Remove ACM1/ACM2 switching from VoiceEngine tests

The option to run VoiceEngine tests with both ACM1 and ACM2 was
introduced while the two versions of AudioCoding module where both
in use. Now, ACM1 is being deprecated, and the tests should use the
defualt one (ACM2).

BUG=2996
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5964 4adac7df-926f-26a2-2b94-8c16560cd09d
AEC: Adds a reported_delay_enabled_ flag

Adds a feature to completely turn on or off buffer handling based on reported delay values. During startup, reported delays are controlled differently through, e.g., WEBRTC_UNTRUSTED_DELAY. By default, the feature is enabled giving the same output as before this change.

TESTED=trybots, modules_unittest
R=aluebs@webrtc.org, andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12349005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5965 4adac7df-926f-26a2-2b94-8c16560cd09d
Use unique filenames in AudioProcessingTests for parallelization.

TBR=bjornv
TESTED="gtest-parallel -w 32 --gtest_filter=*AudioProcessingTests*
out/Debug/modules_unittests" passes.

Review URL: https://webrtc-codereview.appspot.com/14369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5968 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixing a bug in ACM2 where the output frame energy was incorrectly set

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/webrtc@5969 4adac7df-926f-26a2-2b94-8c16560cd09d
Rename Start/Stop in Video{Send,Receive}Streams.

Rename {Start,Stop}{Sending,Receving} to Start/Stop. StartSending
provides no extra information in the context of a VideoSendStream, as
what it does is to send.

R=mflodman@webrtc.org
BUG=3227

Review URL: https://webrtc-codereview.appspot.com/12329005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5970 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove TraceCallback use from Call.

Non-global logging isn't supported, and having a per-call logging
dispatch seems over-eager and adds more complexity than it's worth. The
current implementation is racy on initialization due to missing atomics
support. Besides, logging support should be separate from use of Call.

R=mflodman@webrtc.org
BUG=3250,3157

Review URL: https://webrtc-codereview.appspot.com/12329006

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5971 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix the Android compilation (better structure for NetEq test libs)

This change should make the Android targets compile again. The reason
for the failure was a highly dubious structure in the gypi files. With
this fix, the structure is somewhat cleaner. Still room for improvement.

BUG=3254
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5972 4adac7df-926f-26a2-2b94-8c16560cd09d
Add keyboard channel support to AudioBuffer.

Also use local aliases for AudioBuffers for brevity.

BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5973 4adac7df-926f-26a2-2b94-8c16560cd09d
Add an output capacity parameter to ACMResampler::Resample10Msec()

Also adding a unit tests to make sure that a desired output frequency
of 0 passed to AudioCodingModule::PlayoutData10Ms() is invalid.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5974 4adac7df-926f-26a2-2b94-8c16560cd09d
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
* Cast rtp header extension to int in log in rtp_utility.cc.

BUG=3237
TEST=try bots
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5975 4adac7df-926f-26a2-2b94-8c16560cd09d
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.

BUG=3111
TEST=new performance tests
R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5976 4adac7df-926f-26a2-2b94-8c16560cd09d
Returns a NULL frame on all platforms if the captured window is closed.
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/webrtc@5977 4adac7df-926f-26a2-2b94-8c16560cd09d
Casting char to int in logs.

BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12369006

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5979 4adac7df-926f-26a2-2b94-8c16560cd09d
Support for simulating multiple independent flows in a network.

R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5980 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix leak in remote bitrate estimator tests introduced in r5980

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5981 4adac7df-926f-26a2-2b94-8c16560cd09d
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 1d95c5aecd0a4924f39f24834fb06d06e61f181e

This commit was generated by merge_from_chromium.py.

Change-Id: I2f672cab65a26078c9d4425c47474c22cda410ea
Replace scoped_array<T> with scoped_ptr<T[]>.

scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar...

except for the few not-built-on-Linux files which were updated manually.

TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
Add henrik.lundin as owner in AudioCoding module

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5988 4adac7df-926f-26a2-2b94-8c16560cd09d
Include buffer size limits in NetEq config struct

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/webrtc@5989 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix the NetEq build

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5990 4adac7df-926f-26a2-2b94-8c16560cd09d