Enforces previous kProtectionKeyOnLoss as the permanent method which was
the only one used in use. This simplifies SetVideoProtection and
transition over to SetReceiverRobustnessMode.
BUG=webrtc:1596
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1244753002
Cr-Commit-Position: refs/heads/master@{#9641}
These small packets are common for H.264 where the first packet of an IDR
contains the parameter sets.
BUG=4806
Review URL: https://codereview.webrtc.org/1221943002
Cr-Commit-Position: refs/heads/master@{#9639}
-- Made plot logging from MetricRecorder and from RateCounterFilter/PacketReceiver standard to fit python and shell plotting scripts likewise.
-- RateCounterFilter is initialized with algorithm name.
-- Removed spare commas and duplicated flow ids from RateCounterFilter name.
-- Added optional plot_delay and plot_loss in MetricRecorder.
-- PacketReceiver can plot directly plot delay if there is no metric_recorder_.
-- Added comments to plot scripts.
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1253473004 .
Cr-Commit-Position: refs/heads/master@{#9636}
Permits setting RTP extensions for AudioReceiveStream without enabling
combined A/V BWE. This prevents spamming the log with "Failed to find
extension id:".
BUG=webrtc:4870
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1256803004
Cr-Commit-Position: refs/heads/master@{#9633}
Reason for revert:
I think this causes WebRtcBrowserTest.CallAndModifyStream to fail on Android. See https://code.google.com/p/webrtc/issues/detail?id=4857 for more info.
Original issue's description:
> Fixing scenario where track is rejected and later un-rejected.
>
> Added `RestartLocalTracks` and `RestartRemoteTracks` methods to
> `MediaStreamHandlerContainer` which will redo the track handlers'
> initial setup; most importantly, this will re-connect the
> renderer/capturer/etc. to a channel which was destroyed and then
> re-created.
>
> Also added `AcceptRemoteTracks` method to MediaStreamSignaling, which
> does the inverse of `RejectRemoteTracks`. Effectively this will notify
> sinks that the track is live again, after previously being set to
> `kEnded` when it was rejected.
>
> BUG=webrtc:2136
>
> Committed: https://crrev.com/be37888b6d5d269dbd5385569dba15c0d70594f2
> Cr-Commit-Position: refs/heads/master@{#9600}
TBR=pthatcher@webrtc.org,juberti@webrtc.org,deadbeef@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2136
Review URL: https://codereview.webrtc.org/1247443005
Cr-Commit-Position: refs/heads/master@{#9622}
Reason for revert:
Breaks Chromium FYI content_browsertest on all platforms. The testcase that fails is WebRtcAecDumpBrowserTest.CallWithAecDump.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/19388
Sample output:
[ RUN ] WebRtcAecDumpBrowserTest.CallWithAecDump
Xlib: extension "RANDR" missing on display ":9".
[4:14:0722/211548:1282124453:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: ISAC/48000/1 (105)
[4:14:0722/211548:1282124593:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMU/8000/2 (110)
[4:14:0722/211548:1282124700:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMA/8000/2 (118)
[4:14:0722/211548:1282124815:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: G722/8000/2 (119)
[19745:19745:0722/211548:1282133667:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
[19745:19745:0722/211548:1282136892:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
../../content/test/webrtc_content_browsertest_base.cc:62: Failure
Value of: ExecuteScriptAndExtractString( shell()->web_contents(), javascript, &result)
Actual: false
Expected: true
Failed to execute javascript call({video: true, audio: true});.
From javascript: (nothing)
When executing 'call({video: true, audio: true});'
../../content/test/webrtc_content_browsertest_base.cc:75: Failure
Failed
../../content/browser/media/webrtc_aecdump_browsertest.cc:26: Failure
Expected: (base::kNullProcessId) != (*id), actual: 0 vs 0
../../content/browser/media/webrtc_aecdump_browsertest.cc:95: Failure
Value of: GetRenderProcessHostId(&render_process_id)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:99: Failure
Value of: base::PathExists(dump_file)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:101: Failure
Value of: base::GetFileSize(dump_file, &file_size)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:102: Failure
Expected: (file_size) > (0), actual: 0 vs 0
[ FAILED ] WebRtcAecDumpBrowserTest.CallWithAecDump, where TypeParam = and GetParam() = (361 ms)
Original issue's description:
> Allow more than 2 input channels in AudioProcessing.
