zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Misc. small cleanups.

Browse Source 
* 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
Download Patch File Download Diff File Expand all files Collapse all files

21
webrtc/modules/audio_coding/test/opus_test.cc
View File

@ -206,16 +206,16 @@ void OpusTest::Perform() {
}
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 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
// 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 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
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 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
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

7
webrtc/modules/audio_coding/test/opus_test.h
View File

@ -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_;