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/modules/utility/source/file_player_impl.cc b/webrtc/modules/utility/source/file_player_impl.cc
index df6a5bf..8c94caa 100644
--- a/webrtc/modules/utility/source/file_player_impl.cc
+++ b/webrtc/modules/utility/source/file_player_impl.cc
@@ -95,7 +95,7 @@
int32_t FilePlayerImpl::Get10msAudioFromFile(
int16_t* outBuffer,
- int& lengthInSamples,
+ size_t& lengthInSamples,
int frequencyInHz)
{
if(_codec.plfreq == 0)
@@ -127,8 +127,7 @@
return 0;
}
// One sample is two bytes.
- unresampledAudioFrame.samples_per_channel_ =
- (uint16_t)lengthInBytes >> 1;
+ unresampledAudioFrame.samples_per_channel_ = lengthInBytes >> 1;
} else {
// Decode will generate 10 ms of audio data. PlayoutAudioData(..)
@@ -156,14 +155,14 @@
}
}
- int outLen = 0;
+ size_t outLen = 0;
if(_resampler.ResetIfNeeded(unresampledAudioFrame.sample_rate_hz_,
frequencyInHz, 1))
{
LOG(LS_WARNING) << "Get10msAudioFromFile() unexpected codec.";
// New sampling frequency. Update state.
- outLen = frequencyInHz / 100;
+ outLen = static_cast<size_t>(frequencyInHz / 100);
memset(outBuffer, 0, outLen * sizeof(int16_t));
return 0;
}
@@ -177,7 +176,7 @@
if(_scaling != 1.0)
{
- for (int i = 0;i < outLen; i++)
+ for (size_t i = 0;i < outLen; i++)
{
outBuffer[i] = (int16_t)(outBuffer[i] * _scaling);
}