The playout mode in NetEq can still be set through the constructor
configuration.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1362943004
Cr-Commit-Position: refs/heads/master@{#10089}
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}
This should be safe to land now that issue 4143 was resolved (in r8492).
This change effectively reverts 8488.
TBR=kwiberg@webrtc.org
Original commit message:
This CL changes the way the decoder sample rate is set and updated. In
practice, it only concerns the iSAC (float) codec.
One single iSAC decoder instance is used for both wideband and
super-wideband decoding, and the instance must be told to switch
output frequency if the payload type changes. This used to be done
through a call to UpdateDecoderSampleRate, but is now instead done in
the Decode call as an extra parameter.
Review URL: https://webrtc-codereview.appspot.com/39289004
Cr-Commit-Position: refs/heads/master@{#8496}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8496 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL changes the way the decoder sample rate is set and updated. In
practice, it only concerns the iSAC (float) codec.
One single iSAC decoder instance is used for both wideband and
super-wideband decoding, and the instance must be told to switch
output frequency if the payload type changes. This used to be done
through a call to UpdateDecoderSampleRate, but is now instead done in
the Decode call as an extra parameter.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34349004
Cr-Commit-Position: refs/heads/master@{#8476}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8476 4adac7df-926f-26a2-2b94-8c16560cd09d
The problem was that if the output sample rate is changed such from one
where no resampling is needed to a rate that requires resampling, the
first output from the resampler will contain an onset period. The
solution provided in this CL is to keep a copy of the last output frame
in ACM, and if the resampler is engaged, it will be primed with this
old frame before resampling the current frame.
BUG=3919
R=bjornv@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7479 4adac7df-926f-26a2-2b94-8c16560cd09d
This change prepares for switching default background noise (bgn) mode
from on to off. The actual switch will be done later.
In this change, the bgn mode is included as a setting in NetEq's config
struct. We're also removing the connection between playout modes and
bgn modes in ACM. In practice this means that bgn mode will change from
off to on for streaming mode, but since the playout modes are not used
it does not matter.
BUG=3519
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6843 4adac7df-926f-26a2-2b94-8c16560cd09d
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.
Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.
Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.
BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc
Review URL: https://webrtc-codereview.appspot.com/14559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
