Commit Graph

7 Commits

Author SHA1 Message Date
dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00
439a4c49f9 Add an output capacity parameter to ACMResampler::Resample10Msec()
Also adding a unit tests to make sure that a desired output frequency
of 0 passed to AudioCodingModule::PlayoutData10Ms() is invalid.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 19:05:33 +00:00
f5a33f145b Resampler modifications in preparation for arbitrary audioproc rates.
- Templatize PushResampler to support int16 and float.
- Add a helper method to PushSincResampler to compute the algorithmic
delay.

This is a prerequisite of:
http://review.webrtc.org/9919004/

BUG=2894
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-19 00:32:07 +00:00
40ee3d07ed Consolidate audio conversion from Channel and TransmitMixer.
Replace the two versions with a single DownConvertToCodecFormat. As
mentioned in comments, this could be further consolidated with
RemixAndResample but we should write a full audio converter class in
that case.

Along the way:
- Fix the bug present in Channel::Demultiplex with mono input and a
stereo codec.
- Remove the 32 kHz max from the OnDataAvailable path. This avoids a
48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we
get a straight pass-through to ACM. The 32 kHz conversion is still
needed in the RecordedDataIsAvailable path until APM natively supports
48 kHz.
- Merge resampler improvements from ACM1 to ACM2. This allows ACM to
handle 44.1 kHz audio passed to VoE and was originally done here:
https://webrtc-codereview.appspot.com/1590004
- Reuse the RemixAndResample unit tests for DownConvertToCodecFormat.
- Remove unused functions from utility.cc.

BUG=3155,3000,b/12867572
TESTED=voe_cmd_test using both the OnDataAvailable and
RecordedDataIsAvailable paths, with a captured audio format of all
combinations of {44.1,48} kHz and {1,2} channels, running through all
codecs, and finally using both ACM1 and ACM2.

R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-03 21:56:01 +00:00
6d5d248075 Wrap ACM2 code inside acm2 namespace. This gurantees that one ACM would not use components of others by accident.
BUG=
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2344004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4933 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-06 04:47:28 +00:00
48af652ea5 Prepare to compile ACM1 and ACM2.
ACM1 code is wrapped in namespace acm1. Inculde paths and define guards of ACM2 source codes are corrected. gypi file of ACM2 is changed so that ACM1 will later on depends on ACM2.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2206004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4743 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-13 23:06:59 +00:00
7959e16cc2 ACM2 integration with NetEq 4.
nack{.cc, .h, _unittest.cc} are basically copies from main/source/ folder, with cpplint warning cleaned up.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4736 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 18:30:26 +00:00