Add audio streams to CallTest and a first A/V call test.

Add audio streams to CallTest and a first A/V call test.

Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code.

R=pbos@webrtc.org
TBR=kjellander@webrtc.org

BUG=webrtc:5263
Committed: https://crrev.com/9fea80f50daab46f20d4a6fc67b0144fbbbf56cd
Cr-Commit-Position: refs/heads/master@{#11171}

[stefan-webrtc, 2015-12-21 16:42:53]

The CQ bit was checked by stefan@webrtc.org to run a CQ dry run

[commit-bot: I haz the power, 2015-12-21 16:43:02]

Dry run: CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1542653002/40001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1542653002/40001

[stefan-webrtc, 2015-12-21 16:43:11]

stefan@webrtc.org changed reviewers:
+ pbos@webrtc.org

[stefan-webrtc, 2015-12-21 16:44:37]

Description was changed from

==========
Add audio streams to CallTest and a first A/V call test.

Add audio send and receive streams to CallTest and call the necessary voice
engine APIs for the streams to be usable. Verifies the implementation by adding
a simple test which monitors outgoing packets and checks that both audio and
video is being sent.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and
should in the future replace a lot of that code.

BUG=webrtc:5263
==========

to

==========
Add audio streams to CallTest and a first A/V call test.

Add audio send and receive streams to CallTest and call the necessary voice
engine APIs for the streams to be usable. Verifies the implementation by adding
a simple test which monitors outgoing packets and checks that both audio and
video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and
should in the future replace a lot of that code.

BUG=webrtc:5263
==========

[commit-bot: I haz the power, 2015-12-21 16:46:14]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2015-12-21 16:46:15]

Dry run: Try jobs failed on following builders:
  win_compile_dbg on tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/win_compile_dbg/builds/...)

[stefan-webrtc, 2015-12-21 17:05:24]

On 2015/12/21 16:46:15, commit-bot: I haz the power wrote:
> Dry run: Try jobs failed on following builders:
>   win_compile_dbg on tryserver.webrtc (JOB_FAILED,
>
http://build.chromium.org/p/tryserver.webrtc/builders/win_compile_dbg/builds/...)

It appears that this is somehow using the sound card even though it is being set
up with a fake audio device. I'll have to look into that.

[stefan-webrtc, 2015-12-22 08:54:29]

Create audio devices earlier

[stefan-webrtc, 2015-12-22 08:56:01]

The CQ bit was checked by stefan@webrtc.org to run a CQ dry run

[commit-bot: I haz the power, 2015-12-22 08:56:14]

Dry run: CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1542653002/60001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1542653002/60001

[commit-bot: I haz the power, 2015-12-22 09:00:47]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2015-12-22 09:00:48]

Dry run: Exceeded global retry quota

[stefan-webrtc, 2015-12-22 10:09:00]

Disable test on Android since no files are available.

[stefan-webrtc, 2015-12-22 10:37:40]

Fix win compile issue?

[stefan-webrtc, 2015-12-22 10:51:42]

Fix win compile issue?

[stefan-webrtc, 2015-12-22 11:14:01]

Fix win compile issue?

[stefan-webrtc, 2015-12-22 13:51:36]

Now then?

[pbos-webrtc, 2015-12-28 22:22:31]

https://codereview.webrtc.org/1542653002/diff/60001/webrtc/call/call_unittest.cc
File webrtc/call/call_unittest.cc (right):

https://codereview.webrtc.org/1542653002/diff/60001/webrtc/call/call_unittest...
webrtc/call/call_unittest.cc:189: RunBaseTest(&test, FakeNetworkPipe::Config());
Wondering if this could be confused being put under namespace test, maybe worth
figuring out?

