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5 commits

Author SHA1 Message Date
Jonas Olsson
a4d873786f Format almost everything. 
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
 2019-07-08 13:45:15 +00:00
Johannes Kron
b5d918324c Add RTP timestamp to contributing sources 
RTP timestamp was recently added to contributing sources in the WebRTC
specification. This CL implements that change in WebRTC.

Bug: webrtc:10650
Change-Id: Ic0ccfbea7049a5b66063fa6cf60d01d5bd713132
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137515
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28020}
 2019-05-22 08:53:08 +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
Jonas Oreland
967f7d5497 Add audio level to CSRC class 
This patch adds (optional) csrc to ContributingSources.
This will be used if using virtual audio ssrc, since
the audio level is otherwise unaccessible in that configuration.

BUG=webrtc:3333

Change-Id: Ied263b8f0850553cd637fd6bead373ed4252fd1e
Reviewed-on: https://webrtc-review.googlesource.com/c/109281
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25516}
 2018-11-06 12:10:05 +00:00
Niels Möller
af17595879 Refactor RtpReceiverImpl, extracting CSRC book-keeping to its own class 
Bug: webrtc:7135
Change-Id: I7ce9afe575241542e4e3f7e2e8459ee3257eec76
Reviewed-on: https://webrtc-review.googlesource.com/93466
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24271}
 2018-08-13 11:59:08 +00:00