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/test/opus_test.h b/webrtc/modules/audio_coding/test/opus_test.h
index 090c8fa..88ef0ec 100644
--- a/webrtc/modules/audio_coding/test/opus_test.h
+++ b/webrtc/modules/audio_coding/test/opus_test.h
@@ -31,7 +31,10 @@
void Perform();
private:
- void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
+ void Run(TestPackStereo* channel,
+ int channels,
+ int bitrate,
+ size_t frame_length,
int percent_loss = 0);
void OpenOutFile(int test_number);
@@ -44,7 +47,7 @@
PCMFile out_file_standalone_;
int counter_;
uint8_t payload_type_;
- int rtp_timestamp_;
+ uint32_t rtp_timestamp_;
acm2::ACMResampler resampler_;
WebRtcOpusEncInst* opus_mono_encoder_;
WebRtcOpusEncInst* opus_stereo_encoder_;