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}
CodecOwner is introduced here; AudioEncoderMutable was introduced in a
previous commit, but had no users until now. The only remaining task
for ACMGenericCodec was to construct and maintain the stack of speech,
CNG, and RED encoders. This task is now handled by the CodecOwner,
which is owned and used by the CodecManager.
COAUTHOR=henrik.lundin@webrtc.org
BUG=4228
R=jmarusic@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43189004
Cr-Commit-Position: refs/heads/master@{#9152}
The Mac64 Debug builder is broken for an unknown failure (trybot is
green, no failure obvious in the commit break). Reverting this CL to see
if it goes green again, and then relanding to see if it is just some
weird flaky build issue.
This reverts commit 5ea8eff55ec21a1d81aaf7d29c0106fe13256150.
BUG=
TBR=rollback
Review URL: https://webrtc-codereview.appspot.com/47019004
Cr-Commit-Position: refs/heads/master@{#9074}
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.
BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45549004
Cr-Commit-Position: refs/heads/master@{#8864}
The current way that iSAC RCU is packetized and sent as a RED packet,
with the same payload type for primary and redundant payloads, does
not follow the specification for RED. As it is now, it is impossible
for a receiver to know if an incoming RED packet with iSAC payloads
inside consists of two "primary" (but time-shifted) payloads, or one
primary and one RCU payload. The RED standard stipulates that the
former option is the correct interpretation, while our implementation
currently applies the latter.
This CL removes support for iSAC RCU from Audio Coding Module, but
leaves it in the iSAC codec itself (i.e., in the C implementation).
BUG=4402
COAUTHOR=kwiberg@webrtc.orgR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45569004
Cr-Commit-Position: refs/heads/master@{#8713}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8713 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL adds support for unittests of the AudioDeviceModule on Android using both Java and C++. The new framework uses ::testing::TesWithParam to support both Java-based audio and OpenSL ES based audio. However, given existing issues in our OpenSL ES implementation, the list of test parameters only contains Java in this first version. Open SL ES will be enabled as soon as the backend has been refactored.
It also:
- Removes the redundant JNIEnv* argument in webrtc::VoiceEngine::SetAndroidObjects().
- Modifies usage of enable_android_opensl and the WEBRTC_ANDROID_OPENSLES define.
- Adds kAndroidJavaAudio and kAndroidOpenSLESAudio to AudioLayer enumerator.
- Fixes some bugs which were discovered when running the tests.
BUG=NONE
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40069004
Cr-Commit-Position: refs/heads/master@{#8651}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8651 4adac7df-926f-26a2-2b94-8c16560cd09d
Mostly, it's about moving constructors and descructors to the .cc
files, so that they won't be inlined everywhere.
The reason this CL is so big is that a lot of code was using
common_types.h without declaring a dependency on webrtc_common, which
broke the build once common_types.h started to depend on
common_types.cc.
BUG=163
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26089004
Cr-Commit-Position: refs/heads/master@{#8516}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
