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2017-06-14Move all libwebrtc* to vendor image.Yifan Hong
Native tests are not moved. Tests include: webrtc_apm_process_test webrtc_isac_test webrtc_apm_unit_test Test: mma -j Test: m -j -k BOARD_VNDK_VERSION=current has no errors for webrtc Bug: 62489821 Merged-In: I013303de263866cbf368f3f89327c5357f9cecdb Change-Id: I013303de263866cbf368f3f89327c5357f9cecdb (cherry picked from commit 8df7e85368569b7cd0afc1ce231b8b3a0ab49333)
2017-02-23Leave only an empty top level OWNERS file.Chih-Hung Hsieh
We should not copy OWNERS files from upstream, or the owners should be registered in Gerrit Code Review. Bug: 33166666 Test: default build targets Change-Id: Ibfd47e643f03678bb65880653383adb84809169d
2016-02-11Suppress unused-parameter warnings.Chih-Hung Hsieh
Suppress warnings until upstream can fix them. BUG: 27074506 Change-Id: If7e6f190100fba025d25d2634d1c9a657cc24854
2016-01-15Merge upstream SHA 04cb763Alex Luebs
* git merge 04cb763 * See all upstream changes since the previous merge in branch aosp/upstream-master: git diff cb3f9bd..04cb763 * Modify webrtc/.gitignore to keep *.mk files. * Removed old files from *.mk files: - thread.cc - thread_posix.cc * Add new files to *.mk files: - event_tracer.cc * Android relevant upstream changes: - Make Beamforming dynamically settable for Android platform builds - Remove additional channel constraints when Beamforming is enabled in AudioProcessing - Use an explicit identifier in Config Change-Id: I384a4e8f6982c31c5bc70eef521bb2d90800b9a4
2016-01-14Eliminate defines in talk/kjellander
Replace LINUX, OSX and IOS defines with WEBRTC_ prefixed versions. Remove no longer used defines from talk/build/common.gypi due to previously migrated sources (into webrtc/p2p and webrtc/libjingle). When this is rolled into Chromium, we can also clean up the platform defines in https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp NOTRY=True BUG=webrtc:5420 TESTED=Ran all compile trybots with --clobber flag. TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1588453005 Cr-Commit-Position: refs/heads/master@{#11254}
2016-01-14Re-land: "Use an explicit identifier in Config"aluebs
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS. Original CL: https://codereview.webrtc.org/1538643004/ TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1589573004 Cr-Commit-Position: refs/heads/master@{#11248}
2016-01-13Revert of Use an explicit identifier in Config (patchset #4 id:60001 of ↵tommi
https://codereview.webrtc.org/1538643004/ ) Reason for revert: Reverting due to problem with roll: /b/build/slave/linux/build/src/buildtools/linux64/gn gen //out/Release '--args=ffmpeg_branding="Chrome" proprietary_codecs=true is_debug=false is_component_build=false use_goma=true goma_dir="/b/build/goma" symbol_level=1 dcheck_always_on=true' --check --runtime-deps-list-file=/b/build/slave/linux/build/src/out/Release/runtime_deps -> returned 1 ERROR at //third_party/webrtc/BUILD.gn:245:18: Item not found configs -= [ "//build/config/clang:find_bad_constructs" ] ^----------------------------------------- You were trying to remove "//build/config/clang:find_bad_constructs" from the list but it wasn't there. GN gen failed: 1 step returned non-zero exit code: 1 @@@STEP_FAILURE@@@ Original issue's description: > Use an explicit identifier in Config > > This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS. > > Committed: https://crrev.com/25249d92d3cf105bcc7b684c8924ccdbc9afcb93 > Cr-Commit-Position: refs/heads/master@{#11231} TBR=henrik.lundin@webrtc.org,stefan@webrtc.org,tommi@chromium.org,aluebs@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1586563003 Cr-Commit-Position: refs/heads/master@{#11239}
2016-01-13Clean the code for external denoiser.jackychen
BUG=webrtc:5255 Review URL: https://codereview.webrtc.org/1578373003 Cr-Commit-Position: refs/heads/master@{#11235}
2016-01-13[rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on ↵danilchap
