Misc. small cleanups.
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks
All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1534193008
Cr-Commit-Position: refs/heads/master@{#11191}
diff --git a/webrtc/modules/audio_coding/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/acm2/acm_resampler.cc
index 5df87d2..d7ceb8a 100644
--- a/webrtc/modules/audio_coding/acm2/acm_resampler.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_resampler.cc
@@ -32,7 +32,6 @@
size_t out_capacity_samples,
int16_t* out_audio) {
size_t in_length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
- int out_length = out_freq_hz * num_audio_channels / 100;
if (in_freq_hz == out_freq_hz) {
if (out_capacity_samples < in_length) {
assert(false);
@@ -49,7 +48,7 @@
return -1;
}
- out_length =
+ int out_length =
resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
if (out_length == -1) {
LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "