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Reformat existing code. There should be no functional effects.

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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}
This commit is contained in:
Peter Kasting
2015-06-11 14:31:38 -07:00
parent b7e5054414
commit 728d9037c0
90 changed files with 416 additions and 490 deletions
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4
webrtc/modules/audio_coding/main/test/opus_test.cc
View File

@ -277,12 +277,12 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
bitstream_len_byte = WebRtcOpus_Encode(
opus_mono_encoder_, &audio[read_samples],
frame_length, kMaxBytes, bitstream);
ASSERT_GT(bitstream_len_byte, -1);
ASSERT_GE(bitstream_len_byte, 0);
} else {
bitstream_len_byte = WebRtcOpus_Encode(
opus_stereo_encoder_, &audio[read_samples],
frame_length, kMaxBytes, bitstream);
ASSERT_GT(bitstream_len_byte, -1);
ASSERT_GE(bitstream_len_byte, 0);
}
// Simulate packet loss by setting |packet_loss_| to "true" in