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Lint fix for webrtc/modules/video_coding PART 3!

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Trying to submit all changes at once proved impossible since there were
too many changes in too many files. The changes to PRESUBMIT.py
will be uploaded in the last CL.
(original CL: https://codereview.webrtc.org/1528503003/)

BUG=webrtc:5309
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1540243002

Cr-Commit-Position: refs/heads/master@{#11105}
This commit is contained in:
philipel
2015-12-21 08:23:20 -08:00
committed by Commit bot
parent f5b1abf5b0
commit 5908c71128
29 changed files with 662 additions and 767 deletions
Download Patch File Download Diff File Expand all files Collapse all files

49
webrtc/modules/video_coding/test/rtp_player.cc
View File

@ -26,9 +26,9 @@
#include "webrtc/test/rtp_file_reader.h"
#if 1
# define DEBUG_LOG1(text, arg)
#define DEBUG_LOG1(text, arg)
#else
# define DEBUG_LOG1(text, arg) (printf(text "\n", arg))
#define DEBUG_LOG1(text, arg) (printf(text "\n", arg))
#endif
namespace webrtc {
@ -41,7 +41,9 @@ enum {
class RawRtpPacket {
public:
RawRtpPacket(const uint8_t* data, size_t length, uint32_t ssrc,
RawRtpPacket(const uint8_t* data,
size_t length,
uint32_t ssrc,
uint16_t seq_num)
: data_(new uint8_t[length]),
length_(length),
@ -140,7 +142,7 @@ class LostPackets {
CriticalSectionScoped cs(crit_sect_.get());
int count = 0;
for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
++it) {
++it) {
if ((*it)->resend_time_ms() >= 0) {
count++;
}
@ -164,7 +166,7 @@ class LostPackets {
printf("Packets still lost: %zd\n", packets_.size());
printf("Sequence numbers:\n");
for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
++it) {
++it) {
printf("%u, ", (*it)->seq_num());
}
printf("\n");
@ -231,17 +233,14 @@ class SsrcHandlers {
kDefaultTransmissionTimeOffsetExtensionId);
for (PayloadTypesIterator it = payload_types_.begin();
it != payload_types_.end(); ++it) {
it != payload_types_.end(); ++it) {
VideoCodec codec;
memset(&codec, 0, sizeof(codec));
strncpy(codec.plName, it->name().c_str(), sizeof(codec.plName)-1);
strncpy(codec.plName, it->name().c_str(), sizeof(codec.plName) - 1);
codec.plType = it->payload_type();
codec.codecType = it->codec_type();
if (handler->rtp_module_->RegisterReceivePayload(codec.plName,
codec.plType,
90000,
0,
codec.maxBitrate) < 0) {
if (handler->rtp_module_->RegisterReceivePayload(
codec.plName, codec.plType, 90000, 0, codec.maxBitrate) < 0) {
return -1;
}
}
@ -267,7 +266,8 @@ class SsrcHandlers {
private:
class Handler : public RtpStreamInterface {
public:
Handler(uint32_t ssrc, const PayloadTypes& payload_types,
Handler(uint32_t ssrc,
const PayloadTypes& payload_types,
LostPackets* lost_packets)
: rtp_header_parser_(RtpHeaderParser::Create()),
rtp_payload_registry_(new RTPPayloadRegistry(
@ -290,9 +290,7 @@ class SsrcHandlers {
}
virtual uint32_t ssrc() const { return ssrc_; }
virtual const PayloadTypes& payload_types() const {
return payload_types_;
}
virtual const PayloadTypes& payload_types() const { return payload_types_; }
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
@ -351,8 +349,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
virtual int NextPacket(int64_t time_now) {
// Send any packets ready to be resent.
for (RawRtpPacket* packet = lost_packets_.NextPacketToResend(time_now);
packet != NULL;
packet = lost_packets_.NextPacketToResend(time_now)) {
packet != NULL; packet = lost_packets_.NextPacketToResend(time_now)) {
int ret = SendPacket(packet->data(), packet->length());
if (ret > 0) {
printf("Resend: %08x:%u\n", packet->ssrc(), packet->seq_num());
@ -392,8 +389,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
if (!packet_source_->NextPacket(&next_packet_)) {
end_of_file_ = true;
return 0;
}
else if (next_packet_.length == 0) {
} else if (next_packet_.length == 0) {
return 0;
}
}
@ -406,7 +402,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
virtual uint32_t TimeUntilNextPacket() const {
int64_t time_left = (next_rtp_time_ - first_packet_rtp_time_) -
(clock_->TimeInMilliseconds() - first_packet_time_ms_);
(clock_->TimeInMilliseconds() - first_packet_time_ms_);
if (time_left < 0) {
return 0;
}
@ -438,7 +434,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
if (no_loss_startup_ > 0) {
no_loss_startup_--;
} else if ((rand() + 1.0)/(RAND_MAX + 1.0) < loss_rate_) {
} else if ((rand() + 1.0) / (RAND_MAX + 1.0) < loss_rate_) { // NOLINT
uint16_t seq_num = header.sequenceNumber;
lost_packets_.AddPacket(new RawRtpPacket(data, length, ssrc, seq_num));
DEBUG_LOG1("Dropped packet: %d!", header.header.sequenceNumber);
@ -470,9 +466,12 @@ class RtpPlayerImpl : public RtpPlayerInterface {
};
RtpPlayerInterface* Create(const std::string& input_filename,
PayloadSinkFactoryInterface* payload_sink_factory, Clock* clock,
const PayloadTypes& payload_types, float loss_rate, int64_t rtt_ms,
bool reordering) {
PayloadSinkFactoryInterface* payload_sink_factory,
Clock* clock,
const PayloadTypes& payload_types,
float loss_rate,
int64_t rtt_ms,
bool reordering) {
rtc::scoped_ptr<test::RtpFileReader> packet_source(
test::RtpFileReader::Create(test::RtpFileReader::kRtpDump,
input_filename));