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Author SHA1 Message Date
pbos@webrtc.org
b7192b8247 WebRtc_Word32 -> int32_t in audio_processing/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1307004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 07:50:54 +00:00
marpan@webrtc.org
557e92515d Reapply the reverted r3747. 
https://code.google.com/p/webrtc/source/detail?r=3747

r3747 timed-out on a tsan test. Verified that it passes
the test and reduced the execution time of that test (r3782).

TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1292006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 21:21:32 +00:00
hclam@chromium.org
806dc3b0e6 More trace events 
The goal of this change is to unify tracing events styles
and add trace events for all RTP traffic.

BUG=1555
Review URL: https://webrtc-codereview.appspot.com/1290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 19:54:10 +00:00
stefan@webrtc.org
4d2f5de67a Improve how NACK lists are generated before a frame has been decoded. 
BUG=1598

Review URL: https://webrtc-codereview.appspot.com/1295004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 18:24:41 +00:00
pbos@webrtc.org
ac891627c6 WebRtc_Word32 -> int32_t in audio_conference_mixer/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1306004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 17:40:15 +00:00
stefan@webrtc.org
7da3459b2a Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." 
This reverts commit 4954b3650192d78037714138a5c519ef08f2670e.
Reverts r3799

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1308004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 14:56:29 +00:00
pbos@webrtc.org
1ab45f6dd5 WebRtc_Word32 -> int32_t in video_processing/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1297006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3800 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:38:10 +00:00
stefan@webrtc.org
afcc6101d0 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. 
We should consider making the same change to the render timestamps generated at the receiver.

BUG=1563

Review URL: https://webrtc-codereview.appspot.com/1283005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:37:40 +00:00
pbos@webrtc.org
c75102eba7 WebRtc_Word32 -> int32_t in utility/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1307005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:32:55 +00:00
pbos@webrtc.org
0ea11c1768 WebRtc_Word32 -> int32_t in media_file/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1304005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:31:37 +00:00
pbos@webrtc.org
2550988baa WebRtc_Word32 -> int32_t in audio_device/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 10:30:35 +00:00
pbos@webrtc.org
0946a56023 WebRtc_Word32 => int32_t etc. in audio_coding/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1271006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 00:28:06 +00:00
pwestin@webrtc.org
6faf71d27b Remove the old unused udp_transport 
Review URL: https://webrtc-codereview.appspot.com/1272009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 23:25:25 +00:00
marpan@webrtc.org
6ff76c7404 Reduce execution time of rate control test. 
TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1289005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 20:32:48 +00:00
kma@webrtc.org
cf8e108158 Fixed a bug in isac-fix's entropy coding function: out of bounds acces to array. 
BUG=227286
Review URL: https://webrtc-codereview.appspot.com/1293005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 16:37:53 +00:00
pbos@webrtc.org
034f004a4f WebRtc_Word32 => int32_t in video_coding/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1203008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 11:13:29 +00:00
pbos@webrtc.org
2f44673d66 WebRtc_Word32 => int32_t for rtp_rtcp/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1279007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 11:08:41 +00:00
pbos@webrtc.org
ff7e1303e8 WebRtc_Word32 => int32_t remote_bitrate_estimator/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1275009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 10:04:37 +00:00
turaj@webrtc.org
2e6b7e938f In streaming mode it is preferable to fade to silence when sender stops sending, or long period of packet loss. 
test=try bots.
Review URL: https://webrtc-codereview.appspot.com/1272004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-06 00:08:11 +00:00
henrika@webrtc.org
19da719a5f Resolves TSan v2 reports data races in voe_auto_test. 
--- Note that I will add more fixes to this CL ---

BUG=1590

Review URL: https://webrtc-codereview.appspot.com/1286005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 14:34:57 +00:00
pbos@webrtc.org
b5bf54c4e7 Permit arbitrary payload names for kVideoCodecGeneric. 
BUG=1575

Review URL: https://webrtc-codereview.appspot.com/1282005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 13:27:38 +00:00
edjee@google.com
79b0289bfc Adds event traces and counters for WebRTC receive side. 
Review URL: https://webrtc-codereview.appspot.com/1279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 19:43:34 +00:00
henrika@webrtc.org
bb8ada686e TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC 
BUG=226044
TEST=content_unittests in Chrome with TSan v2 enabled

Review URL: https://webrtc-codereview.appspot.com/1201010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 08:39:09 +00:00
pbos@webrtc.org
7b859cc1e9 Webrtc_Word32 => int32_t in video_coding/main/ 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 15:54:38 +00:00
henrike@webrtc.org
cfc07c943f Revert of r3747. 
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1277005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 14:55:44 +00:00
hta@webrtc.org
95d88735ee Two more sleep calls converted to use SleepMs(). 
BUG=603

Review URL: https://webrtc-codereview.appspot.com/753005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3751 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 14:46:33 +00:00
henrika@webrtc.org
4ff956f428 Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer 
BUG=159112

Review URL: https://webrtc-codereview.appspot.com/1201007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 11:59:11 +00:00
justinlin@chromium.org
f81fad6267 Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher 
than 2^16kbps.
Review URL: https://webrtc-codereview.appspot.com/1275004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3748 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-01 22:25:11 +00:00
fbarchard@google.com
747c4cc96e For VGA (640x360), currently 1 thread is used. This change increases it to 2 threads. For HD, 4 threads are enabled. 
BUG=none
TEST=run a hangout and screencast high framerate, high resolution windows of youtube.  Observe that 1 cpu is insufficient to maintain high framerate with complex content.
Review URL: https://webrtc-codereview.appspot.com/1203006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-01 22:16:45 +00:00
marpan@webrtc.org
29f34b8727 Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549 
Review URL: https://webrtc-codereview.appspot.com/1270004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 18:57:46 +00:00
henrike@webrtc.org
93bea51517 Removed CPU APIs from VoEHardware. Code is now only used by test applications. 
Recommitting https://code.google.com/p/webrtc/source/detail?r=3736 after fixing build break.

