Delete VideoReceiveStream channels in destructor.

R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/31909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7611 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix problem with late packets in NetEq

Since r7255, it could happen that an old packet would block the decoding
process until enough packet was received for the buffer to flush. This
CL fixes that by:
- Partially reverting r7255;
- Remove recent old packets before taking a decision for GetAudio;
- Remove all old packets after a packet has been extracted for decoding;
- Adding tests for reordered packets.

BUG=chrome:423985
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7612 4adac7df-926f-26a2-2b94-8c16560cd09d
Adds support for finch experiments to video_loopback.

Adds support for logging to stderr via -logs.

Enables abs-send-time by default.

R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7613 4adac7df-926f-26a2-2b94-8c16560cd09d
Presubmit was not whitelisting libjingle_tests.gyp or sound.gyp due to a missing comma leading to a concatenation of the two strings in the whitelist.

R=kjellander@webrtc.org
TBR=niklas.enbom@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/25999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7614 4adac7df-926f-26a2-2b94-8c16560cd09d
Falling back on single-stream on multiple SSRC.

Instead of failing, use one stream. Also clamp video min bitrate.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/31949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7615 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove protected files from talk/PRESUBMIT.py.

All files may now be committed to.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7616 4adac7df-926f-26a2-2b94-8c16560cd09d
Reworked paced sender queue

Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage.

Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these.

Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long.

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
Add kjellander as PRESUBMIT.py OWNER

BUG=
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31949005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7618 4adac7df-926f-26a2-2b94-8c16560cd09d
AppRTCDemoActivity: use differnet Themes for different API levels

The current AndroidManifest.xml hardcodes a Theme that
is only available in Android L or later (Material). To
maintain backwards compatibility, and for better App
style, create a single Theme/Style and define it for
different APIs.

I tested this in two Nexus %, one with prerelease L
and another with a KK 4.4.2 and the Theme is indeed
automagically updated :)

Note that this is compatible with
https://webrtc-codereview.appspot.com/26979004/

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7619 4adac7df-926f-26a2-2b94-8c16560cd09d
Add UMA for measuring the diff between the BWE at 2 seconds compared to the BWE at 20 seconds when the BWE should have converged.

BUG=crbug/425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7620 4adac7df-926f-26a2-2b94-8c16560cd09d
Reduce to 2 probes when probing for initial bandwidth.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23359005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7621 4adac7df-926f-26a2-2b94-8c16560cd09d
Add support for VP9 in webrtc::Call and video_loopback.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7622 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove the state_ member from AudioDecoder

The subclasses that need a state pointer should declare them---with
the right type, not void*, to get rid of all those casts.

Two small but not quite trivial cleanups are included because they
blocked the state_ removal:

  - AudioDecoderG722Stereo now inherits directly from AudioDecoder
    instead of being a subclass of AudioDecoderG722.

  - AudioDecoder now has a CngDecoderInstance member function, which
    is implemented only by AudioDecoderCng. This replaces the previous
    practice of calling AudioDecoder::state() and casting the result
    to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
    plainly visible in the AudioDecoder class declaration.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24169005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7623 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 79169148-> 79192489

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7624 4adac7df-926f-26a2-2b94-8c16560cd09d
Don't use DCHECK when you need the side effects...

R=pbos@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7625 4adac7df-926f-26a2-2b94-8c16560cd09d