the way Class renamed to indicated use of the rtcp::Empty class: it can only be used as container for other rtcp packets. All tests that use Append function moved from rtcp_packet_unittest, even if they did not use Empty class. This is because there is plan to make RtcpPacket class lighter by moving Append functionality to CompoundPacket class. BUG=webrtc:5260 R=åsapersson Review URL: https://codereview.webrtc.org/1582503002 Cr-Commit-Position: refs/heads/master@{#11234}
2016-01-13Use an explicit identifier in Configaluebs
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS. Review URL: https://codereview.webrtc.org/1538643004 Cr-Commit-Position: refs/heads/master@{#11231}
2016-01-13Convert channel counts to size_t.Peter Kasting
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1316523002 . Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-12[rtp_rtcp] rtcp::Sli packet moved into own file and got Parse functiondanilchap
BUG=webrtc:5260 R=åsapersson Review URL: https://codereview.webrtc.org/1551893002 Cr-Commit-Position: refs/heads/master@{#11228}
2016-01-12NetEq: Fix a typo in a commentHenrik Lundin
TBR=minyue@webrtc.org NOTRY=true Review URL: https://codereview.webrtc.org/1578223003 . Cr-Commit-Position: refs/heads/master@{#11226}
2016-01-12Wire-up BWE feedback for audio receive streams.Stefan Holmer
Also wires up receiving transport sequence numbers. BUG=webrtc:5263 R=mflodman@webrtc.org, pbos@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1535963002 . Cr-Commit-Position: refs/heads/master@{#11220}
2016-01-12Add unit test for stand-alone denoiser and fixed some bugs.jackychen
The unit test will run the pure C denoiser and SSE2/NEON denoiser (based on the CPU detection) and compare the denoised frames to ensure the bit exact. TBR=tommi@webrtc.org BUG=webrtc:5255 Review URL: https://codereview.webrtc.org/1492053003 Cr-Commit-Position: refs/heads/master@{#11216}
2016-01-12Remove additional channel constraints when Beamforming is enabled in ↵aluebs
AudioProcessing The general constraints on number of channels for AudioProcessing is: num_in_channels == num_out_channels || num_out_channels == 1 When Beamforming is enabled and additional constraint was added forcing: num_out_channels == 1 This artificial constraint was removed by adding upmixing support in CopyTo, since it was already supported for the AudioFrame interface using InterleaveTo. Review URL: https://codereview.webrtc.org/1571013002 Cr-Commit-Position: refs/heads/master@{#11215}
2016-01-12Make Beamforming dynamically settable for Android platform buildsaluebs
Review URL: https://codereview.webrtc.org/1563493005 Cr-Commit-Position: refs/heads/master@{#11213}
2016-01-11clang-format audio_device/mac.andrew
NOTRY=true Review URL: https://codereview.webrtc.org/1570063003 Cr-Commit-Position: refs/heads/master@{#11212}
2016-01-11[rtp_rtcp] rtcp::Tmmbr moved into own filedanilchap
BUG=webrtc:5260 R=åsapersson Review URL: https://codereview.webrtc.org/1575023002 Cr-Commit-Position: refs/heads/master@{#11206}
2016-01-11H.264: Default flags and pulling in openh264 and ffmpeg.hbos
Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg). These will be used in a follow-up CL that adds an encoder and decoder implementation. The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added. BUG=468365 Review URL: https://codereview.webrtc.org/1575913003 Cr-Commit-Position: refs/heads/master@{#11204}
2016-01-11[rtp_rtcp] rtcp::Tmmbn moved into own filedanilchap
explicetly unchanged. BUG=webrtc:5260 R=åsapersson Review URL: https://codereview.webrtc.org/1578713002 Cr-Commit-Position: refs/heads/master@{#11201}
2016-01-08Misc. small cleanups.pkasting
* Better param names * Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases. * Use arraysize() * Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers * reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead * Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition * Fix indenting * Use uint32_t for timestamps (matching how it's already a uint32_t in most places) * Spelling * RTC_CHECK_EQ(expected, actual) * Rewrap * Use .empty() * Be more pedantic about matching int/int32_t/ * Remove pointless consts on input parameters to functions * Add missing sanity checks All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first. BUG=none TEST=none Review URL: https://codereview.webrtc.org/1534193008 Cr-Commit-Position: refs/heads/master@{#11191}
