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/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index b61099c..0a88e70 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -231,8 +231,8 @@
if (!audio_decoder) {
ret_val = neteq_->RegisterPayloadType(neteq_decoder, name, payload_type);
} else {
- ret_val = neteq_->RegisterExternalDecoder(
- audio_decoder, neteq_decoder, name, payload_type);
+ ret_val = neteq_->RegisterExternalDecoder(audio_decoder, neteq_decoder,
+ name, payload_type);
}
if (ret_val != NetEq::kOK) {
RTC_LOG(LERROR) << "AcmReceiver::AddCodec " << acm_codec_id
@@ -402,10 +402,9 @@
// the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
// We masked 6 most significant bits of 32-bit so there is no overflow in
// the conversion from milliseconds to timestamp.
- const uint32_t now_in_ms = static_cast<uint32_t>(
- clock_->TimeInMilliseconds() & 0x03ffffff);
- return static_cast<uint32_t>(
- (decoder_sampling_rate / 1000) * now_in_ms);
+ const uint32_t now_in_ms =
+ static_cast<uint32_t>(clock_->TimeInMilliseconds() & 0x03ffffff);
+ return static_cast<uint32_t>((decoder_sampling_rate / 1000) * now_in_ms);
}
void AcmReceiver::GetDecodingCallStatistics(
