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Update a ton of audio code to use size_t more correctly and in general reduce

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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}
This commit is contained in:
Peter Kasting
2015-08-24 14:52:23 -07:00
parent b594041ec8
commit dce40cf804
471 changed files with 3716 additions and 3499 deletions
Download Patch File Download Diff File Expand all files Collapse all files

36
webrtc/modules/audio_coding/codecs/g711/g711_interface.c
View File

@ -12,40 +12,40 @@
#include "g711_interface.h"
#include "webrtc/typedefs.h"
int16_t WebRtcG711_EncodeA(const int16_t* speechIn,
int16_t len,
uint8_t* encoded) {
int n;
size_t WebRtcG711_EncodeA(const int16_t* speechIn,
size_t len,
uint8_t* encoded) {
size_t n;
for (n = 0; n < len; n++)
encoded[n] = linear_to_alaw(speechIn[n]);
return len;
}
int16_t WebRtcG711_EncodeU(const int16_t* speechIn,
int16_t len,
uint8_t* encoded) {
int n;
size_t WebRtcG711_EncodeU(const int16_t* speechIn,
size_t len,
uint8_t* encoded) {
size_t n;
for (n = 0; n < len; n++)
encoded[n] = linear_to_ulaw(speechIn[n]);
return len;
}
int16_t WebRtcG711_DecodeA(const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType) {
int n;
size_t WebRtcG711_DecodeA(const uint8_t* encoded,
size_t len,
int16_t* decoded,
int16_t* speechType) {
size_t n;
for (n = 0; n < len; n++)
decoded[n] = alaw_to_linear(encoded[n]);
*speechType = 1;
return len;
}
int16_t WebRtcG711_DecodeU(const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType) {
int n;
size_t WebRtcG711_DecodeU(const uint8_t* encoded,
size_t len,
int16_t* decoded,
int16_t* speechType) {
size_t n;
for (n = 0; n < len; n++)
decoded[n] = ulaw_to_linear(encoded[n]);
*speechType = 1;