Format almost everything.

This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
This commit is contained in:
Jonas Olsson
2019-07-05 19:08:33 +02:00
committed by Commit Bot
parent c93bfcfd2f
commit a4d873786f
1202 changed files with 2991 additions and 1995 deletions

View File

@ -12,6 +12,7 @@
#include <stdio.h>
#include <stdlib.h>
#include <memory>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
@ -23,14 +24,10 @@
namespace webrtc {
TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency)
: _rtpStream(rtpStream),
_frequency(frequency),
_seqNo(0) {
}
TestPacketization::TestPacketization(RTPStream* rtpStream, uint16_t frequency)
: _rtpStream(rtpStream), _frequency(frequency), _seqNo(0) {}
TestPacketization::~TestPacketization() {
}
TestPacketization::~TestPacketization() {}
int32_t TestPacketization::SendData(const AudioFrameType /* frameType */,
const uint8_t payloadType,
@ -43,15 +40,14 @@ int32_t TestPacketization::SendData(const AudioFrameType /* frameType */,
}
Sender::Sender()
: _acm(NULL),
_pcmFile(),
_audioFrame(),
_packetization(NULL) {
}
: _acm(NULL), _pcmFile(), _audioFrame(), _packetization(NULL) {}
void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
std::string in_file_name, int in_sample_rate,
int payload_type, SdpAudioFormat format) {
void Sender::Setup(AudioCodingModule* acm,
RTPStream* rtpStream,
std::string in_file_name,
int in_sample_rate,
int payload_type,
SdpAudioFormat format) {
// Open input file
const std::string file_name = webrtc::test::ResourcePath(in_file_name, "pcm");
_pcmFile.Open(file_name, in_sample_rate, "rb");
@ -96,11 +92,13 @@ void Sender::Run() {
Receiver::Receiver()
: _playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
_payloadSizeBytes(MAX_INCOMING_PAYLOAD) {
}
_payloadSizeBytes(MAX_INCOMING_PAYLOAD) {}
void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
std::string out_file_name, size_t channels, int file_num) {
void Receiver::Setup(AudioCodingModule* acm,
RTPStream* rtpStream,
std::string out_file_name,
size_t channels,
int file_num) {
EXPECT_EQ(0, acm->InitializeReceiver());
if (channels == 1) {
@ -187,14 +185,14 @@ bool Receiver::PlayoutData() {
return false;
}
EXPECT_EQ(0, ok);
if (ok < 0){
if (ok < 0) {
return false;
}
if (_playoutLengthSmpls == 0) {
return false;
}
_pcmFile.Write10MsData(audioFrame.data(),
audioFrame.samples_per_channel_ * audioFrame.num_channels_);
_pcmFile.Write10MsData(audioFrame.data(), audioFrame.samples_per_channel_ *
audioFrame.num_channels_);
return true;
}
@ -225,17 +223,15 @@ void Receiver::Run() {
EncodeDecodeTest::EncodeDecodeTest() = default;
void EncodeDecodeTest::Perform() {
const std::map<int, SdpAudioFormat> send_codecs = {{103, {"ISAC", 16000, 1}},
{104, {"ISAC", 32000, 1}},
{107, {"L16", 8000, 1}},
{108, {"L16", 16000, 1}},
{109, {"L16", 32000, 1}},
{0, {"PCMU", 8000, 1}},
{8, {"PCMA", 8000, 1}},
const std::map<int, SdpAudioFormat> send_codecs = {
{103, {"ISAC", 16000, 1}}, {104, {"ISAC", 32000, 1}},
{107, {"L16", 8000, 1}}, {108, {"L16", 16000, 1}},
{109, {"L16", 32000, 1}}, {0, {"PCMU", 8000, 1}},
{8, {"PCMA", 8000, 1}},
#ifdef WEBRTC_CODEC_ILBC
{102, {"ILBC", 8000, 1}},
{102, {"ILBC", 8000, 1}},
#endif
{9, {"G722", 8000, 1}}};
{9, {"G722", 8000, 1}}};
int file_num = 0;
for (const auto& send_codec : send_codecs) {
RTPFile rtpFile;