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_processing/audio_processing_impl_locking_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
index cd4b31c..7e96774 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
@@ -478,7 +478,7 @@
ASSERT_GT(amplitude, 0);
ASSERT_LE(amplitude, 32767);
for (int ch = 0; ch < frame->num_channels_; ch++) {
- for (int k = 0; k < static_cast<int>(frame->samples_per_channel_); k++) {
+ for (size_t k = 0; k < frame->samples_per_channel_; k++) {
// Store random 16 bit number between -(amplitude+1) and
// amplitude.
frame->data_[k * ch] =
