NetEq network statistics contains discard rate but has not been used and even not been implemented until recently.
According to w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsdiscarded,
this statistics needs to be replaced with an accumulative stats. Such work will be carried out separately.
Meanwhile, we need to add a rate to reflect rate of discarded redundant packets. See webrtc:8025.
In this CL, we replace the existing discard rate with secondary discarded rate, so as to
1. fulfill the requests on webrtc:8025
2. get ready to implement an accumulative statistics for discarded packets.
BUG: webrtc:7903,webrtc:8025
Change-Id: Idbf143a105db76ca15f0af54848e1448f2a810ec
Reviewed-on: https://chromium-review.googlesource.com/582863
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19495}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
The network stats used to be polled from the NetEq object once at the
very end of the simulation. With this change, the stats are polled
once every second, and then aggregated at the end of the run. This
leads to more meaningful numbers.
Bug: webrtc:2692
Change-Id: I9e0f4ddada2f9e42fb9234970deb1af235fffc8c
Reviewed-on: https://chromium-review.googlesource.com/541218
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18682}
This change adds an option to have neteq_rtpplay generate a Matlab
script. When executed in Matlab, the script will generate graphs with
the timing information from the test run.
The script is generated when the flag --matlabplot is passed to
neteq_rtpplay.
The CL also adds better checking and reporting about packets discarded
in the process of finding out the initial sampling rate.
Bug: webrtc:2692, webrtc:7467
Change-Id: I805e7c83b82533142b6e74bf065506e3d60a8170
Reviewed-on: https://chromium-review.googlesource.com/541276
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18680}
Imprecisions in floating point representation caused noise in the
graphs. The integer division is in fact exact.
BUG= webrtc:7467
Review-Url: https://codereview.webrtc.org/2933053002
Cr-Commit-Position: refs/heads/master@{#18592}
This CL adds the capability to analyze and plot how NetEq behaves in
response to a network trace.
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2876423002
Cr-Commit-Position: refs/heads/master@{#18590}
LastDecoderError was only used in tests. LastError was only used in
conjunction with RemovePayloadType, and always to distinguish between
"decoder not found" and "other error". In AcmReceiver, "decoder not
found" was not treated as an error.
With this change, calling NetEq::RemovePayloadType with a payload type
that is not registered is no longer considered to be an error. This
allows to rewrite the code in AcmReceiver, such that it no longer has
to call LastError.
The internal member variables NetEqImpl::error_code_ and
NetEqImpl::decoder_error_code_ are removed, since they were no longer
read.
Bug: none
Change-Id: Ibfe97265954a2870c3caea4a34aac958351d7ff1
Reviewed-on: https://chromium-review.googlesource.com/535533
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18588}
const int16_t* data() const;
int16_t* mutable_data();
- data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames.
- mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_.
These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation.
This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later.
BUG=webrtc:7343
TBR=henrika
Review-Url: https://codereview.webrtc.org/2750783004
Cr-Commit-Position: refs/heads/master@{#18543}
A recent change (https://codereview.webrtc.org/2855143002/) introduced
a bug in RtcEventLogSource::NextPacket(). The rtp_packet_index_ must
be incremented when a valid packet is found and delivered. Otherwise,
the same packet will be delivered over and over again.
The recent change also altered the way that audio packets are sifted out. Now, the RTP header is always parsed before discarding any non-audio packets. This means that RtpHeaderParser::Parse is always called, also with video packets, which sometimes contain padding. When header-only dumps (such as RtcEventLogs) are created, the payload is stripped, and the payload length is equal to
the RTP header length. However, if the original packet was padded, the
RTP header will carry information about this padding length, and the
parser will check that the pyaload length is at least the header +
padding. This is not the case for header-only dumps, and the parser will return an error. In this CL, we ignore that error when a header-only packet has padding length larger than 0.
BUG=webrtc:7538
Review-Url: https://codereview.webrtc.org/2912323003
Cr-Commit-Position: refs/heads/master@{#18385}
Biggest change is to Remove MediaType as argument to RtcEventLog::LogRtpHeader and RtcEventLog::LogRtcpHeader.
Since the type is used by tools, these tools are rewritten to figure out the media type from the configurations instead.
BUG=webrtc:7538
TBR=solenberg@webrtc.org // For call.cc and voiceengine.cc
Review-Url: https://codereview.webrtc.org/2855143002
Cr-Commit-Position: refs/heads/master@{#18324}
This change allows more callbacks to be registered to the test object.
The callbacks are used to give the user of the test object the ability
to instrument the test object. This CL specifically adds
instrumentation points just after a packet is inserted into NetEq, and
just after audio is pulled out of NetEq.
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2851383004
Cr-Commit-Position: refs/heads/master@{#18014}
Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2799033006
Cr-Commit-Position: refs/heads/master@{#17977}
This new method prints information about the packet.
