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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/test/audio_processing_unittest.cc
View File

@ -129,21 +129,23 @@ void VerifyChannelsAreEqual(int16_t* stereo, int samples_per_channel) {
}
void SetFrameTo(AudioFrame* frame, int16_t value) {
for (int i = 0; i < frame->samples_per_channel_ * frame->num_channels_; ++i) {
for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
++i) {
frame->data_[i] = value;
}
}
void SetFrameTo(AudioFrame* frame, int16_t left, int16_t right) {
ASSERT_EQ(2, frame->num_channels_);
for (int i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
frame->data_[i] = left;
frame->data_[i + 1] = right;
}
}
void ScaleFrame(AudioFrame* frame, float scale) {
for (int i = 0; i < frame->samples_per_channel_ * frame->num_channels_; ++i) {
for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
++i) {
frame->data_[i] = FloatS16ToS16(frame->data_[i] * scale);
}
}
@ -676,13 +678,18 @@ void ApmTest::ProcessDelayVerificationTest(int delay_ms, int system_delay_ms,
}
// Calculate expected delay estimate and acceptable regions. Further,
// limit them w.r.t. AEC delay estimation support.
const int samples_per_ms = std::min(16, frame_->samples_per_channel_ / 10);
const size_t samples_per_ms =
std::min(static_cast<size_t>(16), frame_->samples_per_channel_ / 10);
int expected_median = std::min(std::max(delay_ms - system_delay_ms,
delay_min), delay_max);
int expected_median_high = std::min(std::max(
expected_median + 96 / samples_per_ms, delay_min), delay_max);
int expected_median_low = std::min(std::max(
expected_median - 96 / samples_per_ms, delay_min), delay_max);
int expected_median_high = std::min(
std::max(expected_median + static_cast<int>(96 / samples_per_ms),
delay_min),
delay_max);
int expected_median_low = std::min(
std::max(expected_median - static_cast<int>(96 / samples_per_ms),
delay_min),
delay_max);
// Verify delay metrics.
int median;
int std;
@ -998,8 +1005,8 @@ TEST_F(ApmTest, DISABLED_EchoCancellationReportsCorrectDelays) {
2,
false);
// Sampling frequency dependent variables.
const int num_ms_per_block = std::max(4,
640 / frame_->samples_per_channel_);
const int num_ms_per_block =
std::max(4, static_cast<int>(640 / frame_->samples_per_channel_));
const int delay_min_ms = -kLookaheadBlocks * num_ms_per_block;
const int delay_max_ms = (kMaxDelayBlocks - 1) * num_ms_per_block;