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}diff --git a/common_audio/wav_header.cc b/common_audio/wav_header.cc
index a57e917..d093fa0 100644
--- a/common_audio/wav_header.cc
+++ b/common_audio/wav_header.cc
@@ -112,17 +112,23 @@
}
#ifdef WEBRTC_ARCH_LITTLE_ENDIAN
-static inline void WriteLE16(uint16_t* f, uint16_t x) { *f = x; }
-static inline void WriteLE32(uint32_t* f, uint32_t x) { *f = x; }
+static inline void WriteLE16(uint16_t* f, uint16_t x) {
+ *f = x;
+}
+static inline void WriteLE32(uint32_t* f, uint32_t x) {
+ *f = x;
+}
static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) {
- *f = static_cast<uint32_t>(a)
- | static_cast<uint32_t>(b) << 8
- | static_cast<uint32_t>(c) << 16
- | static_cast<uint32_t>(d) << 24;
+ *f = static_cast<uint32_t>(a) | static_cast<uint32_t>(b) << 8 |
+ static_cast<uint32_t>(c) << 16 | static_cast<uint32_t>(d) << 24;
}
-static inline uint16_t ReadLE16(uint16_t x) { return x; }
-static inline uint32_t ReadLE32(uint32_t x) { return x; }
+static inline uint16_t ReadLE16(uint16_t x) {
+ return x;
+}
+static inline uint32_t ReadLE32(uint32_t x) {
+ return x;
+}
static inline std::string ReadFourCC(uint32_t x) {
return std::string(reinterpret_cast<char*>(&x), 4);
}
@@ -131,11 +137,12 @@
#endif
static inline uint32_t RiffChunkSize(size_t bytes_in_payload) {
- return static_cast<uint32_t>(
- bytes_in_payload + kWavHeaderSize - sizeof(ChunkHeader));
+ return static_cast<uint32_t>(bytes_in_payload + kWavHeaderSize -
+ sizeof(ChunkHeader));
}
-static inline uint32_t ByteRate(size_t num_channels, int sample_rate,
+static inline uint32_t ByteRate(size_t num_channels,
+ int sample_rate,
size_t bytes_per_sample) {
return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample);
}
@@ -166,8 +173,8 @@
WriteLE16(&header.fmt.AudioFormat, format);
WriteLE16(&header.fmt.NumChannels, static_cast<uint16_t>(num_channels));
WriteLE32(&header.fmt.SampleRate, sample_rate);
- WriteLE32(&header.fmt.ByteRate, ByteRate(num_channels, sample_rate,
- bytes_per_sample));
+ WriteLE32(&header.fmt.ByteRate,
+ ByteRate(num_channels, sample_rate, bytes_per_sample));
WriteLE16(&header.fmt.BlockAlign, BlockAlign(num_channels, bytes_per_sample));
WriteLE16(&header.fmt.BitsPerSample,
static_cast<uint16_t>(8 * bytes_per_sample));
@@ -239,5 +246,4 @@
*bytes_per_sample, *num_samples);
}
-
} // namespace webrtc