8 Commits

Author SHA1 Message Date
Henrik Kjellander
74640895fa audio_coding: rename interface -> include
BUG=webrtc:5095
R=henrik.lundin@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417173004 .

Cr-Commit-Position: refs/heads/master@{#10444}
2015-10-29 10:31:11 +00:00
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
kwiberg
98ab3a46d6 Don't link with audio codecs that we don't use
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.

This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means,
likely just the linker omitting compilation units with no incoming
references.

(This was previously landed as revisions 10046 and 10060, and got
reverted because it broke several of the Chromium FYI bots.)

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1368843003

Cr-Commit-Position: refs/heads/master@{#10127}
2015-10-01 04:54:29 +00:00
kwiberg
844a91081e Remove the preprocessor symbol WEBRTC_CODEC_PCM16 (it was always defined)
BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1336923002

Cr-Commit-Position: refs/heads/master@{#9955}
2015-09-16 16:42:26 +00:00
minyue@webrtc.org
41d2befe9f Limit RED audio payload to narrow band.
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
2015-03-23 12:58:17 +00:00
henrik.lundin@webrtc.org
f56c162310 Remove AudioCodingModule::Process()
An earlier change moved the encoding work from Process to
Add10MsData; process was just a no-op.

BUG=3520
COAUTHOR=kwiberg@webrtc.org
R=henrika@webrtc.org, minyue@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43439004

Cr-Commit-Position: refs/heads/master@{#8553}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8553 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 12:30:19 +00:00
minyue@webrtc.org
60fbd65482 Removing error triggered for disabling FEC on non-opus
A failure was triggered when one sets FEC status on a codec that does not support FEC. While it is definitely logical when one wants to enable it, it makes no good sense if one tries to disable it.

BUG=
R=tina.legrand@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7298 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 14:36:30 +00:00
minyue@webrtc.org
aa5ea1c0f9 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED
2. Add two new APIs to configure codec internal FEC

3. Add a test and listened to results. This is based modifying EncodeDecodeTest and deriving a new class from it.

New ACM gives good result.
Old ACM does not use NetEq 4, so FEC won't be decoded.

BUG=
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6233 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 15:16:51 +00:00