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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

15
webrtc/modules/audio_processing/include/audio_processing.h
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@ -311,7 +311,7 @@ class AudioProcessing {
//
// TODO(mgraczyk): Remove once clients are updated to use the new interface.
virtual int ProcessStream(const float* const* src,
int samples_per_channel,
size_t samples_per_channel,
int input_sample_rate_hz,
ChannelLayout input_layout,
int output_sample_rate_hz,
@ -357,7 +357,7 @@ class AudioProcessing {
// of |data| points to a channel buffer, arranged according to |layout|.
// TODO(mgraczyk): Remove once clients are updated to use the new interface.
virtual int AnalyzeReverseStream(const float* const* data,
int samples_per_channel,
size_t samples_per_channel,
int rev_sample_rate_hz,
ChannelLayout layout) = 0;
@ -510,8 +510,8 @@ class StreamConfig {
int num_channels() const { return num_channels_; }
bool has_keyboard() const { return has_keyboard_; }
int num_frames() const { return num_frames_; }
int num_samples() const { return num_channels_ * num_frames_; }
size_t num_frames() const { return num_frames_; }
size_t num_samples() const { return num_channels_ * num_frames_; }
bool operator==(const StreamConfig& other) const {
return sample_rate_hz_ == other.sample_rate_hz_ &&
@ -522,14 +522,15 @@ class StreamConfig {
bool operator!=(const StreamConfig& other) const { return !(*this == other); }
private:
static int calculate_frames(int sample_rate_hz) {
return AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000;
static size_t calculate_frames(int sample_rate_hz) {
return static_cast<size_t>(
AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000);
}
int sample_rate_hz_;
int num_channels_;
bool has_keyboard_;
int num_frames_;
size_t num_frames_;
};
class ProcessingConfig {