zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

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

Browse Source 
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

8
webrtc/modules/audio_processing/rms_level.h
View File

@ -11,6 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
#include <cstddef>
#include "webrtc/typedefs.h"
namespace webrtc {
@ -35,11 +37,11 @@ class RMSLevel {
void Reset();
// Pass each chunk of audio to Process() to accumulate the level.
void Process(const int16_t* data, int length);
void Process(const int16_t* data, size_t length);
// If all samples with the given |length| have a magnitude of zero, this is
// a shortcut to avoid some computation.
void ProcessMuted(int length);
void ProcessMuted(size_t length);
// Computes the RMS level over all data passed to Process() since the last
// call to RMS(). The returned value is positive but should be interpreted as
@ -48,7 +50,7 @@ class RMSLevel {
private:
float sum_square_;
int sample_count_;
size_t sample_count_;
};
} // namespace webrtc