These tests will be reenabled and updated after Opus has been updated in
Chromium and rolled into WebRTC.
BUG=737323, webrtc:8024
Review-Url: https://codereview.webrtc.org/2963673002
Cr-Commit-Position: refs/heads/master@{#19118}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
PlayoutFrequency(), at least, is called ~200 times a second. The others
appear to not be in practice, but it's unclear what value they serve.
They were traces before https://chromium-review.googlesource.com/c/518133/,
which was more reasonable, as you could enable them for just audio
coding traces. But now that they are just logs, they make all VERBOSE
logging unusable.
Bug: webrtc:7959
Change-Id: I190a61c8ff4c0f047798087e80adbb41d791fc29
Reviewed-on: https://chromium-review.googlesource.com/563881
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Noah Richards <noahric@chromium.org>
Cr-Commit-Position: refs/heads/master@{#18956}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2973183002
Cr-Commit-Position: refs/heads/master@{#18948}
Prior to https://codereview.webrtc.org/2750783004/ Reset() intentionally
did not zero out the buffer. After that change, callers calling Reset()
and then mutable_data() were performing a wasteful zeroing.
This change adds ResetWithoutMuting() to match the old behavior and
switches the sole non-test caller of Reset() to use ResetWithoutMuting()
instead.
Prior to this change (optimized, Linux):
$ out/Default/webrtc_perf_tests --gtest_filter=NetEqPerformanceTest.Run* \
--gtest_repeat=10 | grep neteq_performance
*RESULT neteq_performance: 10_pl_10_drift= 4051 ms
*RESULT neteq_performance: 0_pl_0_drift= 1768 ms
*RESULT neteq_performance: 10_pl_10_drift= 3666 ms
*RESULT neteq_performance: 0_pl_0_drift= 1690 ms
*RESULT neteq_performance: 10_pl_10_drift= 3685 ms
*RESULT neteq_performance: 0_pl_0_drift= 1693 ms
*RESULT neteq_performance: 10_pl_10_drift= 3720 ms
*RESULT neteq_performance: 0_pl_0_drift= 1690 ms
*RESULT neteq_performance: 10_pl_10_drift= 3780 ms
*RESULT neteq_performance: 0_pl_0_drift= 1728 ms
*RESULT neteq_performance: 10_pl_10_drift= 3733 ms
*RESULT neteq_performance: 0_pl_0_drift= 1737 ms
*RESULT neteq_performance: 10_pl_10_drift= 3781 ms
*RESULT neteq_performance: 0_pl_0_drift= 1744 ms
*RESULT neteq_performance: 10_pl_10_drift= 3712 ms
*RESULT neteq_performance: 0_pl_0_drift= 1731 ms
*RESULT neteq_performance: 10_pl_10_drift= 3681 ms
*RESULT neteq_performance: 0_pl_0_drift= 1691 ms
*RESULT neteq_performance: 10_pl_10_drift= 3681 ms
*RESULT neteq_performance: 0_pl_0_drift= 1690 ms
With this change:
$ out/Default/webrtc_perf_tests --gtest_filter=NetEqPerformanceTest.Run* \
--gtest_repeat=10 | grep neteq_performance
*RESULT neteq_performance: 10_pl_10_drift= 3824 ms
*RESULT neteq_performance: 0_pl_0_drift= 1632 ms
*RESULT neteq_performance: 10_pl_10_drift= 3502 ms
*RESULT neteq_performance: 0_pl_0_drift= 1521 ms
*RESULT neteq_performance: 10_pl_10_drift= 3520 ms
*RESULT neteq_performance: 0_pl_0_drift= 1534 ms
*RESULT neteq_performance: 10_pl_10_drift= 3517 ms
*RESULT neteq_performance: 0_pl_0_drift= 1530 ms
*RESULT neteq_performance: 10_pl_10_drift= 3521 ms
*RESULT neteq_performance: 0_pl_0_drift= 1527 ms
*RESULT neteq_performance: 10_pl_10_drift= 3511 ms
*RESULT neteq_performance: 0_pl_0_drift= 1533 ms
*RESULT neteq_performance: 10_pl_10_drift= 3518 ms
*RESULT neteq_performance: 0_pl_0_drift= 1523 ms
*RESULT neteq_performance: 10_pl_10_drift= 3503 ms
*RESULT neteq_performance: 0_pl_0_drift= 1524 ms
*RESULT neteq_performance: 10_pl_10_drift= 3514 ms
*RESULT neteq_performance: 0_pl_0_drift= 1534 ms
*RESULT neteq_performance: 10_pl_10_drift= 3501 ms
*RESULT neteq_performance: 0_pl_0_drift= 1530 ms
BUG=webrtc:7343,chromium:738852,chromium:738839
Change-Id: Idcbb276ca0ed27fff95164a73f1c1fa310175ee5
Reviewed-on: https://chromium-review.googlesource.com/563021
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18939}
In practice, this change will make AudioFrame::muted_ replicate the
explicit muted variable, passed as a pointer to NetEq::GetAudio.
BUG=webrtc:7944
Review-Url: https://codereview.webrtc.org/2965203002
Cr-Commit-Position: refs/heads/master@{#18914}
This reverts commit 2d54784d890be462a7fbf0fcfdc633bc4791982a.
Reason for revert: upstream conflicts
Original change's description:
> Reland "Adding ANA config event to debug dump."
>
> Originally review in https://chromium-review.googlesource.com/c/535554/
>
> Reverted in https://chromium-review.googlesource.com/c/539737/ due to upstreaming failure.
