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

Use backticks not vertical bars to denote variables in comments for /modules/audio_processing

Browse Source 
Bug: webrtc:12338
Change-Id: I85bff694dd2ead83c939c4d1945eff82e1296001
No-Presubmit: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227161
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34690}
This commit is contained in:
Artem Titov
2021-07-28 20:50:03 +02:00
committed by WebRTC LUCI CQ
parent dc6801c618
commit 0b489303d2
102 changed files with 483 additions and 483 deletions
Download Patch File Download Diff File Expand all files Collapse all files

14
modules/audio_processing/audio_processing_unittest.cc
View File

@ -321,10 +321,10 @@ void OpenFileAndReadMessage(const std::string& filename, MessageLite* msg) {
// Reads a 10 ms chunk of int16 interleaved audio from the given (assumed
// stereo) file, converts to deinterleaved float (optionally downmixing) and
// returns the result in |cb|. Returns false if the file ended (or on error) and
// returns the result in `cb`. Returns false if the file ended (or on error) and
// true otherwise.
//
// |int_data| and |float_data| are just temporary space that must be
// `int_data` and `float_data` are just temporary space that must be
// sufficiently large to hold the 10 ms chunk.
bool ReadChunk(FILE* file,
int16_t* int_data,
@ -596,7 +596,7 @@ void ApmTest::ProcessDelayVerificationTest(int delay_ms,
int system_delay_ms,
int delay_min,
int delay_max) {
// The |revframe_| and |frame_| should include the proper frame information,
// The `revframe_` and `frame_` should include the proper frame information,
// hence can be used for extracting information.
Int16FrameData tmp_frame;
std::queue<Int16FrameData*> frame_queue;
@ -606,7 +606,7 @@ void ApmTest::ProcessDelayVerificationTest(int delay_ms,
SetFrameTo(&tmp_frame, 0);
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
// Initialize the |frame_queue| with empty frames.
// Initialize the `frame_queue` with empty frames.
int frame_delay = delay_ms / 10;
while (frame_delay < 0) {
Int16FrameData* frame = new Int16FrameData();
@ -1884,7 +1884,7 @@ TEST_F(ApmTest, Process) {
if (!absl::GetFlag(FLAGS_write_apm_ref_data)) {
const int kIntNear = 1;
// When running the test on a N7 we get a {2, 6} difference of
// |has_voice_count| and |max_output_average| is up to 18 higher.
// `has_voice_count` and `max_output_average` is up to 18 higher.
// All numbers being consistently higher on N7 compare to ref_data.
// TODO(bjornv): If we start getting more of these offsets on Android we
// should consider a different approach. Either using one slack for all,
@ -2058,7 +2058,7 @@ class AudioProcessingTest
static void TearDownTestSuite() { ClearTempFiles(); }
// Runs a process pass on files with the given parameters and dumps the output
// to a file specified with |output_file_prefix|. Both forward and reverse
// to a file specified with `output_file_prefix`. Both forward and reverse
// output streams are dumped.
static void ProcessFormat(int input_rate,
int output_rate,
@ -2277,7 +2277,7 @@ TEST_P(AudioProcessingTest, Formats) {
out_ptr = cmp_data.get();
}
// Update the |sq_error| and |variance| accumulators with the highest
// Update the `sq_error` and `variance` accumulators with the highest
// SNR of reference vs output.
UpdateBestSNR(ref_data.get(), out_ptr, ref_length, expected_delay,
&variance, &sq_error);