2016-01-08Cleaning neteq_unittest resource files.minyue
BUG=webrtc:2692 Review URL: https://codereview.webrtc.org/1563983003 Cr-Commit-Position: refs/heads/master@{#11189}
2016-01-08Add tracing to NetEqImpl::GetAudiohenrik.lundin
BUG=webrtc:5167 R=pbos@webrtc.org NOTRY=true Review URL: https://codereview.webrtc.org/1571693002 Cr-Commit-Position: refs/heads/master@{#11183}
2016-01-08Check the mic volume only periodically on Mac.andrew
Ask the OS for the mic volume every 1 second rather than with every 10 ms chunk. The previous behavior was consuming ~2% of the CPU load of a voice engine call, and is now negligible. This is consistent with the webrtc Windows Core Audio implementation, as well as the Chromium Mac implementation: https://code.google.com/p/chromium/codesearch#chromium/src/media/audio/agc_audio_stream.h TEST=voe_cmd_test with AGC continues to work well on Mac. Review URL: https://codereview.webrtc.org/1564223002 Cr-Commit-Position: refs/heads/master@{#11182}
2016-01-08Disable AudioDeviceAPITest.MicrophoneVolumeTests on Linux.kjellander
NOTRY=True BUG=webrtc:5414 TBR=pbos@webrtc.org Review URL: https://codereview.webrtc.org/1572503002 Cr-Commit-Position: refs/heads/master@{#11178}
2016-01-07Fixing integer underflow in FileAudioDevice (webrtc issue 4554)A.Brauckmann
Problem is described here: https://code.google.com/p/webrtc/issues/detail?id=4554 Review URL: https://codereview.webrtc.org/1295603002 Cr-Commit-Position: refs/heads/master@{#11174}
2016-01-07vp9 tests: Adjust some parameters and re-enable the tests.Marco
Tests were failing on android with new libvpx. vp9 speed setting was changed to 8 recently and some recent changes in libvpx require update for the tests to pass. TBR=stefan@webrtc.org BUG=webrtc:5401 Review URL: https://codereview.webrtc.org/1569903002 . Cr-Commit-Position: refs/heads/master@{#11173}
2016-01-07Add ImplementationName to SimulcastEncoderAdapter.pbos
BUG=webrtc:4897 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1555673002 Cr-Commit-Position: refs/heads/master@{#11170}
2016-01-07iOS stability improvement for device switching, including BT deviceshenrika
BUG=webrtc:5058 Review URL: https://codereview.webrtc.org/1554163002 Cr-Commit-Position: refs/heads/master@{#11168}
2016-01-07Remove unused methods in VideoCodingModule.Peter Boström
Also voids ::Codec which always passed. BUG= R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1464313004 . Cr-Commit-Position: refs/heads/master@{#11167}
2016-01-07Use pointer to generated FEC packet.Peter Boström
Removes multiple index lookups to generated_fec_packets_ speeding up FecTest.FecTest with >2x in both Debug and Release, improving performance but also readability. On Debug this means that the slowest test in modules_tests now takes ~15-20 seconds instead of 50+ seconds, reducing the overall bottleneck. BUG=webrtc:4712 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1552563003 . Cr-Commit-Position: refs/heads/master@{#11166}
2016-01-07AudioDeviceTest.StartPlayoutOnTwoInstances now verifies two active playing ↵henrika
streams TBR=tkchin_webrtc BUG=b/25343768 Review URL: https://codereview.webrtc.org/1527143007 . Cr-Commit-Position: refs/heads/master@{#11165}
2016-01-07Roll chromium_revision 2a70cb1..4662d4f (367468:368042)kjellander
I had to fix the audio_device BUILD.gn which was forgotten back in https://codereview.webrtc.org/1536923003. It also contained a few missing source files and one library. Change log: https://chromium.googlesource.com/chromium/src/+log/2a70cb1..4662d4f Full diff: https://chromium.googlesource.com/chromium/src/+/2a70cb1..4662d4f Changed dependencies: * src/buildtools: https://chromium.googlesource.com/chromium/buildtools.git/+log/6d0c448..0f8e6e4 * src/third_party/libsrtp: https://chromium.googlesource.com/chromium/deps/libsrtp.git/+log/8a7662a..ebfcc9a DEPS diff: https://chromium.googlesource.com/chromium/src/+/2a70cb1..4662d4f/DEPS No update to Clang. TBR=henrika@webrtc.org NOTRY=True Review URL: https://codereview.webrtc.org/1565093002 Cr-Commit-Position: refs/heads/master@{#11162}
