API
GetScaleFactorForMonitor and adds it to the DesktopFrame. In a follow-up
CL, this value is propagated to
DesktopCaptureDevice::Core::OnCaptureResult where it is added to the
frame metadata.
In a follow-up CL, add RegisterScaleChangeEvent to get notified whenever
the device scale factor changes.
Design doc: go/expose-captured-surface-resolution
Bug: chromium:383946052
Change-Id: I363af33c569419d95ddf31a0cc2f9cecf6fb0c7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374344
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43827}
This change expands on https://webrtc-review.googlesource.com/c/src/+/374420 to cover the error produced when copying microphone samples.
Change-Id: I7aa58c9c9ac175d5f4cfdb60bbd3f16334c03c1b
Bug: webrtc:390314937
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375540
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43826}
This reverts commit 4de5839c1117e5bb96148c8575a74a69bde02768.
Reason for revert: Bug fixed + DCHECK added
Original change's description:
> Revert "Move piggybacking controller from P2PTC to DTLS transport"
>
> This reverts commit 29e639e0a495a537c610182ab9b04aed8cf10426.
>
> Reason for revert: found bug accessing variable after it has been moved.
>
> Original change's description:
> > Move piggybacking controller from P2PTC to DTLS transport
> >
> > The DTLS-STUN piggybacking controller is associated with both the DTLS
> > transport and the ICE transport (P2PTransportChannel). It turned out to
> > be more closely associated with the DTLS transport and requires less
> > plumbing when moved there.
> >
> > The config option to enable the feature remains as part of the ICE
> > transport config since the ICE transport does not know its "upstream"
> > DTLS transport and hence can not query the config from it.
> >
> > BUG=webrtc:367395350
> >
> > Change-Id: Iafd5abd8b65855bcf32bf840414d96513d8e6300
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375283
> > Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> > Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43823}
>
> Bug: webrtc:367395350
> Change-Id: I2d83de8890b0aa230dd9e21cb5ce2eb03c8d3564
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375861
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43824}
Bug: webrtc:367395350
Change-Id: I4b4acccf15de565736b072ca2de88a1551a6378e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375862
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43825}
This reverts commit 29e639e0a495a537c610182ab9b04aed8cf10426.
Reason for revert: found bug accessing variable after it has been moved.
Original change's description:
> Move piggybacking controller from P2PTC to DTLS transport
>
> The DTLS-STUN piggybacking controller is associated with both the DTLS
> transport and the ICE transport (P2PTransportChannel). It turned out to
> be more closely associated with the DTLS transport and requires less
> plumbing when moved there.
>
> The config option to enable the feature remains as part of the ICE
> transport config since the ICE transport does not know its "upstream"
> DTLS transport and hence can not query the config from it.
>
> BUG=webrtc:367395350
>
> Change-Id: Iafd5abd8b65855bcf32bf840414d96513d8e6300
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375283
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43823}
Bug: webrtc:367395350
Change-Id: I2d83de8890b0aa230dd9e21cb5ce2eb03c8d3564
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375861
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43824}
The DTLS-STUN piggybacking controller is associated with both the DTLS
transport and the ICE transport (P2PTransportChannel). It turned out to
be more closely associated with the DTLS transport and requires less
plumbing when moved there.
The config option to enable the feature remains as part of the ICE
transport config since the ICE transport does not know its "upstream"
DTLS transport and hence can not query the config from it.
BUG=webrtc:367395350
Change-Id: Iafd5abd8b65855bcf32bf840414d96513d8e6300
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375283
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43823}
When capturing only one display, it is unnecessary to use
DoDuplicateAll, which allocates bitmap image memory for a rectangle
encompassing all screens and captures all displays. In this case, I
switch to DoDuplicateOne.
I have added an int parameter to GetNumFrameCaptured and
EnsureFrameCaptured to distinguish which display's skip behavior is
currently being executed.
After the modification, when capturing a single monitor in a
multi-monitor environment, only the bitmap image memory of the size of
the captured monitor will be allocated, rather than for all monitors.
Bug: webrtc:391914255
Change-Id: Iee56796c50282beaf1dbeef47f5937fe7fe94a05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375181
Reviewed-by: Joe Downing <joedow@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#43822}
NetworkStateEstimator is not used by WebRTC from receive side.
ReceiveSidesCongestionController::SetTransportOverhead is not needed either since NetworkStateEstimator is removed.
Note, CongestionControlFeedbackGenerator is used with RFC 8888 only and feedback frequency will be refactored in later cl.
