Adding bit exactness test for Opus decoding in NetEq.

Adding bit exactness test for Opus decoding in NetEq.

Opus has become the mostly used codec in WebRTC. There, however, is no bit exactness test for Opus decoding in NetEq.

The new RTP file is generated by the following steps:
    1. Encode a clean RTP file with Opus
RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960 opus 1

    2. Adding jitter to the clean RTP file
RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp
(Note: jitter.dat does not exist in WebRTC resources folder. Check the source code for RTPjitter to know how to define such a file.)

BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the reference output.

Committed: https://crrev.com/93c08b74382b952aec56b0f74484d78dec3398e0
Cr-Commit-Position: refs/heads/master@{#11113}

[minyue-webrtc, 2015-12-17 13:25:34]

Patchset #1 (id:1) has been deleted

[minyue-webrtc, 2015-12-17 13:25:40]

Patchset #1 (id:20001) has been deleted

[minyue-webrtc, 2015-12-17 13:33:12]

Description was changed from

==========
unittest revised


reformat


small


adding Opus to RTPencode

BUG=
==========

to

==========
Adding bit exactness test for Opus decoding in NetEq.

Opus has become the mostly used codec in WebRTC. There, however, is no bit
exactness test for Opus decoding in NetEq.

The new RTP file is generated by the following steps:



BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the
reference output.
==========

[minyue-webrtc, 2015-12-17 13:38:22]

Description was changed from

==========
Adding bit exactness test for Opus decoding in NetEq.

Opus has become the mostly used codec in WebRTC. There, however, is no bit
exactness test for Opus decoding in NetEq.

The new RTP file is generated by the following steps:



BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the
reference output.
==========

to

==========
Adding bit exactness test for Opus decoding in NetEq.

Opus has become the mostly used codec in WebRTC. There, however, is no bit
exactness test for Opus decoding in NetEq.

The new RTP file is generated by the following steps:
    1. Encode a clean RTP file with Opus
RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960
opus 1

    2. Adding jitter to the clean RTP file
RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp

BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the
reference output.
==========

[minyue-webrtc, 2015-12-17 13:39:44]

Patchset #1 (id:40001) has been deleted

[minyue-webrtc, 2015-12-17 13:40:36]

minyue@webrtc.org changed reviewers:
+ henrik.lundin@webrtc.org, ivoc@webrtc.org

[minyue-webrtc, 2015-12-17 13:40:37]

Hi,

Here is a  further change in NetEq unittest. Please take a look.

https://codereview.webrtc.org/1515113002/diff/60001/webrtc/modules/audio_codi...
File webrtc/modules/audio_coding/neteq/neteq_unittest.cc (right):

https://codereview.webrtc.org/1515113002/diff/60001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/neteq/neteq_unittest.cc:232:
ASSERT_EQ(stats.secondary_decoded_rate, ref_stats.secondary_decoded_rate());
new Opus bit exactness test utilizes FEC decoding and activates
secondary_decoded_rate

https://codereview.webrtc.org/1515113002/diff/60001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/neteq/neteq_unittest.cc:520:
defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
only a reformatting

[minyue-webrtc, 2015-12-17 13:42:19]

Description was changed from

==========
Adding bit exactness test for Opus decoding in NetEq.

Opus has become the mostly used codec in WebRTC. There, however, is no bit
exactness test for Opus decoding in NetEq.

The new RTP file is generated by the following steps:
    1. Encode a clean RTP file with Opus
RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960
opus 1

    2. Adding jitter to the clean RTP file
RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp

BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the
reference output.
==========

to

==========
Adding bit exactness test for Opus decoding in NetEq.

Opus has become the mostly used codec in WebRTC. There, however, is no bit
exactness test for Opus decoding in NetEq.

The new RTP file is generated by the following steps:
    1. Encode a clean RTP file with Opus
RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960
opus 1

    2. Adding jitter to the clean RTP file
RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp
(Note: jitter.dat does not exist in WebRTC resources folder. Check the source
code for RTPjitter to know how to define such a file.)

BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the
reference output.
==========

[hlundin-webrtc, 2015-12-17 14:53:37]

lgtm with two comments.

Thanks for fixing!

https://codereview.webrtc.org/1515113002/diff/60001/webrtc/modules/audio_codi...
File webrtc/modules/audio_coding/neteq/neteq_unittest.cc (right):

https://codereview.webrtc.org/1515113002/diff/60001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/neteq/neteq_unittest.cc:567: // Note that
neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
Delete this comment, it relates to another reference file.

https://codereview.webrtc.org/1515113002/diff/60001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/neteq/neteq_unittest.cc:572: #if defined(_MSC_VER)
&& (_MSC_VER >= 1700)
Did you check what our minimum VS version is nowadays? Maybe we can get rid of
this extra code.

[ivoc, 2015-12-17 15:18:15]

Nice work! LGTM, see comment below.

https://codereview.webrtc.org/1515113002/diff/60001/webrtc/modules/audio_codi...
File webrtc/modules/audio_coding/neteq/neteq_unittest.cc (right):

https://codereview.webrtc.org/1515113002/diff/60001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/neteq/neteq_unittest.cc:566:
"resources/audio_coding/neteq_opus.rtp";
Why not use webrtc::test::ResourcePath for this one too?

