Commit graph

59 commits

Author SHA1 Message Date
Mirko Bonadei
66e7679fb8 Export symbols needed by the Chromium component build (part 8).
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).

[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md

Bug: webrtc:9419
Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27410}
2019-04-02 10:13:36 +00:00
Steve Anton
df5923da0c scale_resolution_down_by and rid are implemented
Bug: None
Change-Id: Ifccfb2f451fbcbbe9da3cd157dad66999475acce
Reviewed-on: https://webrtc-review.googlesource.com/c/125140
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26911}
2019-03-01 01:42:02 +00:00
Elad Alon
48e7065ac6 Remove default IDs for RTP extensions from rtp_parameters.h
One-byte RTP extensions may only have IDs in the range 1-14.
For higher IDs, the two-byte format must be used.
If default IDs are set for all extensions, once 15 extensions are
defined by the code, some extensions will have IDs greater than 14.
This will happen even if only one extension actually ends up being
offered, so long as it's that unfortunate RTP extension.
It's better to dynamically assign the IDs to those extensions we
actually offer. The code that assigns the IDs is currently
distributed ( WebRtcVoiceEngine::GetCapabilities() and
WebRtcVideoEngine::GetCapabilities()), and without a bigger
refactoring effort would produce some ID collisions and mismatches.
Those are already handled by MergeRtpHdrExts(), so so that
should not be a problem.

Bug: webrtc:10288
Change-Id: I087f1ed5baa9fd61fd5556f1d82f540304ec6b93
Reviewed-on: https://webrtc-review.googlesource.com/c/122480
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26876}
2019-02-27 14:33:03 +00:00
Elad Alon
ccb9b759c5 Create version 01 of Generic Frame Descriptor - with discardability flag
The discardability flag denotes whether the frame may be dropped by
the decoder with no effect on the decodability of subsequent frames.

Bug: webrtc:10214
Change-Id: I3654951d8863b50effe9670b8d1d7eb051240039
Reviewed-on: https://webrtc-review.googlesource.com/c/122241
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26763}
2019-02-20 10:31:58 +00:00
Elad Alon
157540ac05 Stop hard-coding default IDs for RTP extensions
Hard-coding default values forces IDs over 14 to be used even
when we offer less than 15 different extensions.

Note that the code relies on MergeRtpHdrExts for making sure
that extension IDs are kept consistent and non-colliding between
different streams (audio/video).

Bug: webrtc:10288
Change-Id: I3e59f7ddc8ca43cea91084a6b7f36df70fb6be4a
Reviewed-on: https://webrtc-review.googlesource.com/c/121646
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26622}
2019-02-09 01:04:35 +00:00
Johannes Kron
7ff164e6e1 Plumbing of feedback on request setting
Bug: webrtc:10263
Change-Id: I23c09e680d6381598e4172b76025ff84f33aa4de
Reviewed-on: https://webrtc-review.googlesource.com/c/121422
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26606}
2019-02-08 10:54:21 +00:00
Ilya Nikolaevskiy
9f6a0d5d21 In VideoEngine also respect requested TL number even for screenshare
Bug: chromium:927208
Change-Id: Ic20b2da246dac9185375cc42a6a2505aaff95ac6
Reviewed-on: https://webrtc-review.googlesource.com/c/121403
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26546}
2019-02-05 13:21:38 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Steve Anton
1c05765831 (3) Rename files to snake_case: move the files
Mechanically generated with this command:

tools_webrtc/do-rename.sh move all-renames.txt

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26225}
2019-01-11 17:05:20 +00:00
Renamed from api/rtpparameters.h (Browse further)