zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Reformat the WebRTC code base

Browse Source 
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
Download Patch File Download Diff File Expand all files Collapse all files

41
modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
View File

@ -89,7 +89,6 @@ AudioEncoder::EncodedInfo AudioEncoderIlbcImpl::EncodeImpl(
uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) {
// Save timestamp if starting a new packet.
if (num_10ms_frames_buffered_ == 0)
first_timestamp_in_buffer_ = rtp_timestamp;
@ -107,19 +106,15 @@ AudioEncoder::EncodedInfo AudioEncoderIlbcImpl::EncodeImpl(
// Encode buffered input.
RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
num_10ms_frames_buffered_ = 0;
size_t encoded_bytes =
encoded->AppendData(
RequiredOutputSizeBytes(),
[&] (rtc::ArrayView<uint8_t> encoded) {
const int r = WebRtcIlbcfix_Encode(
encoder_,
input_buffer_,
kSampleRateHz / 100 * num_10ms_frames_per_packet_,
encoded.data());
RTC_CHECK_GE(r, 0);
size_t encoded_bytes = encoded->AppendData(
RequiredOutputSizeBytes(), [&](rtc::ArrayView<uint8_t> encoded) {
const int r = WebRtcIlbcfix_Encode(
encoder_, input_buffer_,
kSampleRateHz / 100 * num_10ms_frames_per_packet_, encoded.data());
RTC_CHECK_GE(r, 0);
return static_cast<size_t>(r);
});
return static_cast<size_t>(r);
});
RTC_DCHECK_EQ(encoded_bytes, RequiredOutputSizeBytes());
@ -135,20 +130,24 @@ void AudioEncoderIlbcImpl::Reset() {
if (encoder_)
RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
const int encoder_frame_size_ms = frame_size_ms_ > 30
? frame_size_ms_ / 2
: frame_size_ms_;
const int encoder_frame_size_ms =
frame_size_ms_ > 30 ? frame_size_ms_ / 2 : frame_size_ms_;
RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
num_10ms_frames_buffered_ = 0;
}
size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
switch (num_10ms_frames_per_packet_) {
case 2: return 38;
case 3: return 50;
case 4: return 2 * 38;
case 6: return 2 * 50;
default: FATAL();
case 2:
return 38;
case 3:
return 50;
case 4:
return 2 * 38;
case 6:
return 2 * 50;
default:
FATAL();
}
}