f6975f46131981f83e0c88d276dee6b6c5753180 |
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28-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] Lint errors cleaned from rtp_utility R=åsapersson BUG=webrtc:5277 Review URL: https://codereview.webrtc.org/1539423003 Cr-Commit-Position: refs/heads/master@{#11131}
/external/webrtc/webrtc/test/layer_filtering_transport.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/test/layer_filtering_transport.cc
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ce4aef16eec96862199e89b6d3ffe059558ac2c0 |
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02-Nov-2015 |
sprang <sprang@webrtc.org> |
Adding support for simulcast and spatial layers into VideoQualityTest This is a re-land of https://codereview.webrtc.org/1353263005/ which was reverted because of perf-regressions. Changes since that CL: * Change LayerFilteringTransport to send a padding packet instead of dropping it for data that should be filtered out. This prevents confusion due to changed sequence numbers. * Changed timing of stats poller thread in VideoAnalyzer. Startup was racy wrt initializion of send_stream_. * Minor formatting issues. PERF NOTE: This change will affect some performance numbers slightly. In particular, {encode_frame_rate, encode_time_ms, encode_usage_percent, media_bitrate_bps} will change due to timing of the measurements. BUG= R=pbos@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1412233003 Cr-Commit-Position: refs/heads/master@{#10483}
/external/webrtc/webrtc/test/layer_filtering_transport.cc
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f116bd0d7a3cdad20bb638d5a87427bd920c8904 |
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27-Oct-2015 |
stefan <stefan@webrtc.org> |
Call OnSentPacket for all packets sent in the test framework. Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method. BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1419193002 Cr-Commit-Position: refs/heads/master@{#10430}
/external/webrtc/webrtc/test/layer_filtering_transport.cc
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7a975f75e7fa7a9335411ef22b6687f78f7b297f |
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12-Oct-2015 |
sprang <sprang@webrtc.org> |
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ ) Reason for revert: Temporarily reverting as this causes some issues with perf tests. Especially tests with packet loss no longer works. Original issue's description: > Adding support for simulcast and spatial layers into VideoQualityTest > > The CL includes several changes: > - Adding flags describing the streams and spatial layers. > - Reorganizing the order of the flags, to make them easier to maintain. > - Adding a member .params_ to VideoQualityAnalyzer. > (instead of passing it to every member function manually) > - Updating VideoAnalyzer to support simulcast. > (select appropriate ssrc and fix timestamps which are sometimes increased by 1) > - VP9EncoderImpl already had code for automatic calculation of bitrate for each layer. > Changing to first read bitrates and resolution ratios from the flags, if specified. > If not specified, reverting to the old code are setting the values automatically. > - Changing the parameters in LayerFilteringTransport, replacing > xx_discard_thresholds with selected_xx, to make it easier to use for the end user. > > Committed: https://crrev.com/87f83a9a27d657731ccb54025bc04ccad0da136e > Cr-Commit-Position: refs/heads/master@{#10215} TBR=pbos@webrtc.org,mflodman@webrtc.org,ivica@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1397363002 Cr-Commit-Position: refs/heads/master@{#10252}
/external/webrtc/webrtc/test/layer_filtering_transport.cc
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87f83a9a27d657731ccb54025bc04ccad0da136e |
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08-Oct-2015 |
ivica <ivica@webrtc.org> |
Adding support for simulcast and spatial layers into VideoQualityTest The CL includes several changes: - Adding flags describing the streams and spatial layers. - Reorganizing the order of the flags, to make them easier to maintain. - Adding a member .params_ to VideoQualityAnalyzer. (instead of passing it to every member function manually) - Updating VideoAnalyzer to support simulcast. (select appropriate ssrc and fix timestamps which are sometimes increased by 1) - VP9EncoderImpl already had code for automatic calculation of bitrate for each layer. Changing to first read bitrates and resolution ratios from the flags, if specified. If not specified, reverting to the old code are setting the values automatically. - Changing the parameters in LayerFilteringTransport, replacing xx_discard_thresholds with selected_xx, to make it easier to use for the end user. Review URL: https://codereview.webrtc.org/1353263005 Cr-Commit-Position: refs/heads/master@{#10215}
/external/webrtc/webrtc/test/layer_filtering_transport.cc
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e78e2c714bdbecb910526746d9e3678a245a8f8b |
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08-Oct-2015 |
ivica <ivica@webrtc.org> |
Using different sequence numbers for different SSRCs This seems to solve the unexpected behavior when selecting lower layers. Also, this replaces https://codereview.webrtc.org/1327153002/ Review URL: https://codereview.webrtc.org/1350383004 Cr-Commit-Position: refs/heads/master@{#10206}
/external/webrtc/webrtc/test/layer_filtering_transport.cc
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1d8a506405734d0cef9653704b036ca4f1388960 |
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02-Oct-2015 |
stefan <stefan@webrtc.org> |
Add a PacketOptions struct to webrtc::Transport. This allows us to pass packet meta data, such as transport sequence number, to libjingle and further down to the socket implementation. A similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h. BUG=4173 Review URL: https://codereview.webrtc.org/1376673004 Cr-Commit-Position: refs/heads/master@{#10144}
/external/webrtc/webrtc/test/layer_filtering_transport.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/test/layer_filtering_transport.cc
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7f6a6fc0b23795cd4f0aacbf707618c1f3d0a878 |
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08-Sep-2015 |
ivica <ivica@webrtc.org> |
Enabling spatial layers in VP9Impl. Filter layers in the loopback test. Handling the case when encoder drops only the higher layer. Added options to screenshare loopback test to discard high temporal or spatial layers (to view the lower layers). Review URL: https://codereview.webrtc.org/1287643002 Cr-Commit-Position: refs/heads/master@{#9883}
/external/webrtc/webrtc/test/layer_filtering_transport.cc
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