| 0e7e259ebd69993bb5670a991f43aa1b06c9bf9e |
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13-Nov-2015 |
mflodman <mflodman@webrtc.org> |
Move BitrateAllocator from BitrateController logic to Call.
This is a step on the way to have variable bitrate for audio and is
intended to be as much of a no-op as possible.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1441673002
Cr-Commit-Position: refs/heads/master@{#10630}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
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| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
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04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
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| bbe876f0d30ec806c7c4a12629eb1f19ab45fb86 |
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23-Oct-2015 |
stefan <stefan@webrtc.org> |
Set send times in send time history via OnSentPacket.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1419503004
Cr-Commit-Position: refs/heads/master@{#10384}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
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| c1aeaf0dc37d96f31c92f893f4e30e7a5f7cc2b7 |
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15-Oct-2015 |
stefan <stefan@webrtc.org> |
Wire up packet_id / send time callbacks to webrtc via libjingle.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1363573002
Cr-Commit-Position: refs/heads/master@{#10289}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
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| 6b8d3551681f40b880507cecc88f478a12383cc7 |
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24-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Reland "Wire up send-side bandwidth estimation."
Revert was patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/
The culprit was RTC_DCHECK(poller_thread_->Start()); in rampup_test.cc
BUG=webrtc:4173
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1362303002 .
Cr-Commit-Position: refs/heads/master@{#10052}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
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| c9bbeb03542cffc14b7d306e5f88b6c0e593864d |
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23-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ )
Reason for revert:
Breaking some Android bots.
https://chromegw.corp.google.com/i/client.webrtc/builders/Android32%20Tests%20%28L%20Nexus5%29
Original issue's description:
> Wire up send-side bandwidth estimation.
>
> BUG=webrtc:4173
>
> Committed: https://crrev.com/ef165eefc79cf28bb67779afe303cc2365885547
> Cr-Commit-Position: refs/heads/master@{#10012}
TBR=stefan@webrtc.org, kjellander@webrtc.org
NOPRESUBMIT=false
NOTREECHECKS=false
NOTRY=false
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1362923002 .
Cr-Commit-Position: refs/heads/master@{#10029}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
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| ef165eefc79cf28bb67779afe303cc2365885547 |
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22-Sep-2015 |
sprang <sprang@webrtc.org> |
Wire up send-side bandwidth estimation.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1338203003
Cr-Commit-Position: refs/heads/master@{#10012}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
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| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
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17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
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| 5e023eb337eed9746ecea7fc6fbb0fca386f1961 |
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14-Sep-2015 |
sprang <sprang@webrtc.org> |
Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor
When using send-side bandwidth estimation, the inter-packet delay is
reported back to the sender using RTCP TransportFeedback messages.
Theis data needs to be translated into a format which the bandwidth
estimator (now instantiated on the send side) can use, including looking
up the local absolute send time from the send time history.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1329083005
Cr-Commit-Position: refs/heads/master@{#9929}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
|