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}
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}
"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}
This CL improves the memory footprint a bit by using string references
instead of creating a copy.
Review URL: https://codereview.webrtc.org/1241973002
Cr-Commit-Position: refs/heads/master@{#9592}
BUG=webrtc:4690
Defined classes Stream, SendStream and ReceiveStream. Inherited existing stream classes from either SendStream or ReceiveStream.
This is a step towards having a Transport associated with streams instead of a Call. It will allow a lot of code in the Call to be media type agnostic.
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1226123005 .
Cr-Commit-Position: refs/heads/master@{#9591}
Tested that this doesn't break compatibility with Firefox or older
versions of Chrome, no matter which side generates the initial offer.
BUG=webrtc:2796
Review URL: https://codereview.webrtc.org/1219333002
Cr-Commit-Position: refs/heads/master@{#9589}
2. provide an implementation for SetIceConnectionReceivingTimeout so that Chrome does not complain.
BUG=
Review URL: https://codereview.webrtc.org/1227843006
Cr-Commit-Position: refs/heads/master@{#9574}
Allows removing MediaRecorder which isn't in use apart from channel
unittests, along with it unittests for MediaRecorder that are flaky when
run in parallel can also go.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1219663008
Cr-Commit-Position: refs/heads/master@{#9558}
This fixes a bug in the WebRtcSessionDescriptionFactory where messages would be dropped or worse yet processed after the factory was deleted.
BUG=chromium:507307
Review URL: https://codereview.webrtc.org/1231823002
Cr-Commit-Position: refs/heads/master@{#9557}
We have two histograms today that trigger on large jumps in either platform reported stream delays (WebRTC.Audio.PlatformReportedStreamDelayJump) or the system delay in the AEC (WebRTC.Audio.AecSystemDelayJump). The latter is the internal buffer size in the AEC.
The sizes of such jumps are of relevance since it can harm the AEC and even put it in a complete failure state. It is hard, not to say impossible, to tell how frequent it is.
Therefore, two complementary histograms are added; number of jumps in each metric.
This way we get a quick way to determine how often a jump occurs in general and also how frequent it is within a call.
This is solved by adding a counter for each metric.
The counter is activated either upon an event trigger or if we know for sure when the AEC is running.
Unfortunately, we can't rely on the destructor at the end of a call so we add a public API for the user to take on the action of calling it at the end of a call.
Tested locally by building ToT chromium including changes and three triggered jumps (200, 50 and 60 ms).
The stats picked up the 60 and 200 ms jumps as expected.
BUG=488124
R=asapersson@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1229443003.
Cr-Commit-Position: refs/heads/master@{#9544}
This CL adds an API to the metrics observer interface to report negotiated
ciphers for WebRTC sessions. This can be used from Chromium for UMA metrics
later to get an idea which cipher suites are used by clients (e.g. compare
the use of DTLS 1.0 / 1.2).
BUG=428343
Review URL: https://codereview.webrtc.org/1156143005
Cr-Commit-Position: refs/heads/master@{#9537}
We use this Config struct for enabling/disabling the delay agnostic
AEC. This change renames it to DelayAgnostic for readability reasons.
NOTE: The logic is reversed in this CL. The old ReportedDelay config
turned DA-AEC off, while the new DelayAgnostic turns it on.
The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, ReportedDelay is disabled or DelayAgnostic is enabled, DA-AEC
is engaged in APM.
BUG=webrtc:4651
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1211053006
Cr-Commit-Position: refs/heads/master@{#9531}
Normally the RTP data channel is capped at 30kbps, but by mangling the
SDP string, one could get around this limitation. With this fix,
SdpDeserialize will return an error if it detects this condition.
BUG=280726
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1196403004.
Cr-Commit-Position: refs/heads/master@{#9499}
This CL should not change any visible behaviour. It does the following:
* Extract GLES rendering into separate class GlRectDrawer. This class is also needed for future video encode with OES texture input.
* Clean up current ScalingType -> display size calculation and introduce new SCALE_ASPECT_BALANCED (b/21735609) and remove unused SCALE_FILL.