BUG=8404677
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 15:58:49 +00:00
turaj@webrtc.org
4b1cd5c5c0 G722-stereo has been missing when creating AudioDecoder. 
Review URL: https://webrtc-codereview.appspot.com/1266004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 20:42:48 +00:00
turaj@webrtc.org
4d06db557a NetEq4 fails if the first packets inserted in are out-of-band DTMFs. 
I had to take few steps to solve this issue. I have comments on places I made cahanges to clarify why I did the change.

   
Review URL: https://webrtc-codereview.appspot.com/1195004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 18:31:42 +00:00
stefan@webrtc.org
e1a7193869 Fix flakiness in network up/down event tests when running under memcheck. 
TBR=pwestin@webrtc.org

BUG=1524

Review URL: https://webrtc-codereview.appspot.com/1261005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 17:01:48 +00:00
fischman@webrtc.org
add50b94a5 WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14. 
(required bumping minSdkVersion to 14)

This fixes a RuntimeException thrown on GalaxyNexus (but not N7, N4, or NS)
during startPreview() after the sequence of Start(), Stop(), Start(); seemingly
GN's OMX stack can't deal with parallel startPreview() & setPreviewDisplay() in
this situation.

Also:
- Only set the surface in the camera when valid
- Remove duplicate assignment
- Fix error check on voiceChannel allocation to account for multiple channel creation due to orientation change causing onDestroy()/onCreate() on the app, and rampant use of process-static holders for VoE data.

BUG=1537

Review URL: https://webrtc-codereview.appspot.com/1259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 16:48:34 +00:00
stefan@webrtc.org
bfacda60be Add interface to signal a network down event. 
- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
  buffered at the sender. When the buffer grows above the target delay
  encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
  the pacer to faster get rid of the queue after a network down event.

(Work based on issue 1237004)

BUG=1524
TESTS=trybots,vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/1258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 16:36:01 +00:00
solenberg@webrtc.org
d8a6e72057 Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1232005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-26 14:02:30 +00:00
fischman@webrtc.org
0e3077ab1f Restart Android capture after orientation change. 
Also prevent an NPE on exit.

BUG=1537

Review URL: https://webrtc-codereview.appspot.com/1248004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-25 22:08:51 +00:00
andrew@webrtc.org
c83a00ad49 Add some VoE and AudioProcessing mocks. 
Includes a bit of shared helpers in fake_common.h.

Review URL: https://webrtc-codereview.appspot.com/1221004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-25 21:20:38 +00:00
pwestin@webrtc.org
db4185664c Introduced pause and resume to the pacer 
Review URL: https://webrtc-codereview.appspot.com/1217007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3717 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 23:39:29 +00:00
pbos@webrtc.org
ae4e2b352b WebRtc_Word -> stdint in audio_coding/g711/ 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1223004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3699 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-21 13:38:29 +00:00
stefan@webrtc.org
836af79f58 Remove incorrect asserts. 
BUG=1527

Review URL: https://webrtc-codereview.appspot.com/1214006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-21 12:15:44 +00:00
pbos@webrtc.org
01b507a406 WebRtc_Word -> stdint in audio_coding/cng/ 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1222004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-21 11:28:42 +00:00
wu@webrtc.org
af33b62a72 Fix -Wstring-conversion warnings. 
Review URL: https://webrtc-codereview.appspot.com/1215006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3696 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-20 21:22:48 +00:00
vikasmarwaha@webrtc.org
455370d5b1 Thread safety issue fix in incoming_video_stream.cc. See issue 1465. 
Review URL: https://webrtc-codereview.appspot.com/1216009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3693 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-20 16:57:09 +00:00
pbos@webrtc.org
8685090060 Account for header inside I420Encoder::InitEncode. 
Also verify that the header is part of the received payload inside
I420Decoder::Decode.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1211005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3690 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 11:39:03 +00:00
stefan@webrtc.org
3d0b0d6902 Follow-up fix for r3681. 
TESTS=trybots and vie_auto_test
BUG=1514

Review URL: https://webrtc-codereview.appspot.com/1216006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3689 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 10:04:57 +00:00
kma@webrtc.org
31829a7baf Fixed initialization of SPL in echo_control_mobile. 
BUG=8403556 (a possible fix)
Review URL: https://webrtc-codereview.appspot.com/1220004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3687 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 00:25:01 +00:00
stefan@webrtc.org
f4944d49cf Fix framerate sent to account for actually sent frames. 
TESTS=trybots
BUG=1481

Review URL: https://webrtc-codereview.appspot.com/1195005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3682 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 17:04:52 +00:00
stefan@webrtc.org
abc9d5b6aa Change VCM interface to take target bitrate in bits per second. 
This also solves issue 1469.

TESTS=trybots
BUG=1469

Review URL: https://webrtc-codereview.appspot.com/1215004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 17:00:51 +00:00