Bug: webrtc:42220808
Change-Id: I08980aa19117e1de7a9b7896d05d07715dd9f962
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375460
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43821}
This will cause automation to not pick it up when searching for
headers.
Bug: webrtc:42226155
Change-Id: I4e93cd4eca13af32f76201df784b20a80ac9baed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375581
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43816}
There can be error log each frame when maximum playout delay sent with the frame exceed delay derived from the av-sync.
In such scenario prefer to limit the playout delay by the one attached to the received frame.
Bug: b/390043766
Change-Id: Ia57969df72f7a649e5a9280d5bb29986f5ea14b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374284
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43814}
Extend DtlsRestart test to also
test with Dtls13 (and add variants
that tests caller/callee active).
BUG=webrtc:383141571
Change-Id: Ib8b48653d4ad3cb2f5d66d6e28fc9ab54064d804
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375620
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43813}
This reverts commit a97304ca03c2aeb4267dc1bd794c50aa8bdb9a69.
Reason for revert: performance tests still rely in on global field trials to configure PC created by this test fixture
Original change's description:
> Cleanup usage of the global field trials in the PeerConnectionE2EQualityTest helper
>
> Bug: webrtc:42220378
> Change-Id: I3dc1a71c043ef506b6d592673b04e49f4a022d17
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374901
> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43803}
Bug: webrtc:42220378, webrtc:392672060
Change-Id: Ide265c1284f9d53c0b652ed5e144dfb0a532f87a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375621
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Commit-Queue: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43812}
This test was only testing codec vendor functionality.
Bug: webrtc:360058654
Change-Id: I5763e766a44f6bb1542c4281b1d6c177a52c8c74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375600
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43811}
Since Xrandr depends on Xrender, it needs to be explicitely listed before, otherwise linkers may not find the Xrender symbols.
Bug: None
Change-Id: Ifb1e82f63e1fc1645979c14b65e3beab06637cb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375428
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Florent Castelli SE <fcastelli@nvidia.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43808}
This flip default behavior for webrtc users that create packetizers without help of RtpSenderVideo class.
Bug: webrtc:42226301
Change-Id: I42fe696039334672b7d0b0ed1f87a52c3f6bf5ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374883
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43807}
The new class "CodecVendor" is intended to handle all logic dealing
with codecs. This CL is a no-behavior-change CL, later CLs will
change the logic.
Bug: webrtc:360058654
Change-Id: I44e76f0e0bd364eeb7d4506f3e01e9e00e2843a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375500
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43806}
Whenever encoding info change, this ToString() method is called for some
LS_INFO logging inside video_stream_encoder.cc. Apparently the char
buffer used for constructing this string is not large enough because I
can get WebRTC to crash in a demo page that gets and sets a lot of
parameters.
By changing to rtc::StringBuilder, we don't have to make assumptions
about how long the string can get at runtime.
Bug: None
Change-Id: I32695523282143a301c0e13e06082d55bd2796b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375520
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43805}
instead use the standard API to get the rollover counter and
determine the extended sequence number which is the basis for the packet index.
See https://github.com/cisco/libsrtp/issues/738 and
https://github.com/cisco/libsrtp/issues/721
BUG=webrtc:357776213
Change-Id: I90c5a4a538f56132158aa48db8700187fcdb47d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371960
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43802}
This is a reland of commit 9572b2fa5850da6d319b9efb5ee36290e2895f7f
that does not remove the legacy implementations yet.
Original change's description:
> srtp: spanify Protect + Unprotect
>
> Makes SrtpSession and SrtpTransport use rtc::CopyOnWriteBuffer for the Protect and Unprotect operations instead of passing around void pointers.
>
> Also updates the unit tests to use CopyOnWriteBuffer instead of char arrays with a fixed length.
>
> BUG=webrtc:357776213
> No-Iwyu: missing include is a private libsrtp header
>
> Change-Id: I02a22ceb4e183e93c4ebd8c0a9c931404e0e32f3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358442
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#43601}
No-Iwyu: missing include is a private libsrtp header
Bug: webrtc:357776213
Change-Id: I93704e27a6c48e015b775712fcd848c8c0c753e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372321
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43799}
This was already done in one place but got caught by our linter
nonetheless. For better obfuscation split "PRIVATE" into two pieces.
BUG=None
No-Iwyu: mostly unrelated changes and some require special attention
Change-Id: Iba82b603fd5c5a50c75fc7e27cafbc7237e956f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375063
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43798}
Prior to this CL, IsSameRtpCodecIgnoringLevel() only ignored level IDs
if the codec was H265, incorrectly considering, for example, different
levels of H264 Baseline as not equal.