* Replace current mirror/rotation index juggling with android.opengl.Matrix operations instead.
Review URL: https://codereview.webrtc.org/1191243005
Cr-Commit-Position: refs/heads/master@{#9496}
This change includes several improvements:
* VP8 configured with new rate control
* Detection of frame dropping, with qp bump for next frame
* Increased target and TL0 bitrates
* Reworked rate control (TL allocation) in screenshare_layers
A note on performance: PSNR and SSIM is expected to get slightly worse with this cl. Frame drops and delays should however improve.
BUG=4171
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1193513006.
Cr-Commit-Position: refs/heads/master@{#9495}
With this we can write stuff like
assertThat(result.mandatory,
hasItem(new KeyValuePair("minWidth", "1280")));
The above will currently fail because the object falls back to ==.
BUG=None
Review URL: https://codereview.webrtc.org/1193883006
Cr-Commit-Position: refs/heads/master@{#9494}
This is a follow-up to r9401, where the configuration DelayCorrection
was replaced by ExtendedFilter.
This change also removes the media constraint
kExperimentalEchoCancellation which was replaced by
kExtendedFilterEchoCancellation in the same CL.
Both settings that are now being removed were kept in the code to avoid
API breakages. In https://codereview.chromium.org/1167343004,
depending code has been updated to avoid breakages.
BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1181413004.
Cr-Commit-Position: refs/heads/master@{#9444}
Removed no longer used test_isolation_outdir variable as in
https://codereview.chromium.org/1176463003
The move of a DEPS in https://codereview.chromium.org/1155743013
is causing problems on some trybots. It shouldn't affect developers.
Relevant changes:
* src/third_party/android_tools: a3afc68..ed3dde6
* src/third_party/icu: 9939a5d..a05f412
* src/third_party/libjpeg_turbo: 8ee9bdd..f4631b6
* src/third_party/libyuv: 632c50f..632c50f
Details: e937e5f..c2239a8/DEPS
Clang version was not updated in this roll.
BUG=
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1182043002.
Cr-Commit-Position: refs/heads/master@{#9435}
This constraint will be equal to kEchoCancellation until we've updated Chromium to use kGoogEchoCancellation where that constraint is needed. Once that's done, I'll change kEchoCancellation to be 'echoCancellation'.
BUG=webrtc:4747
R=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1179233003.
Cr-Commit-Position: refs/heads/master@{#9433}
- Remove an option to use MediaCodec SW decoder from Java layer.
- Better handling Java exceptions in JNI - detect exceptions
and either try to reset the codec or fallback to SW decoder.
- If any error is reported by codec try to fallback to SW
codec for VP8 or reset decoder and continue decoding for H.264.
- Add more logging for error conditions.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1178943007.
Cr-Commit-Position: refs/heads/master@{#9431}
Don't dismiss the presented view controller if it's already being dismissed to clear a warning about dismissing from a view controller while a dismiss is in progress.
Remove the sample buffer delegate when capture is being stopped to avoid a crash when a delegate method is sent to a deallocated object.
BUG=webrtc:4734
R=jiayl@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54669004.
Patch from Jon Hjelle <hjon@andyet.net>.
Cr-Commit-Position: refs/heads/master@{#9430}
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example:
* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika
Review URL: https://codereview.webrtc.org/1168753002
Cr-Commit-Position: refs/heads/master@{#9419}
This CL does not make any functional changes. The purpose is to extract some common code that is needed for texture capture and texture encode.
This CL does the following changes:
* Move common EGL functions from org.webrtc.MediaCodecVideoDecoder to org.webrtc.EglBase.
* Move common GL functions from org.webrtc.VideoRendererGui to org.webrtc.GlUtil and org.webrtc.GlShader.
* Remove unused call to surfaceTexture.getTransformMatrix in YuvImageRenderer.
* Add helper functions rotatedWidth()/rotatedHeight() in VideoRenderer.I420Frame.
R=glaznev@webrtc.org, hbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47309005.
Cr-Commit-Position: refs/heads/master@{#9414}