2016-01-07Roll chromium_revision 4df108a..2a70cb1 (367307:367468)kjellander
Mac 32-bit support has been gone in Chromium for a long time, but was removed in https://codereview.chromium.org/1557823002. This called for finally removing our Mac 32-bit builds, which was done in http://crbug.com/574320. Change log: https://chromium.googlesource.com/chromium/src/+log/4df108a..2a70cb1 Full diff: https://chromium.googlesource.com/chromium/src/+/4df108a..2a70cb1 Changed dependencies: * src/third_party/libvpx_new/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/ecb8dff..a9dd8a7 * src/third_party/nss: https://chromium.googlesource.com/chromium/deps/nss.git/+log/aee1b12..225bfc3 DEPS diff: https://chromium.googlesource.com/chromium/src/+/4df108a..2a70cb1/DEPS No update to Clang. TBR=marpan@webrtc.org, stefan@webrtc.org, BUG=webrtc:5401, webrtc:5402 NOTRY=True Review URL: https://codereview.webrtc.org/1556273002 Cr-Commit-Position: refs/heads/master@{#11159}
2016-01-06Disable VideoCaptureTest.Capabilities and CreateDelete fails on Mackjellander
These tests started failing on the bots after switching the build from 32 to 64-bit. NOTRY=True BUG=webrtc:5406 TBR=perkj@webrtc.org Review URL: https://codereview.webrtc.org/1566683002 Cr-Commit-Position: refs/heads/master@{#11154}
2016-01-04Remove DISABLED_ON_ macros.Peter Boström
Macro incorrectly displays DISABLED_ON_ANDROID in test names for parameterized tests under --gtest_list_tests, causing tests to be disabled on all platforms since they contain the DISABLED_ prefix rather than their expanded variants. This expands the macro variants to inline if they're disabled or not, and removes building some tests under configurations where they should fail, instead of building them but disabling them by default. The change also removes gtest_disable.h as an unused include from many other files. BUG=webrtc:5387, webrtc:5400 R=kjellander@webrtc.org, phoglund@webrtc.org TBR=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1547343002 . Cr-Commit-Position: refs/heads/master@{#11150}
2016-01-04Move fake-handle frame creation into test target.Peter Boström
Renames CreateFakeNativeHandleFrame to FakeNativeHandle::CreateFrame and moves into test.gyp target 'fake_video_frames' which contains previous frame_generator target. Removes unused warnings from includers of webrtc/test/fake_texture_frame.h which did not use the function above. BUG=webrtc:5398 R=kjellander@webrtc.org TBR=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1554223002 . Cr-Commit-Position: refs/heads/master@{#11149}
2015-12-29[rtp_rtcp] Fix CL#1539423003danilchap
public function RtpHeaderParser::Parse with old signature restored as deprecated. BUG=webrtc:5277 TBR=åsapersson NOTRY=True Review URL: https://codereview.webrtc.org/1550283002 Cr-Commit-Position: refs/heads/master@{#11135}
2015-12-28[rtp_rtcp] Lint errors cleaned from rtp_utilitydanilchap
R=åsapersson BUG=webrtc:5277 Review URL: https://codereview.webrtc.org/1539423003 Cr-Commit-Position: refs/heads/master@{#11131}
2015-12-22Adding bit exactness test for Opus decoding in NetEq.minyue
Opus has become the mostly used codec in WebRTC. There, however, is no bit exactness test for Opus decoding in NetEq. The new RTP file is generated by the following steps: 1. Encode a clean RTP file with Opus RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960 opus 1 2. Adding jitter to the clean RTP file RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp (Note: jitter.dat does not exist in WebRTC resources folder. Check the source code for RTPjitter to know how to define such a file.) BUG=webrtc:3987 TEST=observed Opus normal decoding and FEC decoding were used, listened to the reference output. Review URL: https://codereview.webrtc.org/1515113002 Cr-Commit-Position: refs/heads/master@{#11113}
