19 Commits

Author SHA1 Message Date
ehmaldonado
f6a861ab6c Remove remains of webrtc/base
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}
2017-07-19 17:40:47 +00:00
ehmaldonado
370dd47973 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
Reason for revert:
Breaks lots of downstream projects.

Original issue's description:
> Remove remains of webrtc/base
>
> 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}
> Committed:
9483b49baf

TBR=kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7634

Review-Url: https://codereview.webrtc.org/2976633002
Cr-Commit-Position: refs/heads/master@{#18949}
2017-07-10 12:58:42 +00:00
ehmaldonado
9483b49baf Remove remains of webrtc/base
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}
2017-07-10 11:50:54 +00:00
Edward Lemur
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
Henrik Kjellander
dca1e09db7 Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"
This reverts commit c8fa692ec44fd6ba4fa3d085ac3161a262fc18c5.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2964773002 .
Cr-Commit-Position: refs/heads/master@{#18872}
2017-07-01 14:42:25 +00:00
kjellander
c8fa692ec4 Update includes for webrtc/{base => rtc_base} rename (1/3)
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}
2017-06-30 21:02:00 +00:00
alessiob
19e087fc91 This CL finalizes the Conversational Speech tool.
The following changes have been made:
- command line args wired,
- user output added,
- final polishing.

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2808053002
Cr-Commit-Position: refs/heads/master@{#18609}
2017-06-15 10:49:57 +00:00
alessiob
f9784f23d7 Reland of Conversational speech tool, simualtor + unit tests (patchset #1 id:1 of https://codereview.webrtc.org/2925123003/ )
Reason for revert:
Build file causing google3 compilation error fixed

Original issue's description:
> Revert of Conversational speech tool, simualtor + unit tests (patchset #12 id:220001 of https://codereview.webrtc.org/2790933002/ )
>
> Reason for revert:
> Compile Error.
>
> Original issue's description:
> > The simulator puts into action the schedule of speech turns encoded in a MultiEndCall instance. The output is a set of audio track pairs. There is one set for each speaker and each set contains one near-end and one far-end audio track. The tracks are directly written into wav files instead of creating them in memory. To speed up the creation of the output wav files, *all* the source audio tracks (i.e., the atomic speech turns) are pre-loaded.
> >
> > The ConversationalSpeechTest.MultiEndCallSimulator unit test defines a conversational speech sequence and creates two wav files (with pure tones at 440 and 880 Hz) that are used as atomic speech turn tracks.
> >
> > This CL also patches MultiEndCall in order to allow input audio tracks with same sample rate and single channel only.
> >
> > BUG=webrtc:7218
> >
> > Review-Url: https://codereview.webrtc.org/2790933002
> > Cr-Commit-Position: refs/heads/master@{#18480}
> > Committed: 6b648c4697
>
> TBR=minyue@webrtc.org,alessiob@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7218
>
> Review-Url: https://codereview.webrtc.org/2925123003
> Cr-Commit-Position: refs/heads/master@{#18481}
> Committed: 4c72cf43df

TBR=minyue@webrtc.org,charujain@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2930853002
Cr-Commit-Position: refs/heads/master@{#18606}
2017-06-15 09:24:59 +00:00
charujain
4c72cf43df Revert of Conversational speech tool, simualtor + unit tests (patchset #12 id:220001 of https://codereview.webrtc.org/2790933002/ )
Reason for revert:
Compile Error.

Original issue's description:
> The simulator puts into action the schedule of speech turns encoded in a MultiEndCall instance. The output is a set of audio track pairs. There is one set for each speaker and each set contains one near-end and one far-end audio track. The tracks are directly written into wav files instead of creating them in memory. To speed up the creation of the output wav files, *all* the source audio tracks (i.e., the atomic speech turns) are pre-loaded.
>
> The ConversationalSpeechTest.MultiEndCallSimulator unit test defines a conversational speech sequence and creates two wav files (with pure tones at 440 and 880 Hz) that are used as atomic speech turn tracks.
>
> This CL also patches MultiEndCall in order to allow input audio tracks with same sample rate and single channel only.
>
> BUG=webrtc:7218
>
> Review-Url: https://codereview.webrtc.org/2790933002
> Cr-Commit-Position: refs/heads/master@{#18480}
> Committed: 6b648c4697

TBR=minyue@webrtc.org,alessiob@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2925123003
Cr-Commit-Position: refs/heads/master@{#18481}
2017-06-07 18:59:09 +00:00
alessiob
6b648c4697 The simulator puts into action the schedule of speech turns encoded in a MultiEndCall instance. The output is a set of audio track pairs. There is one set for each speaker and each set contains one near-end and one far-end audio track. The tracks are directly written into wav files instead of creating them in memory. To speed up the creation of the output wav files, *all* the source audio tracks (i.e., the atomic speech turns) are pre-loaded.
The ConversationalSpeechTest.MultiEndCallSimulator unit test defines a conversational speech sequence and creates two wav files (with pure tones at 440 and 880 Hz) that are used as atomic speech turn tracks.

