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/modules/audio_coding/test/Channel.h b/modules/audio_coding/test/Channel.h
index e01e33e..e5f5b54 100644
--- a/modules/audio_coding/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -20,8 +20,8 @@
namespace webrtc {
-#define MAX_NUM_PAYLOADS 50
-#define MAX_NUM_FRAMESIZES 6
+#define MAX_NUM_PAYLOADS 50
+#define MAX_NUM_FRAMESIZES 6
// TODO(turajs): Write constructor for this structure.
struct ACMTestFrameSizeStats {
@@ -45,7 +45,6 @@
class Channel : public AudioPacketizationCallback {
public:
-
Channel(int16_t chID = -1);
~Channel() override;
@@ -56,7 +55,7 @@
size_t payloadSize,
const RTPFragmentationHeader* fragmentation) override;
- void RegisterReceiverACM(AudioCodingModule *acm);
+ void RegisterReceiverACM(AudioCodingModule* acm);
void ResetStats();
@@ -68,9 +67,7 @@
void PrintStats(CodecInst& codecInst);
- void SetIsStereo(bool isStereo) {
- _isStereo = isStereo;
- }
+ void SetIsStereo(bool isStereo) { _isStereo = isStereo; }
uint32_t LastInTimestamp();