>
> The number of output channels is constrained to be equal to either 1 or the
> number of input channels.
>
> R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org
>
> Committed: c204754b7aTBR=andrew@webrtc.org,aluebs@webrtc.org,ajm@chromium.org,pbos@chromium.org,pbos@webrtc.org,mgraczyk@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1253573005
Cr-Commit-Position: refs/heads/master@{#9621}
This allows us to configure create-time parameters for AudioProcessing
in a voice engine app and avoid the onerous SetExtraOptions.
voe_cmd_test would require significant refactoring to do the same.
Minor cleanups:
- Use agc_manager_direct. This should allow us to remove agc_manager.
- Use CHECKs rather than ASSERTs.
Review URL: https://codereview.webrtc.org/1247033006
Cr-Commit-Position: refs/heads/master@{#9618}
These are mostly trivial changes and are separated out just to reduce the
diff on that change to the minimum possible.
Note explanatory comments on patch set 1.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1235643003
Cr-Commit-Position: refs/heads/master@{#9617}
Android has not yet finalized its libc++ build. Allow compilation with
stlport by removing several C++11 library usages.
BUG=427718,487341,webrtc:4866
R=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1250663007 .
Patch from Jared Duke <jdduke@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#9616}
Supports:
- bwe_plot.sh
---- Histogram plots
---- Baseline histograms
---- Vertical error lines
---- Dashed horizontal lines
---- Different colors for different algorithms
---- Legends read as input
---- Extra horizontal space in case there are TCP flows plotted
---- Multiple windows
---- Auto vertical scale, except for latency plots which are kept constant for all plots
- plot_dynamics.py
---- Dynamic plots
---- Different colors for different flows and algorithms
---- Dashed line plot for available capacity
---- Throughput, latency and loss on separated boxes
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1237903003 .
Cr-Commit-Position: refs/heads/master@{#9614}
Using explicit atomic operations permits TSan to understand them and
prevents false positives.
Downgrading the atomic Load to acquire semantics. This reduces the
number of memory barriers inserted from two down to one at most.
Also renaming Load/Store to AcquireLoad/ReleaseStore.
BUG=chromium:512382
R=dvyukov@chromium.org, glider@chromium.orgTBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1246073002
Cr-Commit-Position: refs/heads/master@{#9613}
It's built as a part of the chromium build(?), but causes problems with XCode 7's new bitcode intermediate format. It should really never been included, but it never caused link failures before.
BUG=
Review URL: https://codereview.webrtc.org/1236383002
Cr-Commit-Position: refs/heads/master@{#9606}
In #9243 we added some thread_checker. But it shouldn't be added into PlayoutDevices() and RecordingDevices(), since these two will be invoked from RecThread and PlayoutThread too, other than the main thread.
BUG=webrtc:4852
TEST=voe_cmd_test
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1249573002 .
Cr-Commit-Position: refs/heads/master@{#9605}
Added `RestartLocalTracks` and `RestartRemoteTracks` methods to
`MediaStreamHandlerContainer` which will redo the track handlers'
initial setup; most importantly, this will re-connect the
renderer/capturer/etc. to a channel which was destroyed and then
re-created.
Also added `AcceptRemoteTracks` method to MediaStreamSignaling, which
does the inverse of `RejectRemoteTracks`. Effectively this will notify
sinks that the track is live again, after previously being set to
`kEnded` when it was rejected.
BUG=webrtc:2136
Review URL: https://codereview.webrtc.org/1231613002
Cr-Commit-Position: refs/heads/master@{#9600}
Essentially we are carrying over the capture timestamp to the encoded frame sent out, so the frame lengths will contain no noise.
Review URL: https://codereview.webrtc.org/1225153002
Cr-Commit-Position: refs/heads/master@{#9597}
This was already working in most cases, but not for some corner cases:
* If the PORTALLOCATOR_ENABLE_SHARED_SOCKET flag is not set
* If both a STUN server and TURN server are configured
I added unit tests for these cases, and centralized the code that gets
STUN server addresses in order to fix these and any related issues.
BUG=webrtc:4215
Review URL: https://codereview.webrtc.org/1215713003
Cr-Commit-Position: refs/heads/master@{#9596}
"field_trial::FindFullName" can return "std::string()" which should not
be referenced by the caller.
Review URL: https://codereview.webrtc.org/1238943003
Cr-Commit-Position: refs/heads/master@{#9594}
