This change adds the ability to optionally provide configuration to the
video_replay binary through JSON. This is a merge of the changes provided by
philipel on this bug: https://bugs.chromium.org/p/chromium/issues/detail?id=840536#c1
However it has been updated to pull all the json parsing into the example binary
itself instead of it being integrated into the core library. Writing test cases
out to JSON configuration will be handled in a different CL. Most likely there
will be a utility class added that takes a Config and converts it to JSON that is
decoupled from the actual implementation.
Bug: webrtc:9609
Change-Id: Icc5900063d7f704825f224240e4b3787c06ca074
Reviewed-on: https://webrtc-review.googlesource.com/95320
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24433}
Replaced by a int64_t representing time in us.
Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
Only user was the replay.cc tool, when dumping frames to a file. It is
changed to instead inject a special decoder.
Bug: None
Change-Id: I521fbba1a0ef440cff7d786f6f4c6397e33f764f
Reviewed-on: https://webrtc-review.googlesource.com/83121
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23675}
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 reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266.
Reason for revert: Regression in ramp up perf tests.
Original change's description:
> Reland "Move rtp-specific config out of EncoderSettings."
>
> This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
>
> Original change's description:
> > Move rtp-specific config out of EncoderSettings.
> >
> > In VideoSendStream::Config, move payload_name and payload_type from
> > EncoderSettings to Rtp.
> >
> > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > and should perhaps be renamed in a follow up cl. It's no longer
> > passed as an argument to VideoCodecInitializer::SetupCodec.
> >
> > The latter then needs a different way to know the codec type,
> > which is provided by a new codec_type member in VideoEncoderConfig.
> >
> > Bug: webrtc:8830
> > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22532}
>
> Bug: webrtc:8830
> Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> Reviewed-on: https://webrtc-review.googlesource.com/63721
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22595}
TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
Bug: webrtc:8830,chromium:827080
Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
Reviewed-on: https://webrtc-review.googlesource.com/65520
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22677}
This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
Original change's description:
> Move rtp-specific config out of EncoderSettings.
>
> In VideoSendStream::Config, move payload_name and payload_type from
> EncoderSettings to Rtp.
>
> EncoderSettings now contains configuration for VideoStreamEncoder only,
> and should perhaps be renamed in a follow up cl. It's no longer
> passed as an argument to VideoCodecInitializer::SetupCodec.
>
> The latter then needs a different way to know the codec type,
> which is provided by a new codec_type member in VideoEncoderConfig.
>
> Bug: webrtc:8830
> Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> Reviewed-on: https://webrtc-review.googlesource.com/62062
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22532}
Bug: webrtc:8830
Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
Reviewed-on: https://webrtc-review.googlesource.com/63721
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22595}
This reverts commit bc900cb1d1810fcf678fe41cf1e3966daa39c88c.
Reason for revert: Broke downstream projects.
Original change's description:
> Move rtp-specific config out of EncoderSettings.
>
> In VideoSendStream::Config, move payload_name and payload_type from
> EncoderSettings to Rtp.
>
> EncoderSettings now contains configuration for VideoStreamEncoder only,
> and should perhaps be renamed in a follow up cl. It's no longer
> passed as an argument to VideoCodecInitializer::SetupCodec.
>
> The latter then needs a different way to know the codec type,
> which is provided by a new codec_type member in VideoEncoderConfig.
>
> Bug: webrtc:8830
> Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> Reviewed-on: https://webrtc-review.googlesource.com/62062
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22532}
TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
Change-Id: I01f06c1fcf21eb2cd40dca7d4f268614200ee490
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/63720
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22537}
In VideoSendStream::Config, move payload_name and payload_type from
EncoderSettings to Rtp.
EncoderSettings now contains configuration for VideoStreamEncoder only,
and should perhaps be renamed in a follow up cl. It's no longer
passed as an argument to VideoCodecInitializer::SetupCodec.
The latter then needs a different way to know the codec type,
which is provided by a new codec_type member in VideoEncoderConfig.
Bug: webrtc:8830
Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
Reviewed-on: https://webrtc-review.googlesource.com/62062
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22532}
instead of pair of pointer + size.
it removes hidden memcpy in RtpPacketReceived::Parse:
RtpPacketReceived keeps a reference to a CopyOnWriteBuffer. By
passing it the same CopyOnWriteBuffer that was created by
BaseChannel, one allocation and memcpy is avoided.
Bug: None
Change-Id: I5f89f478b380fc9aece3762d3a04f228d48598f5
Reviewed-on: https://webrtc-review.googlesource.com/23761
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21143}
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}
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}
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}