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2844283002
Cr-Commit-Position: refs/heads/master@{#17922}
In this CL, the neteq_unittest_tools target is split in two separate
targets. One still called neteq_tools which does not set
testonly=true and that includes code related to audio input,
replacement audio and fake decoding. The other target called
neteq_test_tools contains the remaining files, and is
still under testonly=true.
Other renames:
neteq_test_tools -> neteq_test_tools_deprecated
neteq_test_minimal -> neteq_tools_minimal
Cyclic dependencies were also cleaned up.
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng
BUG=webrtc:7467,webrtc:6828
Review-Url: https://codereview.webrtc.org/2845013003
Cr-Commit-Position: refs/heads/master@{#17921}
This is to be used in tests where the audio output is not interesting.
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2842033003
Cr-Commit-Position: refs/heads/master@{#17893}
With this fix, the size of the fake encoded payload is limited to 120
ms at 48000 samples/second.
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2838353002
Cr-Commit-Position: refs/heads/master@{#17891}
With this CL, all tests and tools under the neteq/ folder are
converted to use RTPHeader instead of WebRtcRTPHeader. WebRtcRTPHeader
has an RTPHeader as a member. None of the other member in
WebRtcRTPHeader where used.
TBR=kjellander@webrtc.org
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2809153002
Cr-Commit-Position: refs/heads/master@{#17845}
Reason for revert:
Downstream roadblock should be cleared by now. Relanding original patch.
Original issue's description:
> Revert of Change NetEq::InsertPacket to take an RTPHeader (patchset #2 id:20001 of https://codereview.webrtc.org/2807273004/ )
>
> Reason for revert:
> Broke downstream dependencies.
>
> Original issue's description:
> > Change NetEq::InsertPacket to take an RTPHeader
> >
> > It used to take a WebRtcRTPHeader as input, which has an RTPHeader as
> > a member. None of the other member in WebRtcRTPHeader where used in
> > NetEq.
> >
> > This CL adapts the production code; tests and tools will be converted
> > in a follow-up CL.
> >
> > BUG=webrtc:7467
> >
> > Review-Url: https://codereview.webrtc.org/2807273004
> > Cr-Commit-Position: refs/heads/master@{#17652}
> > Committed: 4d027576a6
>
> TBR=ivoc@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7467
>
> Review-Url: https://codereview.webrtc.org/2812933002
> Cr-Commit-Position: refs/heads/master@{#17657}
> Committed: 10d095d4f7R=ivoc@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2835093002 .
Cr-Commit-Position: refs/heads/master@{#17843}
Reason for revert:
Broke downstream dependencies.
Original issue's description:
> Change NetEq::InsertPacket to take an RTPHeader
>
> It used to take a WebRtcRTPHeader as input, which has an RTPHeader as
> a member. None of the other member in WebRtcRTPHeader where used in
> NetEq.
>
> This CL adapts the production code; tests and tools will be converted
> in a follow-up CL.
>
> BUG=webrtc:7467
>
> Review-Url: https://codereview.webrtc.org/2807273004
> Cr-Commit-Position: refs/heads/master@{#17652}
> Committed: 4d027576a6TBR=ivoc@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2812933002
Cr-Commit-Position: refs/heads/master@{#17657}
It used to take a WebRtcRTPHeader as input, which has an RTPHeader as
a member. None of the other member in WebRtcRTPHeader where used in
NetEq.
This CL adapts the production code; tests and tools will be converted
in a follow-up CL.
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2807273004
Cr-Commit-Position: refs/heads/master@{#17652}
It's the faster, less strict cousin of checked_cast.
BUG=none
Review-Url: https://codereview.webrtc.org/2714063002
Cr-Commit-Position: refs/heads/master@{#16958}
There's no longer any need to make the two arguments have the same
signedness, so we can drop the "u" suffix on literal integer
arguments.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2535593002
Cr-Commit-Position: refs/heads/master@{#15280}
The original CL (https://codereview.webrtc.org/2315633002) was
reverted since the fuzzer depended on gflags and files in the
resources folder; neither of this is allowed for a fuzzer test in
Chromium. This new version streamlines the dependencies, and changes
the test to generate a sinusoid input audio signal instead of reading
from a file.
Original commit message:
This CL introduces a new fuzzer target neteq_rtp_fuzzer that
manipulates the RTP header fields before inserting the packets into
NetEq. A few helper classes are also introduced.
BUG=webrtc:5447
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device
Review-Url: https://codereview.webrtc.org/2384423002
Cr-Commit-Position: refs/heads/master@{#14523}
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
Reason for revert:
Broke all Chromium libFuzzer builds
https://bugs.chromium.org/p/chromium/issues/detail?id=645069
Original issue's description:
> Setting up an RTP input fuzzer for NetEq
>
> This CL introduces a new fuzzer target neteq_rtp_fuzzer that
> manipulates the RTP header fields before inserting the packets into
> NetEq. A few helper classes are also introduced.
