Now the ChannelBuffer has 2 separate arrays, one for the full-band data and one for the splitted one. The corresponding accessors are added to the ChannelBuffer.
This is done to avoid having to refresh the bands pointers in AudioBuffer. It will also allow us to have a general accessor like data()[band][channel][sample].
All the files using the ChannelBuffer needed to be re-factored.
Tested with modules_unittests, common_audio_unittests, audioproc, audioproc_f, voe_cmd_test.
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36999004
Cr-Commit-Position: refs/heads/master@{#8318}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8318 4adac7df-926f-26a2-2b94-8c16560cd09d
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
One debug recording with non matching sample rates between render and capture revealed a bug in modules/audio_processing/test/process_test.cc
The far_frame (render audio frame) used was loaded with the capture rate instead of the render rate with a data length mismatch error as result.
BUG=N/A
TESTED=manually on linux
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7695 4adac7df-926f-26a2-2b94-8c16560cd09d
- Disables ApmTest.EchoCancellationReportsCorrectDelays
This test relys completely on the structure of how reported system delays are handled in AEC. In addition it assumes a fix setup of delay logging buffers. This test should be refactored.
- Adds flag to turn off reported_delay in audioproc
Now it is feasible to turn on and off the use of reported system delays.
BUG=N/A
R=aluebs@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6492 4adac7df-926f-26a2-2b94-8c16560cd09d
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
- Add an Initialize() overload to allow specification of format
parameters. This is mainly useful for testing, but could be used in
the cases where a consumer knows the format before the streams arrive.
- Add a reverse_sample_rate_hz_ parameter to prepare for mismatched
capture and render rates. There is no functional change as it is
currently constrained to match the capture rate.
- Fix a bug in the float dump: we need to use add_ rather than set_.
- Add a debug dump test for both int and float interfaces.
- Enable unpacking of float dumps.
- Enable audioproc to read float dumps.
- Move more shared functionality to test_utils.h, and generally tidy up
a bit by consolidating repeated code.
BUG=2894
TESTED=Verified that the output produced by the float debug dump test is
correct. Processed the resulting debug dump file with audioproc and
ensured that we get identical output. (This is crucial, as we need to
be able to exactly reproduce online results offline.)
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5676 4adac7df-926f-26a2-2b94-8c16560cd09d
Recordings that had a AnalyzeReverseStream() call prior to ProcessStream() where aborted due to sample rates being set upon call by ProcessStream(). That change was done in r5346.
Before we have a smarter handling on how to set sample rate automatically, this CL adds back that setting.
BUG=
TESTED=trybots, modules_unittests
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5394 4adac7df-926f-26a2-2b94-8c16560cd09d
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
The measurement and trace entries had been mixed up in the calls to webrtc::test::PrintResult, resulting in the plotted graphs were named after the metric. The parameter names are quite confusing which probably led to this.
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/1093007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3496 4adac7df-926f-26a2-2b94-8c16560cd09d
This makes the following files be written into the output dir instead of
the current working dir:
* out.pcm
* vad_out.dat
* ns_prob.dat
TEST=out/Debug/audioproc -aecm -ns -agc --fixed_digital --perf -pb
resources/audioproc.aecdump
All trybots passing.
BUG=none
Review URL: https://webrtc-codereview.appspot.com/1003005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3302 4adac7df-926f-26a2-2b94-8c16560cd09d
