Commit Graph

67 Commits

Author SHA1 Message Date
6fed924857 Delete RTPPayloadRegistry::SetIncomingPayloadType.
It only affects the write-only member |incoming_payload_type_|.

Bug: webrtc:8995
Change-Id: I0cf86a6d0603c809367105cee31eb1b8b2802d32
Reviewed-on: https://webrtc-review.googlesource.com/61040
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22382}
2018-03-12 11:03:59 +00:00
2e1d784956 Delete the VideoCodec::plName string.
It holds the same information as codecType, but in different format.

Bug: webrtc:8830
Change-Id: Ia83e2dff4fd9a5ddb489501b7a1fe80759fa4218
Reviewed-on: https://webrtc-review.googlesource.com/56100
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22307}
2018-03-06 11:17:41 +00:00
0a9f6de446 Removed VCMTiming from RtpVideoStreamReceiver.
Bug: webrtc:8909
Change-Id: Ib42e4cc4c2252f04ea5f7d440352546d36d2899f
Reviewed-on: https://webrtc-review.googlesource.com/58740
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22224}
2018-02-28 11:31:51 +00:00
e7c891f953 Renamed FrameObject to EncodedFrame.
The plan is to:
 1. Move FrameObject to api/video.
 2. Rename FrameObject to EncodedFrame.
 3. Move EncodedFrame out of the video_coding namespace.

This is the 2nd CL.

Bug: webrtc:8909
Change-Id: I5e76a0a3b306156b8bc1de67834b4adf14bebef9
Reviewed-on: https://webrtc-review.googlesource.com/56182
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22158}
2018-02-22 16:12:48 +00:00
d397a0d46e Add dropped frames metric on the receive side
Reported to UMA and logged for at the end of the call.

Bug: webrtc:8355
Change-Id: I4ef31bf9e55feaba9cf28be5cb4fcfae929c7179
Reviewed-on: https://webrtc-review.googlesource.com/53760
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22132}
2018-02-21 15:34:25 +00:00
80ba333fc5 Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix
Bug: chromium:805946
Change-Id: Ibb5dce9af27d0e48c9aee6b0a860b6f62b3c76a0
Reviewed-on: https://webrtc-review.googlesource.com/46145
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21889}
2018-02-05 11:24:59 +00:00
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
7172ea13c0 Don't use old RTCP SR reports for remote clock estimation
At the beginning of the call, when rtt is not yet estimated, SR packets
are not used for estimation. Yet, it may happen that on some non-SR
RTCP packet RTT would become available. At that time an old SR will be
used for remote clock estimation. This will lead to remote clock offset
to the past too much.


Bug: webrtc:8468
Change-Id: I1bdbd56a7bab1c28e73987e5fb307f8e7382b045
Reviewed-on: https://webrtc-review.googlesource.com/16840
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20528}
2017-11-01 12:34:26 +00:00
22ec952829 Delete in_order argument to RtpReceiver::IncomingRtpPacket
Bug: webrtc:7135
Change-Id: I35fbc76a5ca8d50caff918bbfd2cb13dce4cbd21
Reviewed-on: https://webrtc-review.googlesource.com/4141
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20154}
2017-10-05 07:19:20 +00:00
c62f6c7121 RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs
This is needed in the general case, now that we aim to support codecs
other than those built-in to WebRTC.

BUG=webrtc:8159

Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706
Reviewed-on: https://webrtc-review.googlesource.com/5380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20136}
2017-10-04 11:30:14 +00:00
c856dc2b6b Convert PayloadUnion from a union to a class, step 2
Stop using PayloadUnion's public member variables, since a future CL
will make them private.

BUG=webrtc:8159

Change-Id: Ia3dada56be7ef00ed80f3733209b18c178a36561
Reviewed-on: https://webrtc-review.googlesource.com/4380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20027}
2017-09-28 23:23:07 +00:00
3b3622fafc Delete member VideoReceiveStream::Config::Rtp::ulpfec.
Replaced with scalars ulpfec_payload_type and red_payload_type.

In particular, ulpfec.red_rtx_payload_type, which duplicated info in
rtx_associated_payload_types, is deleted. This is a followup to cl
https://codereview.webrtc.org/3012963002.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/3019453002
Cr-Commit-Position: refs/heads/master@{#19965}
2017-09-26 09:49:21 +00:00
b0573bca16 Reorganize config of RTP header extensions for video receive streams.
Bug: webrtc:6847
Change-Id: Iae2386e55520601883379fc7802a5c5246be935e
Reviewed-on: https://webrtc-review.googlesource.com/2001
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19943}
2017-09-25 11:51:20 +00:00
73b60b82ee Remove the redundant method GetPayloadSpecifics
It's in the way of a refactoring.

Also change PayloadTypeToPayload---the method all callers can use instead---to return Optional<Payload> instead of const Payload* (for thread safety reasons: an object that protects itself with a mutex shouldn't be handing out pointers to parts of itself). 

BUG=webrtc:8159

Change-Id: I7ef0d545077ffdea016b309f2165e3c4955a2928
Reviewed-on: https://webrtc-review.googlesource.com/2360
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19917}
2017-09-21 20:19:55 +00:00
7120742701 Adding NOLINT for typedefs.h and common_types.h
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
2017-09-15 13:03:51 +00:00
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
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}
2017-09-15 05:02:56 +00:00
bb547203bf Moving src/webrtc into src/.
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