>
> BUG=webrtc:5447
> NOTRY=True
>
> Committed: https://crrev.com/2d273f1e97cd5030ed1686f27ce1118291b66395
> Cr-Commit-Position: refs/heads/master@{#14103}
TBR=ivoc@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5447
Review-Url: https://codereview.webrtc.org/2328483002
Cr-Commit-Position: refs/heads/master@{#14131}
With this CL, the NetEqReplacementInput class handles reordered and
missing packets in a better way than before, by storing the last
confirmed packet size and using that when the next packet size cannot
be calculated.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2319553003
Cr-Commit-Position: refs/heads/master@{#14122}
This CL introduces a new fuzzer target neteq_rtp_fuzzer that
manipulates the RTP header fields before inserting the packets into
NetEq. A few helper classes are also introduced.
BUG=webrtc:5447
NOTRY=True
Review-Url: https://codereview.webrtc.org/2315633002
Cr-Commit-Position: refs/heads/master@{#14103}
If neteq_rtpplay is invoked with the --ssrc option to select packets
matching a specific SSRC, but no matching packets are found, this CL
provides a meaningful error message.
BUG=webrtc:2692
NOTRY=True
TBR=ivoc@webrtc.org
Review-Url: https://codereview.webrtc.org/2318503002
Cr-Commit-Position: refs/heads/master@{#14083}
This adds a new file, webrtc/modules/audio_coding/neteq/tools/packet_source.cc, so that I'll have somewhere to put the new non-inlined methods.
NOTRY=true
BUG=webrtc:163
Review-Url: https://codereview.webrtc.org/2290593002
Cr-Commit-Position: refs/heads/master@{#13956}
This removes the warning printouts about unknown header extensions.
BUG=webrtc:2692
Review-Url: https://codereview.webrtc.org/2266403005
Cr-Commit-Position: refs/heads/master@{#13912}
This implementation interprets payloads of size 1 as codec-internal SID
frames, marking the start of a CNG period. Changes were made to other
parts of the test payload chain, since it had to make use of the virtual
payload size in the case of header-only RTP files.
BUG=webrtc:2692
Review-Url: https://codereview.webrtc.org/2275903002
Cr-Commit-Position: refs/heads/master@{#13901}
This change is a major refactoring of the neteq_rtpplay tool. It
consists of the following parts:
- NetEqTest class: Breaks out the main simulation loop from
neteq_rtpplay into a separate class with well defined inputs and
outputs.
- NetEqInput: Interface class for the input to NetEqTest.
- NetEqPacketSourceInput: Implementation of NetEqInput that provides a
PacketSource objects with a NetEqInput interface. This has two
subclasses; one for RtpFileSource and one for RtcEventLogSource.
- NetEqReplacementInput: An object that modifies the packets provided by
another NetEqInput object, and replaces the packet payloads with meta
data readable by a FakeDecodeFromFile decoder.
- FakeDecodeFromFile: An AudioDecoder implementation that produces
"decoded" data by reading from an audio file.
BUG=webrtc:2692, webrtc:5447
Review-Url: https://codereview.webrtc.org/2020363003
Cr-Commit-Position: refs/heads/master@{#13252}
This allows us to get rid of the function that computes it, which gets
us one step closer to getting rid of the NetEqDecoder type.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2021063002
Cr-Commit-Position: refs/heads/master@{#12974}
A regression happened in https://codereview.webrtc.org/2006723002,
causing neteq_rtpplay not to work. The problem was that when the main
code was moved inside of the webrtc::test namespace, it was no longer
visible to the linker. Meanwhile, the dependency on test_support_main
rather than test_support caused the executable to be a gtest.
In this fix, the gyp dependencies are corrected, and a main method is
added outside of the namespaces.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2018473002
Cr-Commit-Position: refs/heads/master@{#12918}
Channel's API remains unchanged, but the creation of a BuiltinAudioDecoderFactory is now in Channel. The next step would be to amend Channel's API (through CreateChannel, I believe) to allow an AudioDecoderFactory to be sent along.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1992763002
Cr-Commit-Position: refs/heads/master@{#12893}
The return type of PacketSource::NextPacket() is changed from a naked
pointer to an std::uniqe_ptr. The interface contract was and still is
that the ownership is passed from the callee to the caller, but a
unique_ptr makes this explicit.
BUG=webrtc:2692
Review-Url: https://codereview.webrtc.org/2005873002
Cr-Commit-Position: refs/heads/master@{#12884}
Changed rtpdump converter and neteq tool to use new parser, but still aborting if the file is corrupt.
Review-Url: https://codereview.webrtc.org/1768773002
Cr-Commit-Position: refs/heads/master@{#12714}
