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
I've kicked of a roll into Chromium with out the WebRtcVideoEngine2 change, to see if it was causing the roll problems, but re-landing in the meantime.
> Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine."
> content_browsertests started failing around the time the change landed and rolls are failing now.
> I'm going to try rolling this back, start a roll, and then re-land.
>
> > Set WebRtcVideoEngine2 as the WebRtcMediaEngine.
> >
> > Removes the experiment launching WebRTC-NewVideoAPI. This field trial
> > has shown no major regressions on Chrome Canary/Dev that haven't been
> > addressed, so enabling it in time before feature freeze.
> >
> > BUG=1788
> > R=mflodman@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/44759004
>
> TBR=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/43889004TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50459004
Cr-Commit-Position: refs/heads/master@{#8817}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8817 4adac7df-926f-26a2-2b94-8c16560cd09d
content_browsertests started failing around the time the change landed and rolls are failing now.
I'm going to try rolling this back, start a roll, and then re-land.
> Set WebRtcVideoEngine2 as the WebRtcMediaEngine.
>
> Removes the experiment launching WebRTC-NewVideoAPI. This field trial
> has shown no major regressions on Chrome Canary/Dev that haven't been
> addressed, so enabling it in time before feature freeze.
>
> BUG=1788
> R=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/44759004TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43889004
Cr-Commit-Position: refs/heads/master@{#8816}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8816 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
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44659004
Cr-Commit-Position: refs/heads/master@{#8801}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8801 4adac7df-926f-26a2-2b94-8c16560cd09d
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.
Also removes unnecessary casts to int32_t from int16_t.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48519004
Cr-Commit-Position: refs/heads/master@{#8800}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8800 4adac7df-926f-26a2-2b94-8c16560cd09d
We are using the Ooura FFT in a few places:
- AGC
- Transient suppression
- Noise suppression
The optimized OpenMAX DL FFT is considerably faster, but currently does
not compile everywhere, notably on iOS. This change will allow us to use
Openmax when possible and otherwise fall back to Ooura.
(Unfortunately, noise suppression won't be able to take advantage of it
since it's not C++. Upgrade time?)
R=aluebs@webrtc.org, mgraczyk@chromium.org
Review URL: https://webrtc-codereview.appspot.com/45789004
Cr-Commit-Position: refs/heads/master@{#8798}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8798 4adac7df-926f-26a2-2b94-8c16560cd09d
Removes a lot of code that prevents refactoring VideoCodingModule. Tests
covering the module should be TEST_Fs, and this looks like like fairly
unused code in general.
Adds a 'rtp_player' binary which performs a small subset.
BUG=4391
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44559004
Cr-Commit-Position: refs/heads/master@{#8787}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8787 4adac7df-926f-26a2-2b94-8c16560cd09d