2015-12-22[rtp_rtcp] cleanup in RTCPSender class internals.danilchap
PrepareReportBlock and AddReportBlock private functions merged: PrepareReportBlock moved report block from statistic to temporary structure AddReportBlock copied that temporary structure into temporary map right after. Thanks to rtcp packet classes that temporary structure is now unneccesary. BUG=webrtc:5260 R=åsapersson Review URL: https://codereview.webrtc.org/1538833002 Cr-Commit-Position: refs/heads/master@{#11112}
2015-12-22rtcp::Nack packet moved into own file and got Parse functiondanilchap
Review URL: https://codereview.webrtc.org/1461623003 Cr-Commit-Position: refs/heads/master@{#11111}
2015-12-21[rtp_rtcp] time helper functionsdanilchap
RTP timestams helper functions moved from rtp_utility added functions to deal with CompactNtp timestamps R=åsapersson BUG=webrtc:5260 Review URL: https://codereview.webrtc.org/1535113002 Cr-Commit-Position: refs/heads/master@{#11106}
2015-12-21Lint fix for webrtc/modules/video_coding PART 3!philipel
Trying to submit all changes at once proved impossible since there were too many changes in too many files. The changes to PRESUBMIT.py will be uploaded in the last CL. (original CL: https://codereview.webrtc.org/1528503003/) BUG=webrtc:5309 TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1540243002 Cr-Commit-Position: refs/heads/master@{#11105}
2015-12-21Lint fix for webrtc/modules/video_coding PART 2!philipel
Trying to submit all changes at once proved impossible since there were too many changes in too many files. The changes to PRESUBMIT.py will be uploaded in the last CL. (original CL: https://codereview.webrtc.org/1528503003/) BUG=webrtc:5309 TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1543503002 Cr-Commit-Position: refs/heads/master@{#11102}
2015-12-21Lint fix for webrtc/modules/video_coding PART 1!philipel
Trying to submit all changes at once proved impossible since there were too many changes in too many files. The changes to PRESUBMIT.py will be uploaded in the last CL. (original CL: https://codereview.webrtc.org/1528503003/) BUG=webrtc:5309 TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1541803002 Cr-Commit-Position: refs/heads/master@{#11100}
2015-12-21Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that ↵asapersson
these are for infrequent updates. This implementation will be replaced by a faster one and sparse will be removed. BUG=webrtc:5283 Review URL: https://codereview.webrtc.org/1530913002 Cr-Commit-Position: refs/heads/master@{#11099}
2015-12-21A unittest that reports the statistics for the duration of an APM stream ↵peah
processing API call. BUG=webrtc:5099 Committed: https://crrev.com/880896ab0976bbf86a6753d0c900c70e51f421cb Cr-Commit-Position: refs/heads/master@{#10786} Review URL: https://codereview.webrtc.org/1436553004 Cr-Commit-Position: refs/heads/master@{#11098}
2015-12-19Revert of Reland "Added option to specify a maximum file size when recording ↵ivoc
an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ ) Reason for revert: Compile error on Android needs to be fixed before relanding. Original issue's description: > Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. > > The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4. > Original review: https://codereview.webrtc.org/1413483003/ > > The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function. > > NOTRY=true > TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org > BUG=webrtc:4741 > > Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a > Cr-Commit-Position: refs/heads/master@{#11093} TBR=glaznev@webrtc.org,henrik.lundin@webrtc.org,solenberg@google.com,henrikg@webrtc.org,perkj@webrtc.org,kwiberg@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4741 Review URL: https://codereview.webrtc.org/1537213002 Cr-Commit-Position: refs/heads/master@{#11094}
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