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

15
webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
View File

@ -43,7 +43,7 @@ struct TestParameters {
class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
protected:
static const int kTimeStepMs = 10;
static const int kMaxBlockSize = 480; // 10 ms @ 48 kHz.
static const size_t kMaxBlockSize = 480; // 10 ms @ 48 kHz.
static const uint8_t kPayloadTypeMono = 95;
static const uint8_t kPayloadTypeMulti = 96;
@ -52,7 +52,8 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
sample_rate_hz_(GetParam().sample_rate),
samples_per_ms_(sample_rate_hz_ / 1000),
frame_size_ms_(GetParam().frame_size),
frame_size_samples_(frame_size_ms_ * samples_per_ms_),
frame_size_samples_(
static_cast<size_t>(frame_size_ms_ * samples_per_ms_)),
output_size_samples_(10 * samples_per_ms_),
rtp_generator_mono_(samples_per_ms_),
rtp_generator_(samples_per_ms_),
@ -212,7 +213,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
}
NetEqOutputType output_type;
// Get audio from mono instance.
int samples_per_channel;
size_t samples_per_channel;
int num_channels;
EXPECT_EQ(NetEq::kOK,
neteq_mono_->GetAudio(kMaxBlockSize, output_,
@ -242,8 +243,8 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
const int sample_rate_hz_;
const int samples_per_ms_;
const int frame_size_ms_;
const int frame_size_samples_;
const int output_size_samples_;
const size_t frame_size_samples_;
const size_t output_size_samples_;
NetEq* neteq_mono_;
NetEq* neteq_;
test::RtpGenerator rtp_generator_mono_;
@ -256,8 +257,8 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
int16_t* output_multi_channel_;
WebRtcRTPHeader rtp_header_mono_;
WebRtcRTPHeader rtp_header_;
int payload_size_bytes_;
int multi_payload_size_bytes_;
size_t payload_size_bytes_;
size_t multi_payload_size_bytes_;
int last_send_time_;
int last_arrival_time_;
rtc::scoped_ptr<test::InputAudioFile> input_file_;