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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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9
webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
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@ -23,8 +23,10 @@ AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
: block_duration_ms_(block_duration_ms),
input_sampling_khz_(input_sampling_khz),
output_sampling_khz_(output_sampling_khz),
input_length_sample_(block_duration_ms_ * input_sampling_khz_),
output_length_sample_(block_duration_ms_ * output_sampling_khz_),
input_length_sample_(
static_cast<size_t>(block_duration_ms_ * input_sampling_khz_)),
output_length_sample_(
static_cast<size_t>(block_duration_ms_ * output_sampling_khz_)),
data_pointer_(0),
loop_length_samples_(0),
max_bytes_(0),
@ -65,8 +67,7 @@ void AudioCodecSpeedTest::SetUp() {
memcpy(&in_data_[loop_length_samples_], &in_data_[0],
input_length_sample_ * channels_ * sizeof(int16_t));
max_bytes_ =
static_cast<size_t>(input_length_sample_ * channels_ * sizeof(int16_t));
max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
out_data_.reset(new int16_t[output_length_sample_ * channels_]);
bit_stream_.reset(new uint8_t[max_bytes_]);

4
webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
View File

@ -55,10 +55,10 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
int output_sampling_khz_;
// Number of samples-per-channel in a frame.
int input_length_sample_;
size_t input_length_sample_;
// Expected output number of samples-per-channel in a frame.
int output_length_sample_;
size_t output_length_sample_;
rtc::scoped_ptr<int16_t[]> in_data_;
rtc::scoped_ptr<int16_t[]> out_data_;