This change essentially divides AudioCodingModuleImpl into two parts:
one is the code related to managing codecs, now moved into CodecManager,
and the other is what remains in AudioCodingModuleImpl.
This change also removes AudioCodingModuleImpl::InitializeSender. The
function was essentially no-op, since it was always called immediately
after construction.
COAUTHOR=kwiberg@webrtc.org
BUG=4228
R=minyue@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51469004
Cr-Commit-Position: refs/heads/master@{#8893}
This is necessary unfortunately since there are a few places where DeRegisterModule does not reliably occur on the same thread.
BUG=4473
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42979004
Cr-Commit-Position: refs/heads/master@{#8891}
We want to crate the illusion that iSAC supports 48000 Hz decoding,
while in fact it outputs 32000 Hz. This is the iSAC fullband mode.
Currently this is (also) handled by higher layers, but in future
changes this will not be the case.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47809004
Cr-Commit-Position: refs/heads/master@{#8889}
All these tests crashed before r8811. These tests should've been with
that change but r8811 was pushed in before to make bots green.
BUG=1788, 1667
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48669004
Cr-Commit-Position: refs/heads/master@{#8881}
These are problematic when running with the default TSan
settings which has deadlock detection enabled.
Our bots still run with it disabled but we want to be
able to turn it back on, thus this is needed.
BUG=3911,4456
TESTED=
Successfully executed:
GYP_DEFINES="tsan=1 release_extra_cflags=-g use_allocator=none" webrtc/build/gyp_webrtc
ninja -C out/Release rtc_unittests
out/Release/rtc_unittests
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44899004
Cr-Commit-Position: refs/heads/master@{#8879}
Setting the member value output_will_be_muted_ in set_output_will_be_muted() was done before the lock.
This caused a data race.
BUG=4477
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44929004
Cr-Commit-Position: refs/heads/master@{#8877}
Also adds a framework for an AudioManager to be used by both sides (playout and recording).
This initial implementation only does very simple tasks like setting up the correct audio
mode (needed for correct volume behavior). Note that this CL is mainly about modifying
the volume. The added AudioManager is only a place holder for future work. I could have
done the same parts in the WebRtcAudioTrack class but feel that it is better to move these
parts to an AudioManager already at this stage.
The AudioManager supports Init() where actual audio changes are done (set audio mode etc.)
but it can also be used a simple "construct-and-store-audio-parameters" unit, which is the
case here. Hence, the AM now serves as the center for getting audio parameters and then inject
these into playout and recording sides. Previously, both sides acquired their own parameters
and that is more error prone.
BUG=NONE
TEST=AudioDeviceTest
R=perkj@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45829004
Cr-Commit-Position: refs/heads/master@{#8875}
Change internal indexing of registered decoders. It makes sense because payload type is unique, while ACM codec ID may not be. This is a step towards allowing for addition of external decoders.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44869004
Cr-Commit-Position: refs/heads/master@{#8867}
These checks would help catching double-deletes, forgetting to destroy
streams and also catch if VideoEngine has held on to any stale
references.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42929004
Cr-Commit-Position: refs/heads/master@{#8866}
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.
BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45549004
Cr-Commit-Position: refs/heads/master@{#8864}
If built-in Echo Cancellation is available on a device it is automatically enabled. The reason is that it in most cases performs better than the WebRTC software echo control for mobile. The drawback is that we can not develop, test and rollout the delay agnostic AEC (DA-AEC) on Android as for desktops.
This CL includes
- adding a media constraint to enable/disable DA-AEC.
- automatically turning on echo cancellation if DA-AEC is enabled.
- a fix in the AEC that enables delay estimation when DA-AEC is enabled, but delay metrics is disabled.
- sets the Config struct ReportedDelay, which controls DA-AEC internally in the AEC.
The test code to verify that it works in AppRTCDemo can be found here:
https://webrtc-codereview.appspot.com/50479004/
BUG=4472
TESTED=locally on N7, N6, Android One
R=glaznev@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48699004
Cr-Commit-Position: refs/heads/master@{#8861}
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
(WEBRTC_SPL_MUL_16_16(a, b) >> (c))
where the latter macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,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)
- minor cleanups like remove of unnecessary parentheses and style changes
BUG=3347, 3348, 3353
TESTED=locally on Linux for both fixed and floating point and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44799004
Cr-Commit-Position: refs/heads/master@{#8857}
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
(WEBRTC_SPL_MUL_16_16(a, b) >> (c))
where the latter macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,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)
- minor cleanups like remove of unnecessary parentheses and style changes
BUG=3348, 3353
TESTED=locally on Linux for both fixed and floating point and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49499004
Cr-Commit-Position: refs/heads/master@{#8853}
And add a constructor for creating an uninitialized Buffer of a
specified size.
(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48579004
Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
I'm splitting the timer functions in EventWrapper into a separate interface.
- Users of the timer functions have different needs than users of a generic event
- Providing a default implementation for EventWrapper that simply uses rtc::Event.
This means that clients of WebRTC that don't use the relatively few classes, typically rendering classes, that depend on the event timer functionality, also don't pull in dependencies on multimedia timers.
R=mflodman@webrtc.org, mflodman
BUG=
Review URL: https://webrtc-codereview.appspot.com/48599004
Cr-Commit-Position: refs/heads/master@{#8833}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8833 4adac7df-926f-26a2-2b94-8c16560cd09d
In SDP, RED audio codec has its own sample rate. Currently, we offer RED/8000 (8 kHz). But the actual send codec can violate this sample rate. The way to solve it is to introduce more RED payload types, e.g., RED/16000, RED/32000.
As a first step towards that, we, in this CL, limit the current RED (RED/8000) to work only with 8 kHz codecs.
BUG=3619
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43849004
Cr-Commit-Position: refs/heads/master@{#8830}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8830 4adac7df-926f-26a2-2b94-8c16560cd09d
A codec's packet-loss concealer is called once from NetEq before
decoding the first packet after a packet loss. The purpose is not to
use the PLC output, but to prepare the state of the decoder such that
it may recover faster after the loss. However, this effect is not
achieved by calling iSAC's PLC. Also, there are some problems with the
fixed-point implementation of the PLC (see the associated bug).
BUG=4423
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42849004
Cr-Commit-Position: refs/heads/master@{#8827}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8827 4adac7df-926f-26a2-2b94-8c16560cd09d
So in the HandleStreamFormatChange() callback, we need to re-initiate the playout as same as what we do in InitPlayout(). Here we merely copy those codes out from InitPlayout() into a new SetDesiredPlayoutFormat() function for the invoking from the two places.
Previously, HandleStreamFormatChange only re-creates the AudioConverter, which is not enough. We also need to reset the buffer size and refresh the latency.
BUG=4240
TEST=Manual Test
R=andrew@webrtc.org, henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36029004
Cr-Commit-Position: refs/heads/master@{#8815}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8815 4adac7df-926f-26a2-2b94-8c16560cd09d
