In some cases it was not possible to set another payload type for CNG
than the default one. This CL fixes this. The problem was also
dependent on whether the comfort noise codec was registered before or
after the speech codec.
A test is implement to expose the bug, registering comfort noise at a
non-default payload type, and both before and after the speech codec.
BUG=4228
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35199004
Cr-Commit-Position: refs/heads/master@{#8380}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8380 4adac7df-926f-26a2-2b94-8c16560cd09d
There are only 2 things that prevent the output to be bit-exact:
* The zero initialization of the postfilter_mask_ and high_pass_postfilter_mask_, which only afects the first blocks.
* The re-tuning of the target presence estimation, since only the bins between low_average_start_bin_ and high_average_end_bin_ are of interest.
This latter was not taken into account before.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35139004
Cr-Commit-Position: refs/heads/master@{#8368}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8368 4adac7df-926f-26a2-2b94-8c16560cd09d
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
Some other minor code cleanup also exists.
BUG=3348, 3353
TESTED=locally on Mac and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34179004
Cr-Commit-Position: refs/heads/master@{#8358}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8358 4adac7df-926f-26a2-2b94-8c16560cd09d
Removes lock-order inversion formed by RTPSenderAudio->RTPSender calls
by doing a lot shorter locking which fetches a current state of
RTPSenderAudio variables before sending.
Thread annotates locked variables and removes one lock in
RTPSenderAudio, bonus fixes data races reported in voe_auto_test
--automated under TSan (DTMF data race).
Also includes some bonus cleanup of RTPSenderVideo which removes the
send critsect completely as all methods using it was always called
from RTPSender under its send_critsect.
R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
BUG=3001, chromium:454654
Review URL: https://webrtc-codereview.appspot.com/41869004
Cr-Commit-Position: refs/heads/master@{#8348}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8348 4adac7df-926f-26a2-2b94-8c16560cd09d
Moving functionality to get max payload length from default RTP module
to the payload router.
I'll make a follow up CL changing asserts to DCHECK in rtp_rtcp_impl.cc.
BUG=769
TEST=New unittest and existing sender mtu test
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36119004
Cr-Commit-Position: refs/heads/master@{#8345}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8345 4adac7df-926f-26a2-2b94-8c16560cd09d
This change switches from the old codec wrapper ACMISAC to the new
AudioEncoderIsac wrapped in an ACMGenericCodecWrapper.
This is also the CL where the old codec for producing redundancy (RED)
is inactivated. All RED payloads are now produces through the
AudioEncoderCopyRed or AudioEncoderIsacRed classes.
BUG=4228
TEST=Please, try the iSAC codec extensively.
COAUTHOR=kwiberg@webrtc.orgR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33249005
Cr-Commit-Position: refs/heads/master@{#8342}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8342 4adac7df-926f-26a2-2b94-8c16560cd09d
ViECapturer is always calling DeliverFrame with an empty CSRC vector, so
this is basically a dead path already. I added a DCHECK in ViEEncoder to
verify this is true.
BUG=769
TEST=Manually verified in Chromium.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39059004
Cr-Commit-Position: refs/heads/master@{#8335}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8335 4adac7df-926f-26a2-2b94-8c16560cd09d
This is a big refactoring of the existing C++/JNI/Java support for audio recording in native WebRTC:
- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioRecord (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup
Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).
BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33969004
Cr-Commit-Position: refs/heads/master@{#8325}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8325 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL performs the following renames of targets to
make GYP and GN more unified and make the targets that
have the same name as the module and include the external
render/capture implementation (the internal one is only
used by WebRTC tests).
This makes it natural to declare dependencies in GN
without having to specify the target.
Summary of the renames:
GYP:
video_render_module_impl -> video_render (new target)
video_capture_module_impl -> video_capture (new target)
GN:
video_capture -> video_capture_module (now identical to the GYP target)
video_capture_impl -> video_capture
video_render -> video_render_module (now identical to the GYP target)
video_render_impl -> video_render
BUG=456815
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35099004
Cr-Commit-Position: refs/heads/master@{#8323}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8323 4adac7df-926f-26a2-2b94-8c16560cd09d
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
This makes it possible to build more flexible simulations, and makes it easier to implement bi-directional simulations. This also removes support for generating baseline files and comparing against a baseline, which hasn't turned out to be particuarly useful.
BUG=4173
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35069004
Cr-Commit-Position: refs/heads/master@{#8311}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8311 4adac7df-926f-26a2-2b94-8c16560cd09d
This makes it possible to build more flexible simulations, and makes it easier to implement bi-directional simulations. This also removes support for generating baseline files and comparing against a baseline, which hasn't turned out to be particuarly useful.
BUG=4173
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34989004
Cr-Commit-Position: refs/heads/master@{#8297}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8297 4adac7df-926f-26a2-2b94-8c16560cd09d
We apparently hit an obscure problem on mac where seemingly an unaligned mutex causes memory corruption.
The effect was that the |modules_| list became corrupt and we crashed. At this point I'm not exactly
sure what the alignment requirements are but for now, I've fixed up the layout in a way that doesn't cause these same issues.
I'm also changing auto->proper type at the request of drive by reviewers from my previous cl in the same file.
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38989004
Cr-Commit-Position: refs/heads/master@{#8286}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8286 4adac7df-926f-26a2-2b94-8c16560cd09d
