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| author | Peter Kasting <pkasting@google.com> | 2015-06-11 14:31:38 -0700 |
|---|---|---|
| committer | Peter Kasting <pkasting@google.com> | 2015-06-11 21:31:48 +0000 |
| commit | 728d9037c016c01295177fa700fc7927f0bb80bb (patch) | |
| tree | 808f3f16c1ef276ebc6ab8e1bacfb29e2de72a96 /webrtc/modules/audio_processing/audio_buffer.cc | |
| parent | b7e5054414ff524f9db81dab7917729b8c4c8bcb (diff) | |
| download | webrtc-728d9037c016c01295177fa700fc7927f0bb80bb.tar.gz | |
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}
Diffstat (limited to 'webrtc/modules/audio_processing/audio_buffer.cc')
| -rw-r--r-- | webrtc/modules/audio_processing/audio_buffer.cc | 13 |
1 files changed, 5 insertions, 8 deletions
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc index ec5e2279ce..04dcaea799 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 @@ AudioBuffer::AudioBuffer(int input_num_frames, 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 @@ int AudioBuffer::num_bands() const { // 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_) { |