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}
diff --git a/webrtc/common_audio/audio_converter.cc b/webrtc/common_audio/audio_converter.cc
index 7e043b7..624c38d 100644
--- a/webrtc/common_audio/audio_converter.cc
+++ b/webrtc/common_audio/audio_converter.cc
@@ -24,8 +24,8 @@
class CopyConverter : public AudioConverter {
public:
- CopyConverter(int src_channels, int src_frames, int dst_channels,
- int dst_frames)
+ CopyConverter(int src_channels, size_t src_frames, int dst_channels,
+ size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
~CopyConverter() override {};
@@ -41,15 +41,15 @@
class UpmixConverter : public AudioConverter {
public:
- UpmixConverter(int src_channels, int src_frames, int dst_channels,
- int dst_frames)
+ UpmixConverter(int src_channels, size_t src_frames, int dst_channels,
+ size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
~UpmixConverter() override {};
void Convert(const float* const* src, size_t src_size, float* const* dst,
size_t dst_capacity) override {
CheckSizes(src_size, dst_capacity);
- for (int i = 0; i < dst_frames(); ++i) {
+ for (size_t i = 0; i < dst_frames(); ++i) {
const float value = src[0][i];
for (int j = 0; j < dst_channels(); ++j)
dst[j][i] = value;
@@ -59,8 +59,8 @@
class DownmixConverter : public AudioConverter {
public:
- DownmixConverter(int src_channels, int src_frames, int dst_channels,
- int dst_frames)
+ DownmixConverter(int src_channels, size_t src_frames, int dst_channels,
+ size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
}
~DownmixConverter() override {};
@@ -69,7 +69,7 @@
size_t dst_capacity) override {
CheckSizes(src_size, dst_capacity);
float* dst_mono = dst[0];
- for (int i = 0; i < src_frames(); ++i) {
+ for (size_t i = 0; i < src_frames(); ++i) {
float sum = 0;
for (int j = 0; j < src_channels(); ++j)
sum += src[j][i];
@@ -80,8 +80,8 @@
class ResampleConverter : public AudioConverter {
public:
- ResampleConverter(int src_channels, int src_frames, int dst_channels,
- int dst_frames)
+ ResampleConverter(int src_channels, size_t src_frames, int dst_channels,
+ size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
resamplers_.reserve(src_channels);
for (int i = 0; i < src_channels; ++i)
@@ -136,9 +136,9 @@
};
rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
- int src_frames,
+ size_t src_frames,
int dst_channels,
- int dst_frames) {
+ size_t dst_frames) {
rtc::scoped_ptr<AudioConverter> sp;
if (src_channels > dst_channels) {
if (src_frames != dst_frames) {
@@ -182,8 +182,8 @@
dst_channels_(0),
dst_frames_(0) {}
-AudioConverter::AudioConverter(int src_channels, int src_frames,
- int dst_channels, int dst_frames)
+AudioConverter::AudioConverter(int src_channels, size_t src_frames,
+ int dst_channels, size_t dst_frames)
: src_channels_(src_channels),
src_frames_(src_frames),
dst_channels_(dst_channels),
@@ -192,8 +192,8 @@
}
void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
- CHECK_EQ(src_size, checked_cast<size_t>(src_channels() * src_frames()));
- CHECK_GE(dst_capacity, checked_cast<size_t>(dst_channels() * dst_frames()));
+ CHECK_EQ(src_size, src_channels() * src_frames());
+ CHECK_GE(dst_capacity, dst_channels() * dst_frames());
}
} // namespace webrtc