[stefan-webrtc, 2015-12-29 09:37:07]

https://codereview.webrtc.org/1542653002/diff/60001/webrtc/call/call_unittest.cc
File webrtc/call/call_unittest.cc (right):

https://codereview.webrtc.org/1542653002/diff/60001/webrtc/call/call_unittest...
webrtc/call/call_unittest.cc:189: RunBaseTest(&test, FakeNetworkPipe::Config());
On 2015/12/28 22:22:31, pbos-webrtc wrote:
> Wondering if this could be confused being put under namespace test, maybe
worth
> figuring out?

I moved it in the latest patch set. Still under test, but so are all of the
other tests in end_to_end_tests.cc. Oh, you are saying that "test" here may be
confused with "namespace test". Maybe :)

[pbos-webrtc, 2015-12-30 17:19:59]

On 2015/12/29 09:37:07, stefan-webrtc (holmer) wrote:
>
https://codereview.webrtc.org/1542653002/diff/60001/webrtc/call/call_unittest.cc
> File webrtc/call/call_unittest.cc (right):
> 
>
https://codereview.webrtc.org/1542653002/diff/60001/webrtc/call/call_unittest...
> webrtc/call/call_unittest.cc:189: RunBaseTest(&test,
FakeNetworkPipe::Config());
> On 2015/12/28 22:22:31, pbos-webrtc wrote:
> > Wondering if this could be confused being put under namespace test, maybe
> worth
> > figuring out?
> 
> I moved it in the latest patch set. Still under test, but so are all of the
> other tests in end_to_end_tests.cc. Oh, you are saying that "test" here may be
> confused with "namespace test". Maybe :)

We could take a look with Mr. Gates when we're both back, that'll probably be
easiest.

[stefan-webrtc, 2016-01-07 11:49:44]

Win issue fixed.

[stefan-webrtc, 2016-01-07 11:50:01]

Mr. Gates to the rescue. PTAL!

[stefan-webrtc, 2016-01-07 11:50:08]

The CQ bit was checked by stefan@webrtc.org to run a CQ dry run

[commit-bot: I haz the power, 2016-01-07 11:50:18]

Dry run: CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1542653002/180001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1542653002/180001

[commit-bot: I haz the power, 2016-01-07 11:56:52]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2016-01-07 11:56:53]

Dry run: Try jobs failed on following builders:
  presubmit on tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/presubmit/builds/2720)

[pbos-webrtc, 2016-01-07 14:23:37]

let me take another look

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.cc
File webrtc/test/call_test.cc (right):

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.cc...
webrtc/test/call_test.cc:56: if (audio_enabled) {
Put down a comment (somewhere) that says that these audio parts of the call
config will be overridden when running RunBaseTest (and why)

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.cc...
webrtc/test/call_test.cc:100: RTC_DCHECK(test->GetNumVideoStreams() > 0 ||
audio_enabled);
GetNumAudioStreams() > 0 instead

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.cc...
webrtc/test/call_test.cc:106: test->OnAudioStreamsCreated(audio_send_stream_,
audio_receive_streams_);
Put inside audio_enabled

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.h
File webrtc/test/call_test.h (right):

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.h#...
webrtc/test/call_test.h:69: void CreateMatchingReceiveConfigs(bool
create_audio_recv_stream,
Store how many was created in CreateSendConfig and remove this parameter.

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.h#...
webrtc/test/call_test.h:103: // TODO(holmer): Remove once voice engine is fully
refactored to the new API.
VoiceEngine

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.h#...
webrtc/test/call_test.h:114: VoiceEngine* send_voice_engine_;
Split into send/recv member "blocks", maybe a struct as you're suggesting to me
right now in person.

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.h#...
webrtc/test/call_test.h:157: std::vector<AudioReceiveStream::Config>*
receive_configsg);
configsgsgsgsgsgsgsg

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/video/end_to_end_...
File webrtc/video/end_to_end_tests.cc (right):

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/video/end_to_end_...
webrtc/video/end_to_end_tests.cc:3251: #if defined(WEBRTC_ANDROID)
Re-enable on Android, look into the isolate files to have this run.