This CL also patches MultiEndCall in order to allow input audio tracks with same sample rate and single channel only.

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2790933002
Cr-Commit-Position: refs/heads/master@{#18480}
2017-06-07 18:04:35 +00:00
alessiob
36e6a8f1bd WavReaderAdaptor is a simple adaptor of the existing class WavReader from webrtc/common_audio/wav_file.h. The adaptor was mainly needed to use dependency injection and easily test the MultiEndCall class (see https://codereview.webrtc.org/2761853002/).
The unit test ConversationalSpeechTest.MultiEndCallWavReaderAdaptorSine uses CreateSineWavFile() and writes temporary wav files that are used for test (deleted only if the test passes).

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2774423005
Cr-Commit-Position: refs/heads/master@{#17608}
2017-04-10 07:53:53 +00:00
alessiob
8d23c050f2 MultiEndCall::CheckTiming() verifies that a set of audio tracks and timing information is valid to simulate conversational speech. Unordered turns are rejected. Self cross-talk and cross-talk with 3 or more speakers are not permitted since it would require mixing at the simulation step.
This CL includes extensive tests to match accept or reject decisions on several different timing setups. The setups are simulated using mocks (by far more light-weight than using actual timing and audio track files).

The client code, the unit tests in this case, passes information about the fake audio tracks to MockWavReaderFactory. MockWavReader instances are then created using the parameters defined in the client code. To improve the readability of the tests, generator_unittest.cc includes a docstring explaining how each MultiEndCallSetup* test is documented.

Run tests as follows:
$ out/Default/modules_unittests --gtest_filter=ConversationalSpeechTest.*

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2781573002
Cr-Commit-Position: refs/heads/master@{#17592}
2017-04-07 19:05:08 +00:00
alessiob
3339743878 MultiEndCall is responsible for analyzing and validating timing information and audiotracks with which a multi-end call can be simulated.
The class creates one WavReaderInterface object for each unique audiotrack and builds the set of speaker names.
Validating if the audiotrack lengths and the timing information are compatible (and hence valid) is not implemented yet.

MultiEndCall is designed using dependency injection. This allows to use mock objects with which we can quickly simulate different timings and track lengths without needing actual wav files.

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2761853002
Cr-Commit-Position: refs/heads/master@{#17421}
2017-03-28 12:39:59 +00:00
alessiob
4b6463c0c9 Conversational Speech tool, rtc_test target replaced with entry in modules_unittests
The ConversationalSpeechTest.* unit tests are now part of modules_unittests.
The rtc_test target has been replaced with an rtc_source_set one.
The latter is included as dependency in audio_processing_unittests.

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2769863005
Cr-Commit-Position: refs/heads/master@{#17360}
2017-03-23 12:17:06 +00:00
alessiob
ce302b82c9 Conversational speech tool: timing model with data access.
The conversational_speech::Timing class models a list of turns.
Each turn, is identified by a speaker, the audiotrack name, and an offset in milliseconds.
The unit test checks that an instance of Timing is correctly populated and that save/reload leads to the same data.

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2750353002
Cr-Commit-Position: refs/heads/master@{#17346}
2017-03-22 15:23:46 +00:00
alessiob
3b51065c55 Conversational Speech generator, adding unit test.
Test for the conversational_speech::Config class and renaming.

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2749573002
Cr-Commit-Position: refs/heads/master@{#17301}
2017-03-18 10:45:31 +00:00
alessiob
5f32aca85e Reland "C++ porting of the initial python script for conversational speech
generation."

This CL removes the Python script and adds its C++ porting.
The former was in its early stage and it has permanently been removed.

This is a reland of https://codereview.webrtc.org/2740063004/ which
was reverted. Now the build errors are fixed.

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2752793002
Cr-Commit-Position: refs/heads/master@{#17300}
2017-03-18 09:29:13 +00:00
alessiob
dbbaa2ad66 Revert of C++ porting of the initial python script for conversational speech generation. (patchset #2 id:100001 of https://codereview.webrtc.org/2740063004/ )
Reason for revert:
Even if the conversational speech tool is external and not a core part of webrtc, there are too many trybots failing.

Original issue's description:
> C++ porting of the initial python script for conversational speech generation.
>
> This CL removes the Python script and adds its C++ porting.
> The former was in its early stage and it has permanently been removed.
>
> BUG=webrtc:7218
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2740063004
> Cr-Commit-Position: refs/heads/master@{#17254}
> Committed: 0cf3aa6d0d

TBR=henrik.lundin@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:7218

Review-Url: https://codereview.webrtc.org/2753843002
Cr-Commit-Position: refs/heads/master@{#17257}
2017-03-15 15:19:25 +00:00
alessiob
0cf3aa6d0d C++ porting of the initial python script for conversational speech generation.
This CL removes the Python script and adds its C++ porting.
The former was in its early stage and it has permanently been removed.

BUG=webrtc:7218
NOTRY=True

Review-Url: https://codereview.webrtc.org/2740063004
Cr-Commit-Position: refs/heads/master@{#17254}
2017-03-15 14:56:26 +00:00