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Reformat the WebRTC code base

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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
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70
rtc_base/strings/audio_format_to_string.cc
View File

@ -13,40 +13,40 @@
#include "rtc_base/strings/string_builder.h"
namespace rtc {
std::string ToString(const webrtc::SdpAudioFormat& saf) {
char sb_buf[1024];
rtc::SimpleStringBuilder sb(sb_buf);
sb << "{name: " << saf.name;
sb << ", clockrate_hz: " << saf.clockrate_hz;
sb << ", num_channels: " << saf.num_channels;
sb << ", parameters: {";
const char* sep = "";
for (const auto& kv : saf.parameters) {
sb << sep << kv.first << ": " << kv.second;
sep = ", ";
}
sb << "}}";
return sb.str();
}
std::string ToString(const webrtc::AudioCodecInfo& aci) {
char sb_buf[1024];
rtc::SimpleStringBuilder sb(sb_buf);
sb << "{sample_rate_hz: " << aci.sample_rate_hz;
sb << ", num_channels: " << aci.num_channels;
sb << ", default_bitrate_bps: " << aci.default_bitrate_bps;
sb << ", min_bitrate_bps: " << aci.min_bitrate_bps;
sb << ", max_bitrate_bps: " << aci.max_bitrate_bps;
sb << ", allow_comfort_noise: " << aci.allow_comfort_noise;
sb << ", supports_network_adaption: " << aci.supports_network_adaption;
sb << "}";
return sb.str();
}
std::string ToString(const webrtc::AudioCodecSpec& acs) {
char sb_buf[1024];
rtc::SimpleStringBuilder sb(sb_buf);
sb << "{format: " << ToString(acs.format);
sb << ", info: " << ToString(acs.info);
sb << "}";
return sb.str();
std::string ToString(const webrtc::SdpAudioFormat& saf) {
char sb_buf[1024];
rtc::SimpleStringBuilder sb(sb_buf);
sb << "{name: " << saf.name;
sb << ", clockrate_hz: " << saf.clockrate_hz;
sb << ", num_channels: " << saf.num_channels;
sb << ", parameters: {";
const char* sep = "";
for (const auto& kv : saf.parameters) {
sb << sep << kv.first << ": " << kv.second;
sep = ", ";
}
sb << "}}";
return sb.str();
}
std::string ToString(const webrtc::AudioCodecInfo& aci) {
char sb_buf[1024];
rtc::SimpleStringBuilder sb(sb_buf);
sb << "{sample_rate_hz: " << aci.sample_rate_hz;
sb << ", num_channels: " << aci.num_channels;
sb << ", default_bitrate_bps: " << aci.default_bitrate_bps;
sb << ", min_bitrate_bps: " << aci.min_bitrate_bps;
sb << ", max_bitrate_bps: " << aci.max_bitrate_bps;
sb << ", allow_comfort_noise: " << aci.allow_comfort_noise;
sb << ", supports_network_adaption: " << aci.supports_network_adaption;
sb << "}";
return sb.str();
}
std::string ToString(const webrtc::AudioCodecSpec& acs) {
char sb_buf[1024];
rtc::SimpleStringBuilder sb(sb_buf);
sb << "{format: " << ToString(acs.format);
sb << ", info: " << ToString(acs.info);
sb << "}";
return sb.str();
}
} // namespace rtc