With these tests excluded and failures suppressed
we should be able to bring Dr Memory Full into a
green state in
http://build.chromium.org/p/client.webrtc.fyi/waterfall
so we can move the bots into the main waterfall.
BUG=3158, 3159
TEST=Ran successful runs of the tests that never completed
using the reproduction steps in the issues listed above on
a local Windows box. The tests that just failed weren't tried,
since they cannot have been blocking other possibly failing
tests in the same binary.
R=pbos@webrtc.org
TBR=pbos
Review URL: https://webrtc-codereview.appspot.com/11209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5850 4adac7df-926f-26a2-2b94-8c16560cd09d
- r5834 made it so that empty fields are a fatal SDP parsing error, exposing
opportunities for improvement in the preferISAC; changed split/join to use
\r\n instead of \n and now omitting the trailing space on the m=audio line
that triggered the new failure.
- DTLS requires a different role for each endpoint so conflicts with loopback
calling. apprtc.py suppresses DTLS for that reason in loopback calls, so the
android demo app now only enables DTLS by default if it is not suppressed by a
constraint (matching Chrome).
BUG=3164,3165,2507
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5847 4adac7df-926f-26a2-2b94-8c16560cd09d
Replace the two versions with a single DownConvertToCodecFormat. As
mentioned in comments, this could be further consolidated with
RemixAndResample but we should write a full audio converter class in
that case.
Along the way:
- Fix the bug present in Channel::Demultiplex with mono input and a
stereo codec.
- Remove the 32 kHz max from the OnDataAvailable path. This avoids a
48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we
get a straight pass-through to ACM. The 32 kHz conversion is still
needed in the RecordedDataIsAvailable path until APM natively supports
48 kHz.
- Merge resampler improvements from ACM1 to ACM2. This allows ACM to
handle 44.1 kHz audio passed to VoE and was originally done here:
https://webrtc-codereview.appspot.com/1590004
- Reuse the RemixAndResample unit tests for DownConvertToCodecFormat.
- Remove unused functions from utility.cc.
BUG=3155,3000,b/12867572
TESTED=voe_cmd_test using both the OnDataAvailable and
RecordedDataIsAvailable paths, with a captured audio format of all
combinations of {44.1,48} kHz and {1,2} channels, running through all
codecs, and finally using both ACM1 and ACM2.
R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
The tool RTPanalyze (used to process an input RTP dump into a
text file of RTP header info) was present in both the neteq and
neteq4 folders. This change pulls in changes from the old to the new
and renames the source file and tool to rtp_analyze.
Removing special code for dummy-rtp files (it is supported without
special code), and making the RED payload type settable using flags.
Moving from test/ to tools/ folder.
BUG=2692
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5832 4adac7df-926f-26a2-2b94-8c16560cd09d
This disables GN use for the moment (Chromium
has disabled it for now but plan to pick up the
work at a later stage). I'm leaving the rest of
the GN stuff in our DEPS since that's how
the Chromium DEPS currently looks like.
Overview of changes in Chrome DEPS:
$ svn diff http://src.chromium.org/chrome/trunk/src/DEPS -r 255773:260462
which can be compared with the output of:
$ svn cat http://webrtc.googlecode.com/svn/trunk/DEPS | grep chromium_deps | sed 's/^ *//' | sort | uniq
in a WebRTC checkout, gives the following relevant changes:
* third_party/android_tools 0582bd:ca3567
* third_party/icu 249466:259309
* third_party/libjpeg_turbo 251747:259851
* third_party/libyuv 979:986
* third_party/nss 254867:259440
* tools/gyp 1860:1880
The following also shows that Clang is upgraded from r198389 to r202554:
$ svn diff http://src.chromium.org/chrome/trunk/src/tools/clang/scripts/update.sh -r 255773:260462
TEST=trybots
BUG=None
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5822 4adac7df-926f-26a2-2b94-8c16560cd09d
Previously GAE Channel callbacks would be handled by JS string-encoding the
payload into a URL. Unfortunately this is limited to the (undocumented,
silently problematic) maximum URL length UIWebView supports. Replaced this
scheme by a notification from JS to ObjC and a getter from ObjC to JS (which
happens out-of-line to avoid worrying about UIWebView's re-entrancy, or lack
thereof). Part of this change also moved from a combination of: JSON,
URL-escaping, and ad-hoc :-separated values to simply JSON.
Also incidentally:
- Removed outdated TODO about onRenegotiationNeeded, which is unneeded
- Move handling of PeerConnection callbacks to the main queue to avoid having
to think about concurrency too hard.
- Replaced a bunch of NSOrderedSame with isEqualToString for clearer code and
not having to worry about the fact that [nil compare:@"foo"]==NSOrderedSame
is always true (yay ObjC!).
- Auto-scroll messages view.
BUG=3117
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/10899006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5814 4adac7df-926f-26a2-2b94-8c16560cd09d
Chromium defines ENABLE_PROFILING under the gyp flag profiling=1. This
corrects the resulting mulitple defintion error:
../../talk/base/profiler.h:61:9: error: 'ENABLE_PROFILING' macro redefined [-Werror]
#define ENABLE_PROFILING
and allows us to use profiling=1 in standalone builds.
TESTED=build passes with profiling=1
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5804 4adac7df-926f-26a2-2b94-8c16560cd09d
