To more easily determine if for example the AEC is not working properly one could monitor how often the estimated delay is out of bounds. With out of bounds we mean either being negative or too large, where both cases will break the AEC.
A new delay metric is added telling the user how often poor delay values were estimated. This is measured in percentage since last time the metrics were calculated.
All APIs have been updated with a third parameter with EchoCancellation::GetDelayMetrics() giving the option to exclude the new metric not to break existing code.
The new metric has been added to audio_processing_unittests with an additional protobuf member, and reference files accordingly updated.
voe_auto_test has not been updated to display the new metric.
BUG=4246
TESTED=audioproc on files
R=aluebs@webrtc.org, andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39739004
Cr-Commit-Position: refs/heads/master@{#8230}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8230 4adac7df-926f-26a2-2b94-8c16560cd09d
Changes differing from https://webrtc-codereview.appspot.com/37859004:
* I put the include_tests==1 stuff of audio_coding.gypi in its
own audio_coding_tests.gypi file, including the Android and isolate
targets which were incorrectly located in the previous CL
* I moved the bwe utilities in remote_bitrate_estimator.gypi
into include_tests==1 since they depend on test.gyp after I
cleaned up the duplicated inclusion of rtp_file_reader.cc
R=stefan@webrtc.orgTBR=tina.legrand@webrtc.org
TESTED=Passing gyp and compile using:
webrtc/build/gyp_webrtc -Dinclude_tests=1
webrtc/build/gyp_webrtc -Dinclude_tests=0
I also setup a Chromium checkout with my checkout mounted in
third_party/webrtc and ran build/gyp_chromium successfully.
BUG=4185
Review URL: https://webrtc-codereview.appspot.com/33159004
Cr-Commit-Position: refs/heads/master@{#8205}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8205 4adac7df-926f-26a2-2b94-8c16560cd09d
- Add max_bit_rate and max_payload_size_bytes to config structs.
- Fix support for 48 kHz sample rate.
- Fix iSAC-RED.
- Add method UpdateDecoderSampleRate().
- Update locking structure with a separate lock for local member
variables used by the encoder methods.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41659004
Cr-Commit-Position: refs/heads/master@{#8204}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8204 4adac7df-926f-26a2-2b94-8c16560cd09d
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.
It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.
BUG=
R=henrika@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37849004
Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
This required to move the AGC tools source files
into webrtc/tools and create a new agc_test_utils target.
Since audio_codec_speed_tests.gypi referenced sources above,
the best approach I could come up with was to add an audio_coding.gypi
file at a higher level and move the targets in there (+ the includes from
modules.gyp which is an improvement IMO).
I also added a PRESUBMIT.py check to prevent new source
entries being added with <(webrtc_root) in the path.
BUG=4185
R=andrew@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37859004
Cr-Commit-Position: refs/heads/master@{#8197}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8197 4adac7df-926f-26a2-2b94-8c16560cd09d
When using the paced sender, packets will be put into the rtp packet
history and then retreived from there again when it is time to send.
In some cases (low send bitrate and very large frames created) this
may overflow, causing packets to be overwritten in the packet history
before they have been sent.
Check this condition and expand history size if needed.
This is primarily triggered during screenshare, when
switching to a large picture with lots of high frequency
details in it.
BUG=4171
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34879004
Cr-Commit-Position: refs/heads/master@{#8195}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8195 4adac7df-926f-26a2-2b94-8c16560cd09d
The following three methods are added:
rtp_timestamp_rate_hz()
SetTargetBitrate()
SetProjectedPacketLossRate()
Default implementations are provided, and a few overrides are
implemented. AudioEncoderCopyRed and AudioEncoderCng propagate the new
methods to the underlying speech codec.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34049004
Cr-Commit-Position: refs/heads/master@{#8171}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8171 4adac7df-926f-26a2-2b94-8c16560cd09d
Previously, the delay estimate history was reset every time the metrics were pulled. This required all clients to be on the same thread and make use of one call.
Now we use a fixed aggregation window of one second and when a client pulls the metrics you get the latest value.
Under certain circumstances like tests you would like to have the aggregation window set to the recording length. We therefore turn on the fixed aggregation window after the first call.
BUG=2994
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38759004
Cr-Commit-Position: refs/heads/master@{#8170}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8170 4adac7df-926f-26a2-2b94-8c16560cd09d
Broke compile on the Chromium FYI bots:
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/3483http://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac/builds/16028http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/14293
Error:
In file included from ../../third_party/webrtc/voice_engine/channel.cc:13:
In file included from ../../third_party/webrtc/base/checks.h:22:
In file included from ../../third_party/webrtc/overrides/webrtc/base/logging.h:35:
../../base/logging.h:367:9:error: 'LOG' macro redefined [-Werror,-Wmacro-redefined]
#define LOG(severity) LAZY_STREAM(LOG_STREAM(severity), LOG_IS_ON(severity))
^
../../third_party/webrtc/system_wrappers/interface/logging.h:123:9: note: previous definition is here
#define LOG(sev) \
^
In file included from ../../third_party/webrtc/voice_engine/channel.cc:13:
In file included from ../../third_party/webrtc/base/checks.h:22:
../../third_party/webrtc/overrides/webrtc/base/logging.h:189:9:error: 'LOG_V' macro redefined [-Werror,-Wmacro-redefined]
#define LOG_V(sev) DIAGNOSTIC_LOG(sev, NONE, 0)
^
../../third_party/webrtc/system_wrappers/interface/logging.h:129:9: note: previous definition is here
#define LOG_V(sev) \
^
2 errors generated.
> Modify some tests to never use DTX disable mode
>
> DTX disable mode will be removed as a part of the ACM redesign work.
>
> COAUTHOR:kwiberg@webrtc.org
>
> R=henrika@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/34769004TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8129 4adac7df-926f-26a2-2b94-8c16560cd09d
