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}
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

View File

@ -45,17 +45,17 @@ enum H264EncoderImplEvent {
int NumberOfThreads(int width, int height, int number_of_cores) {
// TODO(hbos): In Chromium, multiple threads do not work with sandbox on Mac,
// see crbug.com/583348. Until further investigated, only use one thread.
// if (width * height >= 1920 * 1080 && number_of_cores > 8) {
// return 8; // 8 threads for 1080p on high perf machines.
// } else if (width * height > 1280 * 960 && number_of_cores >= 6) {
// return 3; // 3 threads for 1080p.
// } else if (width * height > 640 * 480 && number_of_cores >= 3) {
// return 2; // 2 threads for qHD/HD.
// } else {
// return 1; // 1 thread for VGA or less.
// }
// TODO(sprang): Also check sSliceArgument.uiSliceNum om GetEncoderPrams(),
// before enabling multithreading here.
// if (width * height >= 1920 * 1080 && number_of_cores > 8) {
// return 8; // 8 threads for 1080p on high perf machines.
// } else if (width * height > 1280 * 960 && number_of_cores >= 6) {
// return 3; // 3 threads for 1080p.
// } else if (width * height > 640 * 480 && number_of_cores >= 3) {
// return 2; // 2 threads for qHD/HD.
// } else {
// return 1; // 1 thread for VGA or less.
// }
// TODO(sprang): Also check sSliceArgument.uiSliceNum om GetEncoderPrams(),
// before enabling multithreading here.
return 1;
}
@ -139,10 +139,10 @@ static void RtpFragmentize(EncodedImage* encoded_image,
// Because the sum of all layer lengths, |required_size|, fits in a
// |size_t|, we know that any indices in-between will not overflow.
RTC_DCHECK_GE(layerInfo.pNalLengthInByte[nal], 4);
RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+0], start_code[0]);
RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+1], start_code[1]);
RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+2], start_code[2]);
RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+3], start_code[3]);
RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len + 0], start_code[0]);
RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len + 1], start_code[1]);
RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len + 2], start_code[2]);
RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len + 3], start_code[3]);
frag_header->fragmentationOffset[frag] =
encoded_image->_length + layer_len + sizeof(start_code);
frag_header->fragmentationLength[frag] =
@ -150,8 +150,7 @@ static void RtpFragmentize(EncodedImage* encoded_image,
layer_len += layerInfo.pNalLengthInByte[nal];
}
// Copy the entire layer's data (including start codes).
memcpy(encoded_image->_buffer + encoded_image->_length,
layerInfo.pBsBuf,
memcpy(encoded_image->_buffer + encoded_image->_length, layerInfo.pBsBuf,
layer_len);
encoded_image->_length += layer_len;
}
@ -190,8 +189,7 @@ int32_t H264EncoderImpl::InitEncode(const VideoCodec* codec_settings,
int32_t number_of_cores,
size_t max_payload_size) {
ReportInit();
if (!codec_settings ||
codec_settings->codecType != kVideoCodecH264) {
if (!codec_settings || codec_settings->codecType != kVideoCodecH264) {
ReportError();
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
@ -222,8 +220,7 @@ int32_t H264EncoderImpl::InitEncode(const VideoCodec* codec_settings,
RTC_DCHECK(openh264_encoder_);
if (kOpenH264EncoderDetailedLogging) {
int trace_level = WELS_LOG_DETAIL;
openh264_encoder_->SetOption(ENCODER_OPTION_TRACE_LEVEL,
&trace_level);
openh264_encoder_->SetOption(ENCODER_OPTION_TRACE_LEVEL, &trace_level);
}
// else WELS_LOG_DEFAULT is used by default.
@ -255,8 +252,7 @@ int32_t H264EncoderImpl::InitEncode(const VideoCodec* codec_settings,
}
// TODO(pbos): Base init params on these values before submitting.
int video_format = EVideoFormatType::videoFormatI420;
openh264_encoder_->SetOption(ENCODER_OPTION_DATAFORMAT,
&video_format);
openh264_encoder_->SetOption(ENCODER_OPTION_DATAFORMAT, &video_format);
// Initialize encoded image. Default buffer size: size of unencoded data.
encoded_image_._size = CalcBufferSize(VideoType::kI420, codec_settings->width,
@ -300,8 +296,7 @@ int32_t H264EncoderImpl::SetRateAllocation(
memset(&target_bitrate, 0, sizeof(SBitrateInfo));
target_bitrate.iLayer = SPATIAL_LAYER_ALL,
target_bitrate.iBitrate = target_bps_;
openh264_encoder_->SetOption(ENCODER_OPTION_BITRATE,
&target_bitrate);
openh264_encoder_->SetOption(ENCODER_OPTION_BITRATE, &target_bitrate);
openh264_encoder_->SetOption(ENCODER_OPTION_FRAME_RATE, &max_frame_rate_);
return WEBRTC_VIDEO_CODEC_OK;
}
@ -485,8 +480,7 @@ void H264EncoderImpl::ReportInit() {
if (has_reported_init_)
return;
RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.H264EncoderImpl.Event",
kH264EncoderEventInit,
kH264EncoderEventMax);
kH264EncoderEventInit, kH264EncoderEventMax);
has_reported_init_ = true;
}
@ -494,13 +488,12 @@ void H264EncoderImpl::ReportError() {
if (has_reported_error_)
return;
RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.H264EncoderImpl.Event",
kH264EncoderEventError,
kH264EncoderEventMax);
kH264EncoderEventError, kH264EncoderEventMax);
has_reported_error_ = true;
}
int32_t H264EncoderImpl::SetChannelParameters(
uint32_t packet_loss, int64_t rtt) {
int32_t H264EncoderImpl::SetChannelParameters(uint32_t packet_loss,
int64_t rtt) {
return WEBRTC_VIDEO_CODEC_OK;
}