This will later allow simulcast to be set up without any SDP
manipulation. Currently limited to only one layer as the SDP
generated is not spec compliant and more work is required
to support simulcast.
Initial encoding parameters are deferred and applied when the ssrc
is set on the sender. This allows parameters to be changed before
negotiation is completed.
Bug: webrtc:7600
Change-Id: I0a31cd1c2bfc72ebb61ce0fa4fa69d87e3d8b74d
Reviewed-on: https://webrtc-review.googlesource.com/95488
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24917}
This change injects the FrameEncryptorInterface and the FrameDecryptorInterface
into the RtpSenderInterface and RtpReceiverInterface respectively. This is the
second stage of the injection. In a follow up CL non owning pointers to these
values will be passed down into the media channel.
This change also updates the corresponding mock files.
Bug: webrtc:9681
Change-Id: I964084fc270e10af9d1127979e713493e6fbba7d
Reviewed-on: https://webrtc-review.googlesource.com/96625
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24489}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I5475e574353c772910181495fdb3400b5f0e7399
Reviewed-on: https://webrtc-review.googlesource.com/87240
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24040}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'api'
Then undo changes to optional target itself and optional_unittests
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I44093da213369d6a502e33792c694f620f53b779
Reviewed-on: https://webrtc-review.googlesource.com/84621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23707}
Target bitrate is set to 0.75 of the max bitrate.
Bug: webrtc:9341, webrtc:8655
Change-Id: I9a8c8bb95bb1532d45f05578832418464452340e
Reviewed-on: https://webrtc-review.googlesource.com/79821
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23676}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a reland of 5faf36ef3c
The issue in Chrome has been fixed and this should be safe to reland.
TBR=deadbeef
Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
>
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
>
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}
Bug: webrtc:7580
Change-Id: Iabd41fb21afdf452c039d5513824ae334f8d1d3f
Reviewed-on: https://webrtc-review.googlesource.com/76980
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23247}
This reverts commit 5faf36ef3c.
Reason for revert: fast/peerconnection/RTCRtpSender-setParameters.html
failing in webrtc roll, probably this CL? https://chromium-review.googlesource.com/c/chromium/src/+/1045889.
Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
>
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
>
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,orphis@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:7580
Change-Id: I86da108227f8fc8d235bb2e9559377c800595b8c
Reviewed-on: https://webrtc-review.googlesource.com/74740
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23134}
The transaction_id field should be refreshed for every getParameters()
call and checked at each setParameters() call.
This also checks that getParameters() was ever called to return a proper
error code.
Bug: webrtc:7580
Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
Reviewed-on: https://webrtc-review.googlesource.com/70820
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23120}
This is a reland of f351c3408a
Reland history:
The original CL broke tests in chromium which were manually tested in
the first reland. Another small fix was added to the reland to fix a
downstream bug, which caused separate tests to fail in chromium.
These were not caught because the chromium trybot was down. These
are temporarily disabled in chrome to allow this change to roll in.
Original change's description:
> Reland "Adds support for multiple or no media stream ids."
>
> This is a reland of 1550292efe
>
> Original change's description:
> > Adds support for multiple or no media stream ids.
> >
> > With Unified Plan SDP semantics, this adds support for specifying
> > either no media stream ids or multiple media stream ids for a
> > transceiver/sender/receiver. This includes serializing/deserializing
> > SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> > <appdata>" line to indicate the no stream case. Note that this does
> > not synchronize between multiple streams, this is still just supported
> > based upon the first media stream id.
> >
> > Bug: webrtc:7932, webrtc:7933
> > Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> > Reviewed-on: https://webrtc-review.googlesource.com/61341
> > Commit-Queue: Seth Hampson <shampson@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22611}
>
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed
> Reviewed-on: https://webrtc-review.googlesource.com/65560
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22687}
TBR=deadbeef@webrtc.org
Bug: webrtc:7932, webrtc:7933
Change-Id: Ideb30219b2f952dd51428cd4e8bd43ef49df5b17
Reviewed-on: https://webrtc-review.googlesource.com/66280
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22699}
This reverts commit f351c3408a.
Reason for revert: Breaks chromium import
https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_rel_ng/58012
Failin tests:
WebRtcRtpBrowserTest.TrackAddedToSecondStream
WebRtcRtpBrowserTest.TrackSwitchingStream
Original change's description:
> Reland "Adds support for multiple or no media stream ids."
>
> This is a reland of 1550292efe
>
> Original change's description:
> > Adds support for multiple or no media stream ids.
> >
> > With Unified Plan SDP semantics, this adds support for specifying
> > either no media stream ids or multiple media stream ids for a
> > transceiver/sender/receiver. This includes serializing/deserializing
> > SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> > <appdata>" line to indicate the no stream case. Note that this does
> > not synchronize between multiple streams, this is still just supported
> > based upon the first media stream id.
