We have decided not to do a switch from old (AudioCodingModule) to new
(AudioCoding) API. Instead, we will gradually evolve the old API to
meet the new design goals.
As a consequence of this decision, the AudioCoding and AudioCodingImpl
classes are deleted. Also removing associated unit test sources. No
test coverage is lost with this operation, since the tests for the
"old" API are testing more than the deleted tests did.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1415163002
Cr-Commit-Position: refs/heads/master@{#10406}
Our perf test suite webrtc_perf_tests timed out, which caused most
of the delay landing this (https://crbug.comn/535973 and
https://codereview.chromium.org/1370133004).
Other problems with executing Android tests also needed to be
resolved in order to land this (http://crbug.com/534849).
Libvpx has moved from third_party/libvpx to third_party/libvpx_new
as of https://codereview.chromium.org/1323333002/
Android GN was blocking this roll due to a problem that ended up
being caused by a bug (http://crbug.com/534849).
Relevant changes:
* src/buildtools: f7310ee..8d89c1b
* src/third_party/boringssl/src: 1d128f3..4c60d35
* src/third_party/icu: 6b3ce81..423fc7e
* src/third_party/libjpeg_turbo: 631e2dd..e4e7503
* src/third_party/libvpx: ac1772e..70db223
* src/third_party/libyuv: fcacbfb..62c49dc
* src/tools/gyp: 5d01a8c..01528c7
* src/tools/swarming_client: 77f720b..6e5d2b2
Details: 310ea93..8cf53d6/DEPS
Clang version changed 245965:247874
Details: 310ea93..8cf53d6/tools/clang/scripts/update.sh
BUG=481034, 535973
TBR=marpan@webrtc.org
Review URL: https://codereview.webrtc.org/1355083002
Cr-Commit-Position: refs/heads/master@{#10101}
This updates the isolate.gypi copies we have to maintain in our
code repo to Chromium's revision 310ea93.
The changes about generating .isolated.gen.json files are needed
to support running with Swarming (https://www.chromium.org/developers/testing/isolated-testing)
Since isolated testing is now using a new launch script
in tools: isolate_driver.py, that's added to our links
script.
In order to use isolate_driver.py, the .isolate files must be in the
same directory as the test_name_run target is defined, which meant
I had to move around some of the isolate files and targets below
webrtc/modules.
BUG=497757
R=maruel@chromium.orgTBR=henrik.lundin@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org
TESTED=Clobbered trybots:
git cl try -c --bot=linux_compile_rel --bot=mac_compile_rel --bot=win_compile_rel --bot=android_compile_rel --bot=ios_rel -m tryserver.webrtc
Review URL: https://codereview.webrtc.org/1373513002 .
Cr-Commit-Position: refs/heads/master@{#10081}
When using send-side bandwidth estimation, the inter-packet delay is
reported back to the sender using RTCP TransportFeedback messages.
Theis data needs to be translated into a format which the bandwidth
estimator (now instantiated on the send side) can use, including looking
up the local absolute send time from the send time history.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1329083005
Cr-Commit-Position: refs/heads/master@{#9929}
Merge the contents of audio_encoder_mutable_opus_test.cc into
audio_encoder_opus_unittest.cc, since they're now both testing
AudioEncoderOpus.
(While preparing this CL, I noted a bug in the PacketLossRateOptimized
test. This CL leaves that test essentially unchanged; I've posted bug
4981 about the problem.)
Review URL: https://codereview.webrtc.org/1319713004
Cr-Commit-Position: refs/heads/master@{#9906}
For use when send-side bandwidth estimation is enabled.
Receive times need to be captured, buffered and then sent using
TransportFeedback RTCP messaged back to the send side.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1290813008
Cr-Commit-Position: refs/heads/master@{#9898}
This CL contains major modifications of the audio output parts for WebRTC on iOS:
- general code cleanup
- improves thread handling (added thread checks, remove critical section, atomic ops etc.)
- reduces loopback latency of iPhone 6 from ~90ms to ~60ms ;-)
- improves selection of audio parameters on iOS
- reduces complexity by removing complex and redundant delay estimates
- now instead uses fixed delay estimates if for some reason the SW EAC must be used
- adds AudioFineBuffer to compensate for differences in native output buffer size and
the 10ms size used by WebRTC. Same class as is used today on Android and we have unit tests for
this class (the old code was buggy and we have several issue reports of crashes related to it)
Similar improvements will be done for the recording sid as well in a separate CL.
I will also add support for 48kHz in an upcoming CL since that will improve Opus performance.