[minyue-webrtc, 2015-12-18 12:11:20]

Patchset #2 (id:80001) has been deleted

[minyue-webrtc, 2015-12-18 12:15:12]

https://codereview.webrtc.org/1515113002/diff/60001/webrtc/modules/audio_codi...
File webrtc/modules/audio_coding/neteq/neteq_unittest.cc (right):

https://codereview.webrtc.org/1515113002/diff/60001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/neteq/neteq_unittest.cc:566:
"resources/audio_coding/neteq_opus.rtp";
On 2015/12/17 15:18:15, ivoc wrote:
> Why not use webrtc::test::ResourcePath for this one too?

Done.

https://codereview.webrtc.org/1515113002/diff/60001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/neteq/neteq_unittest.cc:567: // Note that
neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
On 2015/12/17 14:53:37, hlundin-webrtc wrote:
> Delete this comment, it relates to another reference file.

sorry. I just did too much copy and paste.

https://codereview.webrtc.org/1515113002/diff/60001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/neteq/neteq_unittest.cc:572: #if defined(_MSC_VER)
&& (_MSC_VER >= 1700)
On 2015/12/17 14:53:37, hlundin-webrtc wrote:
> Did you check what our minimum VS version is nowadays? Maybe we can get rid of
> this extra code.

I don't know how to test VS version, but I think it should not matter much.

[minyue-webrtc, 2015-12-18 12:15:47]

The CQ bit was checked by minyue@webrtc.org

[minyue-webrtc, 2015-12-18 12:15:47]

The patchset sent to the CQ was uploaded after l-g-t-m from ivoc@webrtc.org,
henrik.lundin@webrtc.org
Link to the patchset: https://codereview.webrtc.org/1515113002/#ps100001 (title:
"clean")

[commit-bot: I haz the power, 2015-12-18 12:15:52]

CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1515113002/100001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1515113002/100001

[commit-bot: I haz the power, 2015-12-18 12:33:04]

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

[commit-bot: I haz the power, 2015-12-18 12:33:05]

Try jobs failed on following builders:
  win_x64_rel on tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/win_x64_rel/builds/11483)

[minyue-webrtc, 2015-12-22 16:46:28]

The CQ bit was checked by minyue@webrtc.org

[minyue-webrtc, 2015-12-22 16:46:28]

The patchset sent to the CQ was uploaded after l-g-t-m from ivoc@webrtc.org,
henrik.lundin@webrtc.org
Link to the patchset: https://codereview.webrtc.org/1515113002/#ps120001 (title:
"adding ref file for win (since non-bit-exact)")

[commit-bot: I haz the power, 2015-12-22 16:46:36]

CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1515113002/120001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1515113002/120001

[commit-bot: I haz the power, 2015-12-22 17:57:44]

Description was changed from

==========
Adding bit exactness test for Opus decoding in NetEq.

Opus has become the mostly used codec in WebRTC. There, however, is no bit
exactness test for Opus decoding in NetEq.

The new RTP file is generated by the following steps:
    1. Encode a clean RTP file with Opus
RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960
opus 1

    2. Adding jitter to the clean RTP file
RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp
(Note: jitter.dat does not exist in WebRTC resources folder. Check the source
code for RTPjitter to know how to define such a file.)

BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the
reference output.
==========

to

==========
Adding bit exactness test for Opus decoding in NetEq.

Opus has become the mostly used codec in WebRTC. There, however, is no bit
exactness test for Opus decoding in NetEq.

The new RTP file is generated by the following steps:
    1. Encode a clean RTP file with Opus
RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960
opus 1

    2. Adding jitter to the clean RTP file
RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp
(Note: jitter.dat does not exist in WebRTC resources folder. Check the source
code for RTPjitter to know how to define such a file.)

BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the
reference output.
==========

[commit-bot: I haz the power, 2015-12-22 17:57:45]

Committed patchset #3 (id:120001)

[commit-bot: I haz the power, 2015-12-22 17:57:51]

Description was changed from

==========
Adding bit exactness test for Opus decoding in NetEq.

Opus has become the mostly used codec in WebRTC. There, however, is no bit
exactness test for Opus decoding in NetEq.

The new RTP file is generated by the following steps:
    1. Encode a clean RTP file with Opus
RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960
opus 1

    2. Adding jitter to the clean RTP file
RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp
(Note: jitter.dat does not exist in WebRTC resources folder. Check the source
code for RTPjitter to know how to define such a file.)

BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the
reference output.
==========

to

==========
Adding bit exactness test for Opus decoding in NetEq.

Opus has become the mostly used codec in WebRTC. There, however, is no bit
exactness test for Opus decoding in NetEq.

The new RTP file is generated by the following steps:
    1. Encode a clean RTP file with Opus
RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960
opus 1

    2. Adding jitter to the clean RTP file
RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp
(Note: jitter.dat does not exist in WebRTC resources folder. Check the source
code for RTPjitter to know how to define such a file.)

BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the
reference output.

Committed: https://crrev.com/93c08b74382b952aec56b0f74484d78dec3398e0
Cr-Commit-Position: refs/heads/master@{#11113}
==========

[commit-bot: I haz the power, 2015-12-22 17:57:53]

Patchset 3 (id:??) landed as
https://crrev.com/93c08b74382b952aec56b0f74484d78dec3398e0
Cr-Commit-Position: refs/heads/master@{#11113}