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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
2016-01-14Eliminate 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}
2016-01-14Revert of Update with new default boringssl no-aes cipher suites. Re-enable ↵sprang
tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ ) Reason for revert: We're getting boringssl version conflicts. Reverting for now. Original issue's description: > Update with new default boringssl no-aes cipher suites. Re-enable tests. > > This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part). > > BUG=webrtc:5381 > R=davidben@webrtc.org, henrika@webrtc.org > > Committed: https://crrev.com/31c8d2eac5aec977f584ab0ae5a1d457d674f101 > Cr-Commit-Position: refs/heads/master@{#11250} TBR=davidben@webrtc.org,henrika@webrtc.org,torbjorng@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5381 Review URL: https://codereview.webrtc.org/1586183002 Cr-Commit-Position: refs/heads/master@{#11253}
2016-01-14Update with new default boringssl no-aes cipher suites. Re-enable tests.Torbjorn Granlund
This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part). BUG=webrtc:5381 R=davidben@webrtc.org, henrika@webrtc.org Review URL: https://codereview.webrtc.org/1550773002 . Cr-Commit-Position: refs/heads/master@{#11250}
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}
2016-01-14Revert of Delete remnants of non-square pixel support from ↵nisse
cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ ) Reason for revert: These changes broke chrome. Need to temporarily keep methods InitToEmptyBuffer, InitToBlack, CreateEmptyFrame with old but ignored arguments for pixel_width and pixel_height. Then update chrome, and delete the old methods in a separate cl. Original issue's description: > Delete remnants of non-square pixel support from cricket::VideoFrame. > > If ever needed, add some aspect ratio parameter, without pixel_width > and pixel_height arguments cluttering commonly used functions. > > BUG=webrtc:5426 > > Committed: https://crrev.com/709513d4133107d5c02aed34a5ee99444c4d4e25 > Cr-Commit-Position: refs/heads/master@{#11243} TBR=pthatcher@webrtc.org,perkj@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5426 Review URL: https://codereview.webrtc.org/1583223002 Cr-Commit-Position: refs/heads/master@{#11246}
2016-01-14Delete remnants of non-square pixel support from cricket::VideoFrame.nisse
If ever needed, add some aspect ratio parameter, without pixel_width and pixel_height arguments cluttering commonly used functions. BUG=webrtc:5426 Review URL: https://codereview.webrtc.org/1586613002 Cr-Commit-Position: refs/heads/master@{#11243}
2016-01-13Revert of Storing raw audio sink for default audio track. (patchset #7 ↵deadbeef
id:120001 of https://codereview.chromium.org/1551813002/ ) Reason for revert: tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach. Original issue's description: > Storing raw audio sink for default audio track. > > BUG=webrtc:5250 > > Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99 > Cr-Commit-Position: refs/heads/master@{#11230} TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5250 Review URL: https://codereview.webrtc.org/1588693002 Cr-Commit-Position: refs/heads/master@{#11241}
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}
2016-01-13Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSankjellander
BUG=webrtc:4963 TBR=pbos@webrtc.org NOTRY=True Review URL: https://codereview.webrtc.org/1577233005 Cr-Commit-Position: refs/heads/master@{#11237}
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}
2016-01-13Storing raw audio sink for default audio track.deadbeef
BUG=webrtc:5250 Review URL: https://codereview.webrtc.org/1551813002 Cr-Commit-Position: refs/heads/master@{#11230}
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}
2016-01-12Delete unused method webrtc::VideoRendererInterface::SetSize.nisse
BUG=webrtc:5426 Review URL: https://codereview.webrtc.org/1582493002 Cr-Commit-Position: refs/heads/master@{#11223}
2016-01-12Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy ↵nisse
later removal. BUG=webrtc:5426 Review URL: https://codereview.webrtc.org/1581583002 Cr-Commit-Position: refs/heads/master@{#11218}
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}
2016-01-11Change DTLS default from 1.0 to 1.2 for webrtc.Guo-wei Shieh
This changes for standalone webrtc applications. BUG= R=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1548733002 . Cr-Commit-Position: refs/heads/master@{#11211}
2016-01-11SCTP: Stopped accepting SSRCs higher than max.lally
