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Misc. small cleanups.

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* 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}
This commit is contained in:
pkasting
2016-01-08 13:50:27 -08:00
committed by Commit bot
parent 5de688ed34
commit 25702cb162
51 changed files with 445 additions and 608 deletions
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7
webrtc/modules/audio_coding/test/opus_test.h
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@ -31,7 +31,10 @@ class OpusTest : public ACMTest {
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 @@ class OpusTest : public ACMTest {
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_;