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.cc b/webrtc/modules/audio_coding/test/opus_test.cc
index a68db91..466db9f 100644
--- a/webrtc/modules/audio_coding/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/test/opus_test.cc
@@ -206,16 +206,16 @@
}
void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
- int frame_length, int percent_loss) {
+ size_t frame_length, int percent_loss) {
AudioFrame audio_frame;
int32_t out_freq_hz_b = out_file_.SamplingFrequency();
- const int kBufferSizeSamples = 480 * 12 * 2; // Can hold 120 ms stereo audio.
+ const size_t kBufferSizeSamples = 480 * 12 * 2; // 120 ms stereo audio.
int16_t audio[kBufferSizeSamples];
int16_t out_audio[kBufferSizeSamples];
int16_t audio_type;
- int written_samples = 0;
- int read_samples = 0;
- int decoded_samples = 0;
+ size_t written_samples = 0;
+ size_t read_samples = 0;
+ size_t decoded_samples = 0;
bool first_packet = true;
uint32_t start_time_stamp = 0;
@@ -268,14 +268,14 @@
// Sometimes we need to loop over the audio vector to produce the right
// number of packets.
- int loop_encode = (written_samples - read_samples) /
+ size_t loop_encode = (written_samples - read_samples) /
(channels * frame_length);
if (loop_encode > 0) {
- const int kMaxBytes = 1000; // Maximum number of bytes for one packet.
+ const size_t kMaxBytes = 1000; // Maximum number of bytes for one packet.
size_t bitstream_len_byte;
uint8_t bitstream[kMaxBytes];
- for (int i = 0; i < loop_encode; i++) {
+ for (size_t i = 0; i < loop_encode; i++) {
int bitstream_len_byte_int = WebRtcOpus_Encode(
(channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
&audio[read_samples], frame_length, kMaxBytes, bitstream);
@@ -326,7 +326,7 @@
first_packet = false;
start_time_stamp = rtp_timestamp_;
}
- rtp_timestamp_ += frame_length;
+ rtp_timestamp_ += static_cast<uint32_t>(frame_length);
read_samples += frame_length * channels;
}
if (read_samples == written_samples) {
@@ -344,8 +344,7 @@
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
// Write stand-alone speech to file.
- out_file_standalone_.Write10MsData(
- out_audio, static_cast<size_t>(decoded_samples) * channels);
+ out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);
if (audio_frame.timestamp_ > start_time_stamp) {
// Number of channels should be the same for both stand-alone and