Seems to fix asan-related crash. BUG=https://code.google.com/p/chromium/issues/detail?id=570261 Review URL: https://codereview.webrtc.org/1571853002 Cr-Commit-Position: refs/heads/master@{#11205}
2016-01-08Properly handle different transports having different SSL roles.Taylor Brandstetter
This meant splitting "transport_options" into audio/video/data options, for when creating the answer, and giving "GetSslRole" a "transport_name" parameter so we can retrieve the current role on a per-transport basis. BUG=webrtc:4525 R=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1516993002 . Cr-Commit-Position: refs/heads/master@{#11192}
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}
2016-01-08Reland "Add APK targets to build libjingle tests for Android."phoglund
patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ This reverts commit bc14164aad254e72ce4d1e381b912b7d3acf5391. We have made more preparations downstream, so this should work now. Original CL by perkj@. BUG=webrtc:2365 The work started from the work by kjellander@ in https://codereview.webrtc.org/1413663003/ Review URL: https://codereview.webrtc.org/1570513004 Cr-Commit-Position: refs/heads/master@{#11186}
2016-01-08Fix clang warning in peerconnection_jni.ccperkj
TEST= export GYP_DEFINES="OS=android clang=1" ... ninja -C out/Debug AppRTCDemo BUG=webrtc:5399 Review URL: https://codereview.webrtc.org/1561073005 Cr-Commit-Position: refs/heads/master@{#11181}
2016-01-07Adding unit test to ensure TURN server priorities are unique.Taylor Brandstetter
BUG=webrtc:5209 R=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1570563002 . Cr-Commit-Position: refs/heads/master@{#11177}
2016-01-07Adding a way for a Java RtpSender to set a track without taking ownership.Taylor Brandstetter
This means that the track will still have a reference count after the PeerConnection and RtpSender have been destroyed. R=glaznev@webrtc.org, pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1566103003 . Cr-Commit-Position: refs/heads/master@{#11176}
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}
2016-01-04Roll chromium_revision d66326c..4df108a (367167:367307)kjellander
The changes in https://chromium.googlesource.com/chromium/src/+/d66326c..4df108a/build/common.gypi enables a lot more warnings, which have been disabled/fixed in this CL. See tracking bugs for remaining work. Change log: https://chromium.googlesource.com/chromium/src/+log/d66326c..4df108a Full diff: https://chromium.googlesource.com/chromium/src/+/d66326c..4df108a Changed dependencies: * src/buildtools: https://chromium.googlesource.com/chromium/buildtools.git/+log/fee7f1e..6d0c448 * src/third_party/libsrtp: https://chromium.googlesource.com/chromium/deps/libsrtp.git/+log/b8dd754..8a7662a DEPS diff: https://chromium.googlesource.com/chromium/src/+/d66326c..4df108a/DEPS No update to Clang. BUG=webrtc:5397, webrtc:5398, webrtc:5399 TBR=hta@webrtc.org, perkj@webrtc.org NOTRY=True Review URL: https://codereview.webrtc.org/1553033002 Cr-Commit-Position: refs/heads/master@{#11147}
2015-12-29Removing webrtc::PortAllocatorFactoryInterface.Taylor Brandstetter
ICE servers are now passed directly into PortAllocator, making PortAllocatorFactoryInterface redundant. This CL also moves SetNetworkIgnoreMask to PortAllocator. R=phoglund@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org Review URL: https://codereview.webrtc.org/1520963002 . Cr-Commit-Position: refs/heads/master@{#11139}
2015-12-28Fixing issue where description contains empty ICE ufrag/pwd.deadbeef
The issue occurred when deserializing and then serializing a rejected content description, which doesn't have the ICE ufrag/pwd in the first place. BUG=webrtc:5105 Review URL: https://codereview.webrtc.org/1534363002 Cr-Commit-Position: refs/heads/master@{#11134}
2015-12-21Deleted VideoCapturer::screencast_max_pixels, together withnisse
VideoChannel::GetScreencastMaxPixels and VideoChannel::GetScreencastFps. Unused in webrtc, also unused in everything indexed by google and chromium code search. With the exception of the magicflute plugin, which I'm told doesn't matter. Review URL: https://codereview.webrtc.org/1532133002 Cr-Commit-Position: refs/heads/master@{#11108}
2015-12-20Roll chromium_revision 1b6c421..db567a8 (365999:366304)kjellander
