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/neteq/sync_buffer.cc b/modules/audio_coding/neteq/sync_buffer.cc
index 28d7649..82ca16f 100644
--- a/modules/audio_coding/neteq/sync_buffer.cc
+++ b/modules/audio_coding/neteq/sync_buffer.cc
@@ -27,7 +27,7 @@
next_index_ -= samples_added;
} else {
// This means that we are pushing out future data that was never used.
-// assert(false);
+ // assert(false);
// TODO(hlundin): This assert must be disabled to support 60 ms frames.
// This should not happen even for 60 ms frames, but it does. Investigate
// why.
@@ -75,9 +75,8 @@
RTC_DCHECK(output);
const size_t samples_to_read = std::min(FutureLength(), requested_len);
output->ResetWithoutMuting();
- const size_t tot_samples_read =
- ReadInterleavedFromIndex(next_index_, samples_to_read,
- output->mutable_data());
+ const size_t tot_samples_read = ReadInterleavedFromIndex(
+ next_index_, samples_to_read, output->mutable_data());
const size_t samples_read_per_channel = tot_samples_read / Channels();
next_index_ += samples_read_per_channel;
output->num_channels_ = Channels();