Reasons for removing:
- Feels like a complete hack IMHO.
- Not used by any client.
- Unclear functionality regarding time stamp, marker bit etc.
- Causes several issues in tests due to a bad design which mainly depends on the fact that this API "breaks" an ongoing data/packet flow and it complicates the threading model and creates risks for deadlock and memory corruption. Not worth trying to fix given the very unclear benefit of maintaining the API. Better to remove the API instead.
- We also see lots of TSan races and memcheck errors related to this API.
BUG=2296,2240
R=mflodman@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5574 4adac7df-926f-26a2-2b94-8c16560cd09d
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
Analogous to the recent libjingle change: http://cl/54929753-p10.
This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather
than scoped_array and scoped_ptr_malloc respectively.
- Add Chromium's template-based COMPILE_ASSERT. We didn't have this
previously in order to support the macro in C. Instead, move the
existing macro to compile_assert_c.h.
- Additionally copy the move.h and template_util.h depedencies and add
the WARN_UNUSED_RESULT macro.
- Leave scoped_array and scoped_ptr_malloc for now, but mark as
deprecated.
- Remove scoped_ptr foo(NULL) use. The default constructor handles it.
- Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc.
- Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove
some repeated code.
TESTED=trybots
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2449005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
- Remove unneeded #defines and switch the remainder to consts.
- All AudioProcessing components are disabled by default, so remove
explicit disables.
- AudioProcessing uses a rational 16 kHz mono default, so no need to
explictly initialize.
- Add assert(false) to real-time errors which should not occur.
TESTED=trybots
R=bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2253005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4924 4adac7df-926f-26a2-2b94-8c16560cd09d
r4374 was mistakenly committed to stable, so this is to re-merge back to trunk.
Store the sequence number in StopSend() and resume it in StartSend().
When restarting the microphone device, we call StopSend() first, then
StartSend() later. Since we reset sequence number in StopSend(), it sometimes
causes libSRTP to complain about packets being replayed. Libjingle work around
it by caching the sequence number in WebRtcVoiceEngine.cc, and call
SetInitSequenceNumber() to resume the sequence number before StartSend().Store the sequence number in StopSend() and resume it in StartSend().
When restarting the microphone device, we call StopSend() first, then
StartSend() later. Since we reset sequence number in StopSend(), it sometimes
causes libSRTP to complain about packets being replayed. Libjingle work around
it by caching the sequence number in WebRtcVoiceEngine.cc, and call
SetInitSequenceNumber() to resume the sequence number before StartSend().
This patch fixes this problem by storing the sequence number in StopSend(), and
resume it in StartSend(). So that we can remove the workaround in libjingle.
BUG=2102
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1922004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4451 4adac7df-926f-26a2-2b94-8c16560cd09d
r4326 was mistakenly committed to stable, so this is to re-merge back to trunk.
Fixed the AGC and interface problems on the new path.
In order to make the AGC work properly, we need to cache the volume value passed
by the callback, compare it with the value returned by
shared->transmit_mixer()->CaptureLevel(). If they are the same, we need to
return 0 to indicate no volume needs changing, otherwise return the new volume.
By doing this, we avoid setting the volume all the same, which allows the users
to change the volume manually.
This patch also fixes some minor issues with the interfaces too: make the int
channel[] const, and correct the order of the input params in
channel::Demultiplex.
R=tommi@webrtc.org
BUG=[2134]
TEST=compile && manual AGC test
Review URL: https://webrtc-codereview.appspot.com/1921004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4450 4adac7df-926f-26a2-2b94-8c16560cd09d
r4326 was mistakenly committed to stable, so this is to re-merge back to trunk.
Add new interface to support multiple sources in webrtc.
CaptureData() will be called by chrome with a flag |need_audio_processing| to
indicate if the data needs to be processed by APM or not. Different from the old
interface that will send the data to all voe channels, the new interface will
specify a list of voe channels that the data is demultiplexing to.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4449 4adac7df-926f-26a2-2b94-8c16560cd09d
TBR=perkj
BUG=227036 (in crbug.com)
TEST=out\Debug\voe_auto_test.exe --automated --gtest_filter=Dtmf* where I
manually modified the test and used 100 as new PT (which I first verified was
already used by CN, 48000).
BUG=
Review URL: https://webrtc-codereview.appspot.com/1319010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3859 4adac7df-926f-26a2-2b94-8c16560cd09d
