34a77354a8Removed unused class members to enable clang=1 android build.
fischman@webrtc.org
2013-06-04 00:37:21 +00:00
6eb0f6a4d9Setting SSRC in vie_loopback_test
mikhal@webrtc.org
2013-06-03 22:54:40 +00:00
0a38432ea5Fix error in mixing test for supported sample rates.
andrew@webrtc.org
2013-06-03 22:52:09 +00:00
fa64a595adChange SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. This makes it easier for the users of the interface, i.e. doesn't need to remember the id in order to disable audio level indication later.
wu@webrtc.org
2013-06-03 21:27:57 +00:00
c1eb560a5cReplace the old resampler with SincResampler in the voice engine signal path.
andrew@webrtc.org
2013-06-03 19:00:29 +00:00
31c5f1c91aRemove ancient and unused CNG test.
andrew@webrtc.org
2013-06-03 16:07:07 +00:00
2b3a86554fRevert 4149 "bug fixes for extremely large images - 10000x10000 ..."
mikhal@webrtc.org
2013-05-31 22:59:38 +00:00
b1bba167f4Prevent excessive logging in jitter buffer
hclam@chromium.org
2013-05-31 18:52:16 +00:00
85f28650d5bug fixes for extremely large images - 10000x10000 and 100000 pixel wide. BUG=none TEST=libyuv unittest with manual LIBYUV_WIDTH=1000000 R=andrew@webrtc.org
fbarchard@google.com
2013-05-31 18:00:36 +00:00
a6494e6902roll libyuv to r711 for scaler fix to webrtc unittests that scale up and down and check for fairly similar results. BUG=none TEST=try bots R=andrew@webrtc.org
fbarchard@google.com
2013-05-31 16:54:42 +00:00
694cdc6e84Revert 4104 "Refactor jitter buffer to use separate lists for de..." Reason - leading suspect of video frame corruption tracked in http://b/9216252 Note that if this turns out to not be the cause, be sure to re-revert both this change and r4145.
tnakamura@webrtc.org
2013-05-31 16:09:48 +00:00
4d9c07ad6dRevert 4127 "Switch frame list implementation to std::map." We want to revert r4104 for b/9216252, but because r4127 was built on top of r4104, we need to revert r4127 first. We'll un/re-revert this if we discover that r4104 is not to blame.
tnakamura@webrtc.org
2013-05-31 16:06:01 +00:00
5ed7051799Apprtc: not to start the call until we get Turn response.
braveyao@webrtc.org
2013-05-31 06:29:41 +00:00
f9f39d59d4Add a drover.properties file for reference.
andrew@webrtc.org
2013-05-30 18:15:54 +00:00
eed919d95dMIPS optimizations for the following functions: WebRtcSpl_ComplexBitReverse, WebRtcSpl_ComplexFFT, WebRtcSpl_ComplexIFFT, WebRtcSpl_DownsampleFast and WebRtcSpl_FilterARFastQ12. Also, moved the common table used in complex_fft functions to a separate header file (webrtc/common_audio/signal_processing/include/complex_fft_tables.h).
andrew@webrtc.org
2013-05-30 16:38:36 +00:00
adc64a7216VCM/Timing: Setting clear names to members & methods
mikhal@webrtc.org
2013-05-30 16:20:18 +00:00
fddf6be339Updated apprtc to use new TURN format for chrome versions M28 & above.
vikasmarwaha@webrtc.org
2013-05-29 22:13:19 +00:00
046bc448d5Fixes the frameRate stats by grouping the frames by timestamp.
jiayl@webrtc.org
2013-05-29 16:33:46 +00:00
4213633a4dUse int for FPS instead of size_t.
pbos@webrtc.org
2013-05-29 15:13:12 +00:00
a048d7cb0aInclude files from webrtc/.. paths in rtp_rtcp/
pbos@webrtc.org
2013-05-29 14:27:38 +00:00
eea2622350Correctly set SSRCs for extra send RTP modules.
stefan@webrtc.org
2013-05-29 14:07:54 +00:00
7bdfff3503Remove assert for aborting FrameGeneratorCapturer.
pbos@webrtc.org
2013-05-29 13:58:11 +00:00
26d12105a4Fake VideoCapturer based on FrameGenerator
pbos@webrtc.org
2013-05-29 13:41:03 +00:00
08994cc525Fix a return value mismatch introduced in r4129.
stefan@webrtc.org
2013-05-29 13:28:21 +00:00
9aca5b34e1Remove #pragma once
pbos@webrtc.org
2013-05-29 13:19:09 +00:00
a5cb98cbbdBreaking out RTP header parsing from the RTP module.