> >
> > Bug: webrtc:7932, webrtc:7933
> > Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> > Reviewed-on: https://webrtc-review.googlesource.com/61341
> > Commit-Queue: Seth Hampson <shampson@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22611}
>
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed
> Reviewed-on: https://webrtc-review.googlesource.com/65560
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22687}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,shampson@webrtc.org
Change-Id: I1835419f963762bc308a91d81c423d8e7bf65026
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7932, webrtc:7933
Reviewed-on: https://webrtc-review.googlesource.com/65700
Reviewed-by: Tomas Gunnarsson <tommi@chromium.org>
Commit-Queue: Tomas Gunnarsson <tommi@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22690}
This is a reland of 1550292efe
Original change's description:
> Adds support for multiple or no media stream ids.
>
> With Unified Plan SDP semantics, this adds support for specifying
> either no media stream ids or multiple media stream ids for a
> transceiver/sender/receiver. This includes serializing/deserializing
> SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> <appdata>" line to indicate the no stream case. Note that this does
> not synchronize between multiple streams, this is still just supported
> based upon the first media stream id.
>
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> Reviewed-on: https://webrtc-review.googlesource.com/61341
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22611}
Bug: webrtc:7932, webrtc:7933
Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed
Reviewed-on: https://webrtc-review.googlesource.com/65560
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22687}
This reverts commit 1550292efe.
Reason for revert:
webkit_layout_tests:fast/peerconnection/RTCPeerConnection-sdpSemantics.html is broken, see below. WebRTC roll isn't going through because of it. This CL looks the most suspicious within the 5 in the range.
https://chromium-review.googlesource.com/c/chromium/src/+/981899https://webrtc.googlesource.com/src.git/+log/bb50ce5bb6d5..27f3bf512827https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_rel_ng/54616
Original change's description:
> Adds support for multiple or no media stream ids.
>
> With Unified Plan SDP semantics, this adds support for specifying
> either no media stream ids or multiple media stream ids for a
> transceiver/sender/receiver. This includes serializing/deserializing
> SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> <appdata>" line to indicate the no stream case. Note that this does
> not synchronize between multiple streams, this is still just supported
> based upon the first media stream id.
>
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> Reviewed-on: https://webrtc-review.googlesource.com/61341
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22611}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,shampson@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:7932, webrtc:7933
Change-Id: I1d4e4332b518ec32b305c8af07679650059d02bb
Reviewed-on: https://webrtc-review.googlesource.com/65000
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22634}
With Unified Plan SDP semantics, this adds support for specifying
either no media stream ids or multiple media stream ids for a
transceiver/sender/receiver. This includes serializing/deserializing
SDPs with multiple a=msid lines in a m section, or an "a=msid:-
<appdata>" line to indicate the no stream case. Note that this does
not synchronize between multiple streams, this is still just supported
based upon the first media stream id.
Bug: webrtc:7932, webrtc:7933
Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
Reviewed-on: https://webrtc-review.googlesource.com/61341
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22611}
Original change's description:
> Update RTCStatsCollector to work with RtpTransceivers
>
> Bug: webrtc:8764
> Change-Id: I8b442345869eb6d8b65fd12241ed7cb6e7d7ce3d
> Reviewed-on: https://webrtc-review.googlesource.com/49580
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22026}
Bug: webrtc:8764
Change-Id: I6a682824febf3f4f41397fc1a8dd7396c4ffa8e3
Reviewed-on: https://webrtc-review.googlesource.com/54160
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22064}
Bug: webrtc:8764
Change-Id: I8b442345869eb6d8b65fd12241ed7cb6e7d7ce3d
Reviewed-on: https://webrtc-review.googlesource.com/49580
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22026}
This is a reland of e357a4dd4e
Original change's description:
> Move stats ID generation from SSRC to local ID
>
> This generates stats IDs for Track stats (which
> represents stats on the attachment of a track to
> a PeerConnection) from being SSRC-based to being
> based on an ID that is allocated when connecting the
> track to the PC.
>
> This is a prerequisite to generating stats before
> the PeerConnection is connected.
>
> Bug: webrtc:8673
> Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4
> Reviewed-on: https://webrtc-review.googlesource.com/38360
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21582}
TBR=solenberg@webrtc.org
Bug: webrtc:8673
Change-Id: I610302efc5393919569b77e3b59aa3384a9b88a5
Reviewed-on: https://webrtc-review.googlesource.com/38842
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21589}
This reverts commit e357a4dd4e.
Reason for revert: Looks like it's breaking some downstream projects.
Original change's description:
> Move stats ID generation from SSRC to local ID
>
> This generates stats IDs for Track stats (which
> represents stats on the attachment of a track to
> a PeerConnection) from being SSRC-based to being
> based on an ID that is allocated when connecting the
> track to the PC.
>
> This is a prerequisite to generating stats before
> the PeerConnection is connected.
>
> Bug: webrtc:8673
> Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4
> Reviewed-on: https://webrtc-review.googlesource.com/38360
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21582}
TBR=solenberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org
Change-Id: I621c10236c02be01d82f4660168f0323b85e24af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8673
Reviewed-on: https://webrtc-review.googlesource.com/38681
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21586}
This generates stats IDs for Track stats (which
represents stats on the attachment of a track to
a PeerConnection) from being SSRC-based to being
based on an ID that is allocated when connecting the
track to the PC.
This is a prerequisite to generating stats before
the PeerConnection is connected.
Bug: webrtc:8673
Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4
Reviewed-on: https://webrtc-review.googlesource.com/38360
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21582}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/api/rtpsenderinterface.h (Browse further)
