Reformat the WebRTC code base

Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This commit is contained in:
Yves Gerey
2018-06-19 15:03:05 +02:00
parent b602123a5a
commit 665174fdbb
1569 changed files with 30495 additions and 30309 deletions

View File

@ -26,15 +26,13 @@ class MockDecoderDatabase : public DecoderDatabase {
: DecoderDatabase(factory, absl::nullopt) {}
virtual ~MockDecoderDatabase() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_CONST_METHOD0(Empty,
bool());
MOCK_CONST_METHOD0(Size,
int());
MOCK_METHOD0(Reset,
void());
MOCK_CONST_METHOD0(Empty, bool());
MOCK_CONST_METHOD0(Size, int());
MOCK_METHOD0(Reset, void());
MOCK_METHOD3(RegisterPayload,
int(uint8_t rtp_payload_type, NetEqDecoder codec_type,
const std::string& name));
int(uint8_t rtp_payload_type,
NetEqDecoder codec_type,
const std::string& name));
MOCK_METHOD2(RegisterPayload,
int(int rtp_payload_type, const SdpAudioFormat& audio_format));
MOCK_METHOD4(InsertExternal,
@ -42,19 +40,15 @@ class MockDecoderDatabase : public DecoderDatabase {
NetEqDecoder codec_type,
const std::string& codec_name,
AudioDecoder* decoder));
MOCK_METHOD1(Remove,
int(uint8_t rtp_payload_type));
MOCK_METHOD1(Remove, int(uint8_t rtp_payload_type));
MOCK_METHOD0(RemoveAll, void());
MOCK_CONST_METHOD1(GetDecoderInfo,
const DecoderInfo*(uint8_t rtp_payload_type));
const DecoderInfo*(uint8_t rtp_payload_type));
MOCK_METHOD2(SetActiveDecoder,
int(uint8_t rtp_payload_type, bool* new_decoder));
MOCK_CONST_METHOD0(GetActiveDecoder,
AudioDecoder*());
MOCK_METHOD1(SetActiveCngDecoder,
int(uint8_t rtp_payload_type));
MOCK_CONST_METHOD0(GetActiveCngDecoder,
ComfortNoiseDecoder*());
int(uint8_t rtp_payload_type, bool* new_decoder));
MOCK_CONST_METHOD0(GetActiveDecoder, AudioDecoder*());
MOCK_METHOD1(SetActiveCngDecoder, int(uint8_t rtp_payload_type));
MOCK_CONST_METHOD0(GetActiveCngDecoder, ComfortNoiseDecoder*());
};
} // namespace webrtc