The modification only uses the unique part of the
WebRtcIsacfix_AutocorrC function. Pass FiltersTest.AutocorrFixTest test
on both ARMv7 and ARM64, and the single function performance is similar
with original assembly version on different platforms. If not
specified, the code is compiled by GCC 4.6. The result is the "X
version / C version" ratio, and the less is better.
| run 100k times | cortex-a7 | cortex-a15 |
| use C as the base on each | (1.2Ghz) | (1.7Ghz) |
| CPU target | | |
|----------------------------+-----------+------------|
| Neon asm | 24% | 23% |
| Neon intrinsics (GCC 4.6) | 33% | 32% |
| Neon intrinsics (GCC 4.8) | 27% | 27% |
BUG=3850
R=andrew@webrtc.org, jridges@masque.com
Change-Id: Id6cd0671502fadbebd10b1f5493f5b16c988286f
Review URL: https://webrtc-codereview.appspot.com/27999004
Patch from Zhongwei Yao <zhongwei.yao@arm.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7802 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL fixes the problem described in issue 4021. In summary, of the
very first packet coming in to NetEq gets rejected because the RTP
payload type is unknown, subsequent GetAudio calls would trigger asserts
(in debug builds). The problem was that the first_packet_ variable was
reset and new_codec_ was set, even though the packet was discarded
further down the line. Now, these variables are modified after the
packet has been verified.
BUG=4021
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7724 4adac7df-926f-26a2-2b94-8c16560cd09d
This should speed up test execution on Android since only
the files needed by the test will be processed (instead
of the whole data + resources directories).
A few files for modules_unittests had to be explicitly added
for Android, since they were previously a part of the
add-whole-directories entries for the resources and data
directories.
BUG=webrtc:3741
TEST=Passing android+android_rel trybots.
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7694 4adac7df-926f-26a2-2b94-8c16560cd09d
The subclasses that need a state pointer should declare them---with
the right type, not void*, to get rid of all those casts.
Two small but not quite trivial cleanups are included because they
blocked the state_ removal:
- AudioDecoderG722Stereo now inherits directly from AudioDecoder
instead of being a subclass of AudioDecoderG722.
- AudioDecoder now has a CngDecoderInstance member function, which
is implemented only by AudioDecoderCng. This replaces the previous
practice of calling AudioDecoder::state() and casting the result
to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
plainly visible in the AudioDecoder class declaration.
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24169005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7644 4adac7df-926f-26a2-2b94-8c16560cd09d
Breaks Chrome compile:
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3861: 'LOG_FERR1': identifier not found
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3861: 'LOG_FERR1': identifier not found
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(169) : error C3867: 'webrtc::NetEqImpl::GetAudioInternal': function call missing argument list; use '&webrtc::NetEqImpl::GetAudioInternal' to create a pointer to member
...
> Remove the state_ member from AudioDecoder
>
> The subclasses that need a state pointer should declare them---with
> the right type, not void*, to get rid of all those casts.
>
> Two small but not quite trivial cleanups are included because they
> blocked the state_ removal:
>
> - AudioDecoderG722Stereo now inherits directly from AudioDecoder
> instead of being a subclass of AudioDecoderG722.
>
> - AudioDecoder now has a CngDecoderInstance member function, which
> is implemented only by AudioDecoderCng. This replaces the previous
> practice of calling AudioDecoder::state() and casting the result
> to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
> plainly visible in the AudioDecoder class declaration.
>
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/24169005TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30879005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7629 4adac7df-926f-26a2-2b94-8c16560cd09d
The subclasses that need a state pointer should declare them---with
the right type, not void*, to get rid of all those casts.
Two small but not quite trivial cleanups are included because they
blocked the state_ removal:
- AudioDecoderG722Stereo now inherits directly from AudioDecoder
instead of being a subclass of AudioDecoderG722.
- AudioDecoder now has a CngDecoderInstance member function, which
is implemented only by AudioDecoderCng. This replaces the previous
practice of calling AudioDecoder::state() and casting the result
to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
plainly visible in the AudioDecoder class declaration.
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24169005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7623 4adac7df-926f-26a2-2b94-8c16560cd09d
Since r7255, it could happen that an old packet would block the decoding
process until enough packet was received for the buffer to flush. This
CL fixes that by:
- Partially reverting r7255;
- Remove recent old packets before taking a decision for GetAudio;
- Remove all old packets after a packet has been extracted for decoding;
- Adding tests for reordered packets.
BUG=chrome:423985
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7612 4adac7df-926f-26a2-2b94-8c16560cd09d
Pick up the libvpx roll: https://codereview.chromium.org/674753002
Summary of changes (28d1981..d3db2ff/DEPS):
* third_party/android_tools 36bf7ac..ea50ccc
* third_party/boringssl 7ea8481..751e889
* third_party/icu 8ac906f..d8b2a9d
* third_party/libvpx efe9712..2e5ced5
* third_party/usrsctp/usrsctplib
* tools/gyp 1990:1991
* tools/swarming_client a57d7db..bcb3bc3
Clang is not updated in this roll.
Made the change getchar() --> getc(stdin) as seems like getchar() isn't supported on android anymore.
(getchar() was causing the error: undefined reference to '__srget')
Update rate control parameter in vp9 test.
R=andrew@webrtc.orgTBR=ajm@google.com
Review URL: https://webrtc-codereview.appspot.com/23229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7598 4adac7df-926f-26a2-2b94-8c16560cd09d