I had to disable some Dtls12Both tests failing under MSan (see bug). Notice those errors started happening in the range of https://boringssl.googlesource.com/boringssl.git/+log/afd565f..9f897b2 while this CL brings in an even newer BoringSSL (that still has the same problem). Change log: https://chromium.googlesource.com/chromium/src/+log/1b6c421..db567a8 Full diff: https://chromium.googlesource.com/chromium/src/+/1b6c421..db567a8 Changed dependencies: * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/afd565f..afe57cb * src/third_party/libyuv: https://chromium.googlesource.com/libyuv/libyuv.git/+log/1019e45..1ccbf8f * src/third_party/nss: https://chromium.googlesource.com/chromium/deps/nss.git/+log/a676aa0..aee1b12 DEPS diff: https://chromium.googlesource.com/chromium/src/+/1b6c421..db567a8/DEPS No update to Clang. NOTRY=True BUG=webrtc:5381 TBR=torbjorng@webrtc.org, pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1533253002 Cr-Commit-Position: refs/heads/master@{#11095}
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}
2015-12-19Reland "Added option to specify a maximum file size when recording an AEC ↵ivoc
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 Review URL: https://codereview.webrtc.org/1541633002 Cr-Commit-Position: refs/heads/master@{#11093}
2015-12-19Adding a MediaStream parameter to createSender.deadbeef
This will allow an app to create senders with the same stream id, without SDP munging. Review URL: https://codereview.webrtc.org/1538673002 Cr-Commit-Position: refs/heads/master@{#11092}
2015-12-18Revert of Added option to specify a maximum file size when recording an AEC ↵ivoc
dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ ) Reason for revert: Breaks Chrome-FYI bots because of a change in the StartDebugRecording function in audio_processing.h, that is called from Chrome. Original issue's description: > Added option to specify a maximum file size when recording an AEC dump. > > For applications with a strict filesize limit for debug files, > I added an option to specify a maximum filesize for AEC dumps. An > existing unit test is extended to check that the feature works as > advertised. > > BUG=webrtc:4741 > TBR=glaznev@webrtc.org > > Committed: https://crrev.com/ae2c5ad12afc8cc29fe9c59dea432b697b871a87 > Cr-Commit-Position: refs/heads/master@{#11081} TBR=pthatcher@webrtc.org,henrik.lundin@webrtc.org,henrikg@webrtc.org,solenberg@webrtc.org,andrew@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org,glaznev@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4741 Review URL: https://codereview.webrtc.org/1533913004 Cr-Commit-Position: refs/heads/master@{#11087}
2015-12-18Expose codec implementation names in stats.Peter Boström
Used to distinguish between software/hardware encoders/decoders and other implementation differences. Useful for tracking quality regressions related to specific implementations. BUG=webrtc:4897 R=hta@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1406903002 . Cr-Commit-Position: refs/heads/master@{#11084}
2015-12-18Added option to specify a maximum file size when recording an AEC dump.ivoc
For applications with a strict filesize limit for debug files, I added an option to specify a maximum filesize for AEC dumps. An existing unit test is extended to check that the feature works as advertised. BUG=webrtc:4741 TBR=glaznev@webrtc.org Review URL: https://codereview.webrtc.org/1413483003 Cr-Commit-Position: refs/heads/master@{#11081}
2015-12-18Use NV21 instead of YUV12 and clean up.perkj
BUG=webrtc:5375 Review URL: https://codereview.webrtc.org/1530843002 Cr-Commit-Position: refs/heads/master@{#11079}
2015-12-18MediaCodecVideoEncoder, set timestamp on the encoder surface when drawing a ↵perkj
texture. BUG=webrtc:4993 Review URL: https://codereview.webrtc.org/1523843006 Cr-Commit-Position: refs/heads/master@{#11078}
2015-12-18Fix build break in google3 import caused by ↵guoweis
https://codereview.webrtc.org/1532543003 TBR=pthatcher@webrtc.org BUG= Review URL: https://codereview.webrtc.org/1537683003 Cr-Commit-Position: refs/heads/master@{#11076}
2015-12-18DTLS-SRTP set up is bypassed when the channel has been writable.guoweis
This regression was introduced by CL 1505573002 to support remote fingerprint update. What happened is that during PrAnswer, we incorrectly do not apply bundle. However, the channel has become writable at that time. When Answer comes, we still reset the srtp_filter but since the channel has been writable, the new SRTP context has never been applied. We're making sure that we could always apply SRTP context even when channel has been writable. We'll address the issue that bundle should apply even in PrAnswer in a different CL. BUG=568734 Review URL: https://codereview.webrtc.org/1532543003 Cr-Commit-Position: refs/heads/master@{#11075}
2015-12-17Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and ↵kwiberg
rtc::ScopedVector We can now use std::move instead! This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them. Review URL: https://codereview.webrtc.org/1460043002 Cr-Commit-Position: refs/heads/master@{#11064}
