Unfortunately needs to introduce yet another workaround
script for the Visual Studio toolchain download.
This will resolve the failures with our Dr Memory Full bot
(see https://code.google.com/p/chromium/issues/detail?id=366637#c2
for details). Long term, I'm considering a better approach
than using the added gclient solution pointing at
svn://svn-mirror.golo.chromium.org/chrome/trunk/deps/third_party/drmemory/drmemory.DEPS
i.e. add an entry that we roll separately in our DEPS file
instead. However, the Dr Memory team assured that changes
in their reporting format like this are rare.
Thanks fischman@ for the video_render.gypi fix!
Thanks kma@ for the transform_neon.S fix even if it turned out
not to be needed right now (probably will come back).
BUG=chromium:366637
TEST=git try -t compile
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13369007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6010 4adac7df-926f-26a2-2b94-8c16560cd09d
This change includes max_packets_in_buffer and max_delay_ms in the
NetEq config struct. The packet buffer is also no longer limited in
terms of payload sizes (bytes), only number of packets.
The old constants governing the packet buffer limits are deleted.
BUG=3083
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5989 4adac7df-926f-26a2-2b94-8c16560cd09d
Part of the fix for crbug/360181.
On Mac/Linux, it previously continues capturing even if the window is closed.
Now it stops by returning a NULL frame.
On Windows, it used to stop capturing when the window is minimized. Now fixed to match other platforms.
Note: the crbug still needs a chrome side fix to close the notification bar.
This fix only stops the stream (i.e. stream onended event fired).
BUG=crbug/360181
TESTED=manually tested in Chrome
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/12329007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5977 4adac7df-926f-26a2-2b94-8c16560cd09d
The value of AudioFrame::energy_ must be set to -1 in order to have the
energy calculated later on in the AudioConferenceMixer module. This was
not the case in ACM2, where the value was set to 0 instead. This
resulted in bad audio for multi-party calls (5 or more participants).
Implemented a unit test to verify ACM output frame.
BUG=3255
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5969 4adac7df-926f-26a2-2b94-8c16560cd09d
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
Long, long ago, webrtc didn't support audio at 44.1 kHz. As a result we
treated 44.1 kHz audio as 44 kHz. We now have an arbitrary rate
resampler and have no trouble supporting 44.1 (see 1395 for all the
details). I must have missed updating iOS at the time.
This shouldn't result in a visible change as 16 kHz is selected as the
preferred hardware rate.
BUG=1395
R=fischman@webrtc.org, henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5957 4adac7df-926f-26a2-2b94-8c16560cd09d
Registeration of iSAC into NetEq is through injecting and external AudioDecoder. This is because iSAC encoder and decoder need to share instances for bandwidth estimator to work. When external decoder is registerred, the sampling rate of the decoder had to be specified. iSAC/48000 decoder has a native sampling rate of 32000 Hz, but it has been registered as 48000 Hz decoder.
This CL fixing this issue by letting NetEq to obtain sampling rate of an external coder according to its existing database.
BUG=3143
TEST=voe_cmd_test,modules_unittest,try-bots
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5936 4adac7df-926f-26a2-2b94-8c16560cd09d
When checking whether we need to release external decoder,
we have to do pointer comparison. We can't rely on payload
types, because payload types can be stale (e.g. before we
decode the first video frame after RegisterReceiveCodec).
This leaves a dangling pointer to external decoder, which
leads to crashes later, after we actually delete the
external decoder object.
This change has been verified in Chromecast code tree.
BUG=chromium:335539
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12049004
Patch from Sergey Volk <servolk@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5922 4adac7df-926f-26a2-2b94-8c16560cd09d
Having this override in the .gypi file avoids having to set it for the bots, which I think is best if we can avoid.
This CL also reverts r5869 so video_capture_tests are compiled for Android again.
BUG=2974,3152
TEST=Successfully ran:
git try -t compile
git try --bot=win_baremetal,mac_baremetal,linux_baremetal -t video_capture_tests
git try --bot=android_apk,android_apk_rel
Verified the change in webrtc/build/apk_tests.gyp makes the build compile successfully from a Chromium+WebRTC configured checkout for Android APK tests.
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5919 4adac7df-926f-26a2-2b94-8c16560cd09d
The design of VoeNetEqStatsTest in voice_engine_unittests depended on
being able to inject a factory for the audio coding module into
voice engine. This functionality is now likely going away, which would
make this test fail to compile. Further, the functionality under test
is mostly ACM functionality, wherefore it makes better sense to test it
at ACM level.
BUG=2996
R=henrika@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5912 4adac7df-926f-26a2-2b94-8c16560cd09d