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/video/end_to_end_...
webrtc/video/end_to_end_tests.cc:3278:
(*receive_configs)[0].rtp.extensions.clear();
(*receive_configs)[0].rtp.extensions = send_config->rtp.extensions; here and
below

[stefan-webrtc, 2016-01-07 15:16:36]

Comments addressed.

[stefan-webrtc, 2016-01-07 15:16:44]

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.cc
File webrtc/test/call_test.cc (right):

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.cc...
webrtc/test/call_test.cc:56: if (audio_enabled) {
On 2016/01/07 14:23:37, pbos-webrtc wrote:
> Put down a comment (somewhere) that says that these audio parts of the call
> config will be overridden when running RunBaseTest (and why)

Done.

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.cc...
webrtc/test/call_test.cc:100: RTC_DCHECK(test->GetNumVideoStreams() > 0 ||
audio_enabled);
On 2016/01/07 14:23:37, pbos-webrtc wrote:
> GetNumAudioStreams() > 0 instead

Done.

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.cc...
webrtc/test/call_test.cc:106: test->OnAudioStreamsCreated(audio_send_stream_,
audio_receive_streams_);
On 2016/01/07 14:23:37, pbos-webrtc wrote:
> Put inside audio_enabled

Done.

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.h
File webrtc/test/call_test.h (right):

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.h#...
webrtc/test/call_test.h:69: void CreateMatchingReceiveConfigs(bool
create_audio_recv_stream,
On 2016/01/07 14:23:37, pbos-webrtc wrote:
> Store how many was created in CreateSendConfig and remove this parameter.

Done.

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.h#...
webrtc/test/call_test.h:103: // TODO(holmer): Remove once voice engine is fully
refactored to the new API.
On 2016/01/07 14:23:37, pbos-webrtc wrote:
> VoiceEngine

Done.

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.h#...
webrtc/test/call_test.h:114: VoiceEngine* send_voice_engine_;
On 2016/01/07 14:23:37, pbos-webrtc wrote:
> Split into send/recv member "blocks", maybe a struct as you're suggesting to
me
> right now in person.

Done.

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/test/call_test.h#...
webrtc/test/call_test.h:157: std::vector<AudioReceiveStream::Config>*
receive_configsg);
On 2016/01/07 14:23:37, pbos-webrtc wrote:
> configsgsgsgsgsgsgsg

Done.

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/video/end_to_end_...
File webrtc/video/end_to_end_tests.cc (right):

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/video/end_to_end_...
webrtc/video/end_to_end_tests.cc:3251: #if defined(WEBRTC_ANDROID)
On 2016/01/07 14:23:37, pbos-webrtc wrote:
> Re-enable on Android, look into the isolate files to have this run.

Done.

https://codereview.webrtc.org/1542653002/diff/180001/webrtc/video/end_to_end_...
webrtc/video/end_to_end_tests.cc:3278:
(*receive_configs)[0].rtp.extensions.clear();
On 2016/01/07 14:23:37, pbos-webrtc wrote:
> (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions; here and
> below

Done.

[stefan-webrtc, 2016-01-07 15:42:15]

The CQ bit was checked by stefan@webrtc.org to run a CQ dry run

[commit-bot: I haz the power, 2016-01-07 15:42:20]

Dry run: CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1542653002/200001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1542653002/200001

[pbos-webrtc, 2016-01-07 15:49:02]

lgtm

https://codereview.webrtc.org/1542653002/diff/200001/webrtc/test/call_test.h
File webrtc/test/call_test.h (right):

https://codereview.webrtc.org/1542653002/diff/200001/webrtc/test/call_test.h#...
webrtc/test/call_test.h:110: struct OldVoiceEngine {
VoiceEngineState

[commit-bot: I haz the power, 2016-01-07 16:39:12]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2016-01-07 16:39:13]

Dry run: This issue passed the CQ dry run.

[stefan-webrtc, 2016-01-07 16:41:26]

Comment addressed

[stefan-webrtc, 2016-01-07 16:42:44]

stefan@webrtc.org changed reviewers:
+ kjellander@webrtc.org

[stefan-webrtc, 2016-01-07 16:42:44]

+kjellander, could you take a look at voice_engine_tests.isolate after the fact?