stefan@webrtc.org
2013-05-29 12:12:51 +00:00
1ecee9a15aBreak video_engine/new_include/common.h into smaller parts.
pbos@webrtc.org
2013-05-29 11:34:32 +00:00
ace7ad2302Switch frame list implementation to std::map.
stefan@webrtc.org
2013-05-29 07:41:48 +00:00
f791b1cebfRename voice_engine_core -> voice_engine and move targets to voice_engine.gyp.
andrew@webrtc.org
2013-05-29 00:38:02 +00:00
a6ae644e52Add comment about test_packet_masks_metrics.
marpan@webrtc.org
2013-05-28 17:42:12 +00:00
a066cbf37cDon't return an estimated receive BW for channels not receiving video.
mflodman@webrtc.org
2013-05-28 15:00:15 +00:00
4079c31c0aInclude gflags with "gflags/gflags.h" instead of <>
pbos@webrtc.org
2013-05-28 10:38:11 +00:00
8c34ceeef1Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD
pbos@webrtc.org
2013-05-28 09:24:03 +00:00
3496ef1087Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing flakiness.
stefan@webrtc.org
2013-05-28 08:36:02 +00:00
15c1c61e2cInclude files from webrtc/.. paths in audio_conference_mixer/
pbos@webrtc.org
2013-05-28 08:13:20 +00:00
7fad4b8c9fInclude files from webrtc/.. paths in audio_processing/
pbos@webrtc.org
2013-05-28 08:11:59 +00:00
eceb53241eDefault constructors for new VideoEngine structs.
pbos@webrtc.org
2013-05-28 08:04:45 +00:00
68c05f498cRemove libvpx_intrinsics_sse4_1.a in Android.mk since this target is no longer generated in libvpx
fischman@webrtc.org
2013-05-28 05:49:43 +00:00
a6db54d4c9- Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. - Changed implementation of SetReceiveAbsoluteSendTimeStatus API so the RBE instance is changed when at least one channel in a group has the extension enabled.
solenberg@webrtc.org
2013-05-27 16:02:56 +00:00
7f944f3027Adding Mac test renderer, some test refactoring and made cpplint pass.
mflodman@webrtc.org
2013-05-27 15:52:38 +00:00
acaf3a1b13Include files from webrtc/.. paths in system_wrappers/
pbos@webrtc.org
2013-05-27 15:07:45 +00:00
1e50231ff8Include files from webrtc/.. paths in test/channel_transport/
pbos@webrtc.org
2013-05-27 15:02:23 +00:00
6f3d8fcfc0Include files from webrtc/.. paths in video_processing/
pbos@webrtc.org
2013-05-27 14:12:16 +00:00
47ce120efbInclude files from webrtc/.. paths in remote_bitrate_estimator/
pbos@webrtc.org
2013-05-27 12:41:33 +00:00
aa30bb7ef5Include files from webrtc/.. paths in common_audio/
pbos@webrtc.org
2013-05-27 09:49:58 +00:00
0afd84067aDisabling a flaky expectation in vie_autotest_rtp_rtcp.cc.
stefan@webrtc.org
2013-05-27 08:58:16 +00:00
34741c8b0eInclude files from webrtc/.. paths in test/
pbos@webrtc.org
2013-05-27 08:02:22 +00:00
7f3f8bc5a6Refactor jitter buffer to use separate lists for decodable and incomplete frames.
stefan@webrtc.org
2013-05-27 07:02:45 +00:00
ead3c6d508Fix bugs in DesktopRegion::IntersectWith() and DesktopRect::IntersectWith().
sergeyu@chromium.org
2013-05-24 21:07:20 +00:00
8665da8926Remove dead testRateControl.cc
pbos@webrtc.org
2013-05-24 13:29:29 +00:00
a01f7f6509Removed dead testH263Parser.cc
pbos@webrtc.org
2013-05-24 13:01:57 +00:00
c1f0eb2c03Remove dead bitstreamTest.cc.
pbos@webrtc.org
2013-05-24 12:46:08 +00:00
28556f5658Make sure GlxRenderer frees its resources.
pbos@webrtc.org
2013-05-24 10:54:56 +00:00
c74c3c2447Adds integration test for RTX and fixes bugs found.
stefan@webrtc.org
2013-05-23 13:48:22 +00:00
5c58f63d3fFix regression where retransmission bitrate is no longer estimated.
stefan@webrtc.org
2013-05-23 13:36:55 +00:00
d445d2229eCreateEmptyFrame casts from size_t to int.
pbos@webrtc.org
2013-05-23 12:59:51 +00:00
9b30348cfcFrameGenerator class for future fake capture device.