BUG=webrtc:4796,webrtc:4817,webrtc:4954, webrtc:4212
TEST=AppRTC demo and iOS modules_unittests using --gtest_filter=AudioDevice*
R=pbos@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1254883002 .
Cr-Commit-Position: refs/heads/master@{#9875}
Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
Re-lands "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module."
This reverts commit b933667a7f97697d6390d1eee5f378cedd9ca208.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1259683003 .
Cr-Commit-Position: refs/heads/master@{#9661}
Computing all metrics using constant extra memory.
PlotHistogram methods are executed in constant time.
-- Previously throughput and delay were using O(num_packets) extra memory and their associated PlotHistograms took linear time complexity.
Added MetricRecorder unittests
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1257683006 .
Cr-Commit-Position: refs/heads/master@{#9658}
Placed the protobuf structures in the namespace webrtc::rtclog. Removed the message field from the DebugEvent structure, since it was not used.
Added an interface to set config information for VideoReceiveStream and VideoSendStream in the event log.
Added function to log full RTCP packets and changed RTP-logging to only log headers.
Significantly extended the unit tests for RtcEventLog.
R=ivoc@webrtc.org, minyue@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1230973005 .
Cr-Commit-Position: refs/heads/master@{#9656}
---- Dynamic receiving rate.
---- Dynamic packet-loss.
---- Dynamic objective function.
---- Dynamic available capacity.
---- Dynamic available capacity per flow.
---- Average delay Histogram with standard deviation or 5th/95th percentiles.
---- Average bitrate Histogram with error bars.
---- Optimal average bitrate dashed line.
---- Average packet-loss Histogram.
---- Total objective function Histogram.
Added media Pause/Resume methods to Video and TcpSender.
Modified LinkedSet: computing GlobalPacketLossRatio even if packet's sequence_number overflows.
Added small randomization to frame send times, modified bwe_test_framework_unittest accordingly.
Taking offset time into account for plotting.
Added nada_unittests.
Added bwe_unittests.
Added a RateCounter to BweReceiver (replaced ReceivingRate)
Added LossAccount.
Fixed NadaBweReceiver issue: using sender_timestamp instead of creation_time.
Fixed memory leaks.
Fixed int division rounding issues.
Supporting plots on bandwidth Estimators:
Logging received packet information on on SubClassesBweReceiver::ReceivePacket
Updating RateCounter, updating packet loss account and relieving LinkedSet when necessary.
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1202253003 .
Cr-Commit-Position: refs/heads/master@{#9585}
Adds a class used to classify whether packet loss events are a single packet or multiple packets as well as how many packets have been lost. Also exposes a new function in the RtpRtcp interface to retrieve these statistics.
BUG=
Review URL: https://codereview.webrtc.org/1198853004
Cr-Commit-Position: refs/heads/master@{#9568}
- New option for computing variance that is more adaptive with lower complexity.
- Fixed related off-by-one errors.
- Added intelligibility unittests.
- Do not enhance if experiencing variance underflow.
R=andrew@webrtc.org, henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1207353002 .
Cr-Commit-Position: refs/heads/master@{#9567}
They make it possible to send bandwidth estimation info from decoder
to encoder even if they are separate objects (which we want them to be
because multithreading).
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1208923002.
Cr-Commit-Position: refs/heads/master@{#9535}
Before this change, it could happen that a caller would get a pointer
to the encoder_ but not use it before another thread called the
Reconstruct method, changing the pointer. This of course resulted in
bad access crashes. With this change, each use of the pointer acquired
from the encoder() method is protected by the same lock that is
required to update the pointer. Note that this fix is probably too
aggressive, since it also affects the Opus implementation; the crash
has so far only been seen for iSAC.
Also adding a test to trigger the problem. The test did not trigger
the problem deterministically, but out would typically find it in less
than 1000 runs.
BUG=chromium:499468
R=jmarusic@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1176303004.
Cr-Commit-Position: refs/heads/master@{#9436}
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
This adds functionality from audioproc_float. The geometry parsing code
is now shared from test_utils.h. I removed the "mic_spacing" flag from
audioproc_float because it's a redundancy that I suspect isn't very
useful.
Includes a cleanup of the audio_processing test utils. They're now
packaged in targets, with the protobuf-using ones split out to avoid
requiring users to depend on protobufs.
pcm_utils is no longer needed and removed.
The primary motivation for this CL is that AudioProcessing currently
doesn't support more than two channels and we'd like a way to pass
more channels to the beamformer.
R=aluebs@webrtc.org, mgraczyk@chromium.org
Review URL: https://webrtc-codereview.appspot.com/50899004
Cr-Commit-Position: refs/heads/master@{#9157}