- This CL fixes that problem by using IsSameCodecSpecific() which is
already used in other places, reducing the risk of different
comparisons using different comparison rules.
This also fixes https://crbug.com/webrtc/391340599 where
setParameters() would throw if unrecognized SDP FMTP parameters were
added to a codec as part of SDP negotiation via SDP munging.
This CL makes the following WPT tests pass:
- external/wpt/webrtc/protocol/h264-unidirectional-codec-offer.https.html
- fast/peerconnection/RTCRtpSender-setParameters.html
Bug: chromium:381407888, webrtc:391340599
Change-Id: I5991403b56c86ba97e670996c6687f6315dde304
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374043
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43797}
DTLS 1.3 encrypts more parts of the handshake so we move from
deep packet inspection to looking at the state of DTLS to
decide whether to intercept the packet.
BUG=webrtc:367395350
Change-Id: Idb1eda0437f24002f48381af5d6a167a4a153381
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374501
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43794}
This change resolves an issue that arises when there is a gap in the
sequence numbers of packets associated with a single frame.
Before this change, the H26x packet buffer could potentially assemble a
frame using only a subset of the packets in the buffer if a packet was
missing in the middle and a packet with a marker bit arrived.
To address this, the change introduces a check before assembling a
frame. This ensures that all packets belonging to a single frame are
correctly collected by iterating backward until the first packet in the
frame is identified.
Bug: webrtc:384391181
Change-Id: I4d09a3d6d569624ece204264cb32e5076ed090a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374183
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Jianlin Qiu <jianlin.qiu@intel.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43793}
This helps Java clients control the port range.
Bug: None
Change-Id: Icfe16cdfac4e08cd21346a3cb4bb65b9fb2fa0d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374841
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Youjie Zhou <youjiezhou@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43789}
This CL implements allowing sendonly codecs in setCodecPreferences(),
i.e. this spec PR: https://github.com/w3c/webrtc-pc/pull/3018. It also
makes the setCodecPreferences() ignore level IDs in the filtering
algorithm (but not in the sCP method call) as per this spec PR:
https://github.com/w3c/webrtc-pc/pull/3023.
In short, before this CL, setCodecPreferences() threw an exception if a
codec was preferred that is not present in receiver codec capabilities.
After this CL, setCodecPreferences() allows you to prefer codecs that
are *either* in the sender capabilities *or* the receiver capabilities.
- This allows you to "offer to send", i.e. prefer sendonly codecs on a
sendonly transceiver.
- The filtering on direction is handled by
RtpTransceiver::filtered_codec_preferences() which is called during
SDP offer/answer (sdp_offer_answer.cc).
Also as per spec changes, if this filtering results in not having any
codecs to offer or answer then this results in not having any codec
preferences as opposed to throwing an exception (old behavior).
- Two old peer_connection_media_unittest.cc tests are updated to
reflect the API failing less.
This CL adds both unit tests (rtp_transceiver_unittest.cc) and full
stack integration tests (peer_connection_encodings_integrationtest.cc).
It also makes us pass the following Web Platform Tests in Chrome:
https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/protocol/h265-level-id.https.html
Bug: chromium:381407888
Change-Id: I98a5ad1acccb56db0538e4d47975b8a725102c33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374520
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43788}
This is a reland of commit bcb19c00ba8ab1788ba3c08f28ee1b23e0cc77b9
Original change's description:
> Allow sending to separate payload types for each simulcast index.
>
> This change is for mixed-codec simulcast.
>
> By obtaining the payload type via RtpConfig::GetStreamConfig(),
> the correct payload type can be retrieved regardless of whether
> RtpConfig::stream_configs is initialized or not.
>
> Bug: webrtc:362277533
> Change-Id: I6b2a1ae66356b20a832565ce6729c3ce9e73a161
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364760
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43197}
Bug: webrtc:362277533
Change-Id: Ia82c3390cceb9f68315c2fd9ba5114693669af32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374780
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43787}
The old version of these returns -1 when the value is not set.
Optional is better.
Bug: webrtc:42220231
Change-Id: Ideb0f51fd8bb7b5aa490743eb3b5d95998efbd1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374483
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43786}
...to use string_view for the mid and prefer .mid() over .name for
ContentInfo.
Bug: webrtc:42233761
Change-Id: Ia9bfe1d7454759ff87295939cda6a71e53cb6b98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374663
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43781}
This adds an optional callback closure and an enum representing the error.
Bug: webrtc:390314937
Change-Id: If9a22dd6d90d5c4d94175e021511766ea49acec2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374420
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43780}
