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

@ -26,7 +26,7 @@ namespace webrtc {
// TestPacketization callback which writes the encoded payloads to file
class TestPacketization : public AudioPacketizationCallback {
public:
TestPacketization(RTPStream *rtpStream, uint16_t frequency);
TestPacketization(RTPStream* rtpStream, uint16_t frequency);
~TestPacketization();
int32_t SendData(const AudioFrameType frameType,
const uint8_t payloadType,
@ -35,8 +35,11 @@ class TestPacketization : public AudioPacketizationCallback {
const size_t payloadSize) override;
private:
static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
static void MakeRTPheader(uint8_t* rtpHeader,
uint8_t payloadType,
int16_t seqNo,
uint32_t timeStamp,
uint32_t ssrc);
RTPStream* _rtpStream;
int32_t _frequency;
int16_t _seqNo;
@ -45,9 +48,12 @@ class TestPacketization : public AudioPacketizationCallback {
class Sender {
public:
Sender();
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
std::string in_file_name, int in_sample_rate,
int payload_type, SdpAudioFormat format);
void Setup(AudioCodingModule* acm,
RTPStream* rtpStream,
std::string in_file_name,
int in_sample_rate,
int payload_type,
SdpAudioFormat format);
void Teardown();
void Run();
bool Add10MsData();
@ -65,8 +71,11 @@ class Receiver {
public:
Receiver();
virtual ~Receiver() {}
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
std::string out_file_name, size_t channels, int file_num);
void Setup(AudioCodingModule* acm,
RTPStream* rtpStream,
std::string out_file_name,
size_t channels,
int file_num);
void Teardown();
void Run();
virtual bool IncomingPacket();