>
> BUG=webrtc:7854
>
> Change-Id: Ie4ad6ecfaf0f6b556dc662512d0be8ce94f8a4a8
> Reviewed-on: https://chromium-review.googlesource.com/541436
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18865}
TBR=minyue@webrtc.org,ossu@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:7854
Change-Id: I28841ed088664d2965454dc52196f83c9d81773e
Reviewed-on: https://chromium-review.googlesource.com/559429
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18904}
The implementation of this method did not follow the description in
the method comment. It was supposed to delete all packets in a range
[A, B], but if at least one packet in the buffer had a timestamp lower
than A, then no packets at all were discarded. This is now fixed.
BUG=webrtc:7937
Review-Url: https://codereview.webrtc.org/2969123003
Cr-Commit-Position: refs/heads/master@{#18903}
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}
(This got reverted because of a problem with the Opus encoder parts.
Re-landing without changes.)
BUG=webrtc:7837
Review-Url: https://codereview.webrtc.org/2950453002
Cr-Commit-Position: refs/heads/master@{#18855}
This was previously reverted, because external projects were using the
internal webrtc::AudioEncoderOpus class and broke when it was renamed.
This re-land avoids renaming it immediately, to give those projects
time to adapt. It also has to revert some of the changes I had made to the
Config struct, since that was also used by the same external projects.
BUG=webrtc:7831
Review-Url: https://codereview.webrtc.org/2948483002
Cr-Commit-Position: refs/heads/master@{#18852}
Some .proto files have newline at the end. This CL levels all our .proto
files. A presubmit check will follow.
NOTRY=True
TBR=minyue@webrtc.org
Bug: None
Change-Id: I988fe94c31abf91c85a45b564c488329d677b958
Reviewed-on: https://chromium-review.googlesource.com/552137
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18823}
We'd like to remove all occurrences of WEBRTC_TRACE and delete the
macro! One logging mechanism is enough.
NOTRY=True
Bug: webrtc:5118
Change-Id: Ic226318e0aebe3a71785fcb4ce07371872ab7128
Reviewed-on: https://chromium-review.googlesource.com/518133
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18712}
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}
BUG=webrtc:7854
Change-Id: I12c33b8558fd49374a55282c391b87fde9e13a28
Reviewed-on: https://chromium-review.googlesource.com/535554
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Michael T <tschumim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18661}
Reason for revert:
Breaking google3 projects
Original issue's description:
> Opus implementation of the AudioEncoderFactoryTemplate API
>
> Now the templated AudioEncoderFactory can create Opus encoders!
>
> BUG=webrtc:7831
>
> Review-Url: https://codereview.webrtc.org/2930243003
> Cr-Commit-Position: refs/heads/master@{#18645}
> Committed: fe1aa82c63TBR=ossu@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7831
Review-Url: https://codereview.webrtc.org/2947563002
Cr-Commit-Position: refs/heads/master@{#18649}
Reason for revert:
breaking downstream projects
Original issue's description:
> Opus implementation of the AudioDecoderFactoryTemplate API
>
> BUG=webrtc:7837
>
> Review-Url: https://codereview.webrtc.org/2942733003
> Cr-Commit-Position: refs/heads/master@{#18646}
> Committed: d053fe4ab3TBR=ossu@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7837
Review-Url: https://codereview.webrtc.org/2944763002
Cr-Commit-Position: refs/heads/master@{#18648}
Now the templated AudioEncoderFactory can create Opus encoders!
BUG=webrtc:7831
Review-Url: https://codereview.webrtc.org/2930243003
Cr-Commit-Position: refs/heads/master@{#18645}
Now the templated AudioEncoderFactory can create G722 encoders!
BUG=webrtc:7833
Review-Url: https://codereview.webrtc.org/2934833002
Cr-Commit-Position: refs/heads/master@{#18644}
Now the templated AudioDecoderFactory can create G722 decoders!
BUG=webrtc:7839
Review-Url: https://codereview.webrtc.org/2940833002
Cr-Commit-Position: refs/heads/master@{#18643}
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}
Specifically, just like SafeMin() and SafeMax() it handles all
combinations of integer and all
combinations of floating-point arguments by picking a
result type that is guaranteed to be able to hold the result.
This CL also replaces a bunch of std::min + std:max call pairs with
calls to SafeClamp()---the ones that could easily be found by grep
because "min" and "max" were on the same line. :-)
BUG=webrtc:7459
Review-Url: https://codereview.webrtc.org/2808513003
Cr-Commit-Position: refs/heads/master@{#18542}
This is a minor clean-up made possible by simplifications done in the
past.
Bug: none
Change-Id: Id0ea167572f8da36db5de949441f67a2a18555be
Reviewed-on: https://chromium-review.googlesource.com/528073
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18490}
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}
Reason for revert:
Fixing and re-landing.
Original issue's description:
> Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ )
>
> Reason for revert:
> ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here.
> #include "webrtc/base/checks.h"
> ^-------------------
> The target:
> //webrtc/test:fileutils_unittests
> is including a file from the target:
> //webrtc/base:rtc_base_approved
>
> It's usually best to depend directly on the destination target.
> In some cases, the destination target is considered a subcomponent
> of an intermediate target. In this case, the intermediate target
> should depend publicly on the destination to forward the ability
> to include headers.
>
> Dependency chain (there may also be others):
> //webrtc/test:fileutils_unittests -->
> //webrtc/test:fileutils --[private]-->
> //webrtc/base:rtc_base_approved
>
>
> Original issue's description:
> > Enabling `gn check` on webrtc/test
> >
> > BUG=webrtc:6828
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2911203002
> > Cr-Commit-Position: refs/heads/master@{#18372}
> > Committed: db5bb404b0
>
> TBR=kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6828
>
> Review-Url: https://codereview.webrtc.org/2920763002
> Cr-Commit-Position: refs/heads/master@{#18375}
> Committed: 1a6f143d07TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2918793002
Cr-Commit-Position: refs/heads/master@{#18376}
