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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

27
webrtc/modules/audio_processing/audio_buffer.cc
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@ -19,9 +19,9 @@
namespace webrtc {
namespace {
const int kSamplesPer16kHzChannel = 160;
const int kSamplesPer32kHzChannel = 320;
const int kSamplesPer48kHzChannel = 480;
const size_t kSamplesPer16kHzChannel = 160;
const size_t kSamplesPer32kHzChannel = 320;
const size_t kSamplesPer48kHzChannel = 480;
int KeyboardChannelIndex(const StreamConfig& stream_config) {
if (!stream_config.has_keyboard()) {
@ -32,23 +32,22 @@ int KeyboardChannelIndex(const StreamConfig& stream_config) {
return stream_config.num_channels();
}
int NumBandsFromSamplesPerChannel(int num_frames) {
int num_bands = 1;
size_t NumBandsFromSamplesPerChannel(size_t num_frames) {
size_t num_bands = 1;
if (num_frames == kSamplesPer32kHzChannel ||
num_frames == kSamplesPer48kHzChannel) {
num_bands = rtc::CheckedDivExact(num_frames,
static_cast<int>(kSamplesPer16kHzChannel));
num_bands = rtc::CheckedDivExact(num_frames, kSamplesPer16kHzChannel);
}
return num_bands;
}
} // namespace
AudioBuffer::AudioBuffer(int input_num_frames,
AudioBuffer::AudioBuffer(size_t input_num_frames,
int num_input_channels,
int process_num_frames,
size_t process_num_frames,
int num_process_channels,
int output_num_frames)
size_t output_num_frames)
: input_num_frames_(input_num_frames),
num_input_channels_(num_input_channels),
proc_num_frames_(process_num_frames),
@ -345,20 +344,20 @@ void AudioBuffer::set_num_channels(int num_channels) {
num_channels_ = num_channels;
}
int AudioBuffer::num_frames() const {
size_t AudioBuffer::num_frames() const {
return proc_num_frames_;
}
int AudioBuffer::num_frames_per_band() const {
size_t AudioBuffer::num_frames_per_band() const {
return num_split_frames_;
}
int AudioBuffer::num_keyboard_frames() const {
size_t AudioBuffer::num_keyboard_frames() const {
// We don't resample the keyboard channel.
return input_num_frames_;
}
int AudioBuffer::num_bands() const {
size_t AudioBuffer::num_bands() const {
return num_bands_;
}