[stefan-webrtc, 2016-01-07 16:43:00]

Description was changed from

==========
Add audio streams to CallTest and a first A/V call test.

Add audio send and receive streams to CallTest and call the necessary voice
engine APIs for the streams to be usable. Verifies the implementation by adding
a simple test which monitors outgoing packets and checks that both audio and
video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and
should in the future replace a lot of that code.

BUG=webrtc:5263
==========

to

==========
Add audio streams to CallTest and a first A/V call test.

Add audio send and receive streams to CallTest and call the necessary voice
engine APIs for the streams to be usable. Verifies the implementation by adding
a simple test which monitors outgoing packets and checks that both audio and
video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and
should in the future replace a lot of that code.

TBR=kjellander@webrtc.org

BUG=webrtc:5263
==========

[stefan-webrtc, 2016-01-07 16:43:37]

Description was changed from

==========
Add audio streams to CallTest and a first A/V call test.

Add audio send and receive streams to CallTest and call the necessary voice
engine APIs for the streams to be usable. Verifies the implementation by adding
a simple test which monitors outgoing packets and checks that both audio and
video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and
should in the future replace a lot of that code.

TBR=kjellander@webrtc.org

BUG=webrtc:5263
==========

to

==========
Add audio streams to CallTest and a first A/V call test.

Add audio send and receive streams to CallTest and call the necessary voice
engine APIs for the streams to be usable. Verifies the implementation by adding
a simple test which monitors outgoing packets and checks that both audio and
video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and
should in the future replace a lot of that code.

R=pbos@webrtc.org
TBR=kjellander@webrtc.org

BUG=webrtc:5263

Committed:
https://chromium.googlesource.com/external/webrtc/+/9fea80f50daab46f20d4a6fc6...
==========

[stefan-webrtc, 2016-01-07 16:43:38]

Committed patchset #12 (id:220001) manually as
9fea80f50daab46f20d4a6fc67b0144fbbbf56cd (presubmit successful).

[commit-bot: I haz the power, 2016-01-07 16:43:41]

Description was changed from

==========
Add audio streams to CallTest and a first A/V call test.

Add audio send and receive streams to CallTest and call the necessary voice
engine APIs for the streams to be usable. Verifies the implementation by adding
a simple test which monitors outgoing packets and checks that both audio and
video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and
should in the future replace a lot of that code.

R=pbos@webrtc.org
TBR=kjellander@webrtc.org

BUG=webrtc:5263

Committed:
https://chromium.googlesource.com/external/webrtc/+/9fea80f50daab46f20d4a6fc6...
==========

to

==========
Add audio streams to CallTest and a first A/V call test.

Add audio send and receive streams to CallTest and call the necessary voice
engine APIs for the streams to be usable. Verifies the implementation by adding
a simple test which monitors outgoing packets and checks that both audio and
video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and
should in the future replace a lot of that code.

R=pbos@webrtc.org
TBR=kjellander@webrtc.org

BUG=webrtc:5263

Committed: https://crrev.com/9fea80f50daab46f20d4a6fc67b0144fbbbf56cd
Cr-Commit-Position: refs/heads/master@{#11171}
==========

[commit-bot: I haz the power, 2016-01-07 16:43:42]

Patchset 12 (id:??) landed as
https://crrev.com/9fea80f50daab46f20d4a6fc67b0144fbbbf56cd
Cr-Commit-Position: refs/heads/master@{#11171}

[stefan-webrtc, 2016-01-07 16:43:52]

On 2016/01/07 16:42:44, stefan-webrtc (holmer) wrote:
> +kjellander, could you take a look at voice_engine_tests.isolate after the
fact?

video_engine_tests.isolate, I mean...

[kjellander_webrtc, 2016-01-07 19:14:16]

webrtc/video_engine_tests.isolate: lgtm