2015-12-17Add ufrag to the ICE candidate signaling.honghaiz
On the receiving side, if a candidate arrives with an old ufrag, it will be dropped. If it contains a new frag that has never seen before, it will hold the ufrag and create connections, although those connections are not pingable until the ICE credentials are received. This could avoid a bunch of ICE generation issues. BUG=webrtc:5138,webrt:5292 Review URL: https://codereview.webrtc.org/1498993002 Cr-Commit-Position: refs/heads/master@{#11060}
2015-12-16Disable warnings failing when using Clang on Windows.kjellander
This makes it possible to build WebRTC using Clang on Windows. Depends on https://codereview.webrtc.org/1524703006/ BUG=webrtc:5360, webrtc:5366 NOTRY=True Review URL: https://codereview.webrtc.org/1522223002 Cr-Commit-Position: refs/heads/master@{#11058}
2015-12-16Fix error prone code in VideoCapturerAndroidperkj
BUG=webrtc:5282 Review URL: https://codereview.webrtc.org/1486423003 Cr-Commit-Position: refs/heads/master@{#11046}
2015-12-16Bugfix that fixes the error where the audio processing module is calledpeah
using the wrong sample rate for the render signal. The CL is basically a partial revert of the related changes done on output_mixer.cc in the CL https://codereview.webrtc.org/1234463003. The CL also reverts the removal of the input_sample_rate_hz() method that was removed as part of the CL https://codereview.webrtc.org/1379123002 (as it was at that point no longer used). It should be noted that this CL turns off the effect of the IntelligibilityEnhancer when the AudioFrame AudioProcessing APIs are used. While it may be possible to solve that by adding upsampling after the API call, that approach was discarded due to that: -That would add extra processing in the echo path, leading to possible AEC performance reduction. -That would add extra complexity for the mobile case. -That would only patch the intelligibility enhancer operation as the proper way to do such an operation is within APM. -The intelligibility enhancer is not active by default anywhere. BUG=webrtc:5237 Review URL: https://codereview.webrtc.org/1525173002 Cr-Commit-Position: refs/heads/master@{#11045}
2015-12-16Restoring behavior where PeerConnection tracks changes to MediaStreams.deadbeef
If a MediaStream is added to a PeerConnection, and later a track is added to the MediaStream, a new RtpSender will now be created for that track, and it will appear in subsequent offers. Similarly, removed tracks will remove RtpSenders. BUG=webrtc:5265 Review URL: https://codereview.webrtc.org/1507973003 Cr-Commit-Position: refs/heads/master@{#11040}
2015-12-16Fixing bug where "mid" wasn't preserved across re-offers.deadbeef
Review URL: https://codereview.webrtc.org/1529673002 Cr-Commit-Position: refs/heads/master@{#11039}
2015-12-15Android: Refactor renderers to allow apps to inject custom shadersMagnus Jedvert
This CL: * Abstracts the functions in GlRectDrawer to an interface. * Adds viewport location as argument to the draw() functions, because this information may be needed by some shaders. This also moves the responsibility of calling GLES20.glViewport() to the drawer. * Moves uploadYuvData() into a separate helper class. * Adds new SurfaceViewRenderer.init() function and new VideoRendererGui.create() function that takes a custom drawer as argument. Each YuvImageRenderer in VideoRendererGui now has their own drawer instead of a common one. BUG=b/25694445 R=nisse@webrtc.org, perkj@webrtc.org Review URL: https://codereview.webrtc.org/1520243003 . Cr-Commit-Position: refs/heads/master@{#11031}
2015-12-15Disable transport sequence numbers for audio.Stefan Holmer
Since this isn't fully wired up yet it shouldn't be part of the SendSideBwe experiment yet. BUG=webrtc:5263 R=pbos@webrtc.org Review URL: https://codereview.webrtc.org/1523283002 . Cr-Commit-Position: refs/heads/master@{#11029}
2015-12-15Add a 'remote' property to MediaSourceInterface. Also adding an ↵tommi
implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :( Additionally: * Moving all implementation inside RemoteAudioTrack into AudioTrack and remove RemoteAudioTrack. * AddSink/RemoveSink are now on all audio sources (like they are for video sources). While doing this I found that some of our tests are broken :) and fixed them. They were broken because AudioTrack didn't previously do much such as updating its state. BUG=chromium:569526 Review URL: https://codereview.webrtc.org/1522903002 Cr-Commit-Position: refs/heads/master@{#11026}
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