pbos@webrtc.org
2013-05-23 12:37:11 +00:00
771cdcbb09Control new VideoEngine tests with gflags.
pbos@webrtc.org
2013-05-23 12:20:16 +00:00
191c596912Adds print out of incoming resolution.
henrike@webrtc.org
2013-05-23 11:57:25 +00:00
a7dc37d568Log the type of recycled frames.
stefan@webrtc.org
2013-05-23 07:21:05 +00:00
8c49c1eab3Log a message when a key frame packet is received
hclam@chromium.org
2013-05-22 21:18:59 +00:00
46db413e22Fix failing tests on 32 bit Linux.
solenberg@webrtc.org
2013-05-22 20:53:42 +00:00
e46c8d3875API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.
turaj@webrtc.org
2013-05-22 20:39:43 +00:00
561990fd73- Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. - Changed RemoteBitrateObserver::OnReceivedBitrateChanged() to use a const & instead of non-const *, to avoid unnecessary copying. - Refactored RemoteBitrateEstimatorTest so it can be instantiated for both single and multi stream BWE (first using a parameterized test, but then as a standard test fixture and a few helper functions). - Refactored some tests in RemoteBitrateEstimatorTest into a common function CapacityDropTestHelper().
solenberg@webrtc.org
2013-05-22 19:04:19 +00:00
6ec25073e3Disable WindowCapturer tests on OSX and Linux
sergeyu@chromium.org
2013-05-22 18:47:07 +00:00
6ebfd346aeAdd direct_dependent_settings in common.gypi.
sergeyu@chromium.org
2013-05-22 18:22:21 +00:00
5f8f112a7bNot to request to TURN server for local tests. Follow-up work to issue1197.
braveyao@webrtc.org
2013-05-22 07:27:05 +00:00
106afffa90Roll libvpx to 196669. -pick up libvpx roll to 9981006d
marpan@webrtc.org
2013-05-21 21:19:03 +00:00
2eaf98b38bRefactor VCM/Timing. No changes in functionality.
mikhal@webrtc.org
2013-05-21 17:58:43 +00:00
3417eb49f6Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted.
stefan@webrtc.org
2013-05-21 15:25:53 +00:00
956aa7e087Include files from webrtc/.. paths in voice_engine/
pbos@webrtc.org
2013-05-21 13:52:32 +00:00
8a025e26dbMake sure VoiceEngine tests only include one test framework.
pbos@webrtc.org
2013-05-21 11:25:12 +00:00
d2541e81c6Remove <iostream> usage from loopback.cc
pbos@webrtc.org
2013-05-21 11:09:36 +00:00
375deb4e19Suffix VcmCapturer's privates with underscore_
pbos@webrtc.org
2013-05-21 09:32:22 +00:00
0d540c3762Log timestamp of the frame when it's dropped from the render module
hclam@chromium.org
2013-05-21 00:16:01 +00:00
69bb348084Log error in ViESender::SendRTCPPacket
hclam@chromium.org
2013-05-20 22:39:39 +00:00
ac0ef48631Revert 4067 "libyuv roll to r698 for Core Media fourccs for OSX ..."
andrew@webrtc.org
2013-05-20 21:36:59 +00:00
c0352d566aFix assertions in rtp_header_extension.h caused by not handling the AudioLevel extension. Added unit tests to do basic checks of the AudioLevel extension.
solenberg@webrtc.org
2013-05-20 20:55:07 +00:00
a58d7294e5libyuv roll to r698 for Core Media fourccs for OSX camtwist support and performance improvements in ARGB scaler. BUG=none TEST=libyuv unittests add CM32 and CM24 types and ARGBScaleClip tests added. Review URL: https://webrtc-codereview.appspot.com/1508004
fbarchard@google.com
2013-05-20 17:46:59 +00:00
cb9cff0c71Add functions to ViE API to enable/disable the absolute send time header extension.
solenberg@webrtc.org
2013-05-20 12:00:23 +00:00
b10ccbec02Window capturer implementation for Windows.
sergeyu@chromium.org
2013-05-19 07:02:48 +00:00
5e2a1bbbc6AppRTC: make requestTurn() failure non-fatal to call establishment.
fischman@webrtc.org
2013-05-17 18:32:23 +00:00
8d6eb56085Avoid NPE crash on Android platforms that don't support getting preview framerate. - catch Camera.setParameters() signaling errors through RuntimeException (!) - make video_demo_apk rebuild when .java sources change
fischman@webrtc.org
2013-05-17 17:33:31 +00:00
5a602d7996Enable WebRTC demo application on x86 Android
fischman@webrtc.org
2013-05-17 17:20:04 +00:00
stefan@webrtc.org