Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index ec5e227..04dcaea 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/modules/audio_processing/audio_buffer.cc
@@ -19,11 +19,9 @@
namespace webrtc {
namespace {
-enum {
- kSamplesPer16kHzChannel = 160,
- kSamplesPer32kHzChannel = 320,
- kSamplesPer48kHzChannel = 480
-};
+const int kSamplesPer16kHzChannel = 160;
+const int kSamplesPer32kHzChannel = 320;
+const int kSamplesPer48kHzChannel = 480;
bool HasKeyboardChannel(AudioProcessing::ChannelLayout layout) {
switch (layout) {
@@ -84,8 +82,7 @@
output_num_frames_(output_num_frames),
num_channels_(num_process_channels),
num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
- num_split_frames_(rtc::CheckedDivExact(
- proc_num_frames_, num_bands_)),
+ num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
mixed_low_pass_valid_(false),
reference_copied_(false),
activity_(AudioFrame::kVadUnknown),
@@ -399,7 +396,7 @@
// The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
assert(frame->num_channels_ == num_input_channels_);
- assert(frame->samples_per_channel_ == input_num_frames_);
+ assert(frame->samples_per_channel_ == input_num_frames_);
InitForNewData();
// Initialized lazily because there's a different condition in CopyFrom.
if ((input_num_frames_ != proc_num_frames_) && !input_buffer_) {