[rtp_rtcp] lint errors about rand() usage fixed.
rand() usage replaced with new Random class, which also makes it clearer what interval random number is in.
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1519503002
Cr-Commit-Position: refs/heads/master@{#11019}
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index ddf24ab..ea62fcb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -11,7 +11,6 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include <assert.h> // assert
-#include <stdlib.h> // rand
#include <string.h> // memcpy
#include <algorithm> // min
@@ -141,6 +140,7 @@
Transport* outgoing_transport)
: audio_(audio),
clock_(clock),
+ random_(clock_->TimeInMicroseconds()),
method_(RtcpMode::kOff),
transport_(outgoing_transport),
@@ -914,15 +914,9 @@
SetFlag(kRtcpXrDlrrReportBlock, true);
// generate next time to send an RTCP report
- // seeded from RTP constructor
- int32_t random = rand() % 1000;
- int32_t timeToNext = RTCP_INTERVAL_AUDIO_MS;
+ uint32_t minIntervalMs = RTCP_INTERVAL_AUDIO_MS;
- if (audio_) {
- timeToNext = (RTCP_INTERVAL_AUDIO_MS / 2) +
- (RTCP_INTERVAL_AUDIO_MS * random / 1000);
- } else {
- uint32_t minIntervalMs = RTCP_INTERVAL_AUDIO_MS;
+ if (!audio_) {
if (sending_) {
// Calculate bandwidth for video; 360 / send bandwidth in kbit/s.
uint32_t send_bitrate_kbit = feedback_state.send_bitrate / 1000;
@@ -931,8 +925,11 @@
}
if (minIntervalMs > RTCP_INTERVAL_VIDEO_MS)
minIntervalMs = RTCP_INTERVAL_VIDEO_MS;
- timeToNext = (minIntervalMs / 2) + (minIntervalMs * random / 1000);
}
+ // The interval between RTCP packets is varied randomly over the
+ // range [1/2,3/2] times the calculated interval.
+ uint32_t timeToNext =
+ random_.Rand(minIntervalMs * 1 / 2, minIntervalMs * 3 / 2);
next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + timeToNext;
StatisticianMap statisticians =
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index fcb9012..11a389e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -17,6 +17,7 @@
#include <string>
#include <vector>
+#include "webrtc/base/random.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
@@ -202,6 +203,7 @@
private:
const bool audio_;
Clock* const clock_;
+ Random random_ GUARDED_BY(critical_section_rtcp_sender_);
RtcpMode method_ GUARDED_BY(critical_section_rtcp_sender_);
Transport* const transport_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
index 541f522..80f961b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
@@ -11,6 +11,7 @@
#include <list>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/random.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
@@ -41,8 +42,12 @@
class RtpFecTest : public ::testing::Test {
protected:
RtpFecTest()
- : fec_(new ForwardErrorCorrection()), ssrc_(rand()), fec_seq_num_(0) {}
+ : random_(0xfec133700742),
+ fec_(new ForwardErrorCorrection()),
+ ssrc_(random_.Rand<uint32_t>()),
+ fec_seq_num_(0) {}
+ webrtc::Random random_;
ForwardErrorCorrection* fec_;
int ssrc_;
uint16_t fec_seq_num_;
@@ -891,22 +896,20 @@
assert(num_media_packets > 0);
ForwardErrorCorrection::Packet* media_packet = NULL;
int sequence_number = start_seq_num;
- int time_stamp = rand();
+ int time_stamp = random_.Rand<int>();
for (int i = 0; i < num_media_packets; ++i) {
media_packet = new ForwardErrorCorrection::Packet;
media_packet_list_.push_back(media_packet);
- media_packet->length = static_cast<size_t>(
- (static_cast<float>(rand()) / RAND_MAX) *
- (IP_PACKET_SIZE - kRtpHeaderSize - kTransportOverhead -
- ForwardErrorCorrection::PacketOverhead()));
+ const uint32_t kMinPacketSize = kRtpHeaderSize;
+ const uint32_t kMaxPacketSize = IP_PACKET_SIZE - kRtpHeaderSize -
+ kTransportOverhead -
+ ForwardErrorCorrection::PacketOverhead();
+ media_packet->length = random_.Rand(kMinPacketSize, kMaxPacketSize);
- if (media_packet->length < kRtpHeaderSize) {
- media_packet->length = kRtpHeaderSize;
- }
// Generate random values for the first 2 bytes
- media_packet->data[0] = static_cast<uint8_t>(rand() % 256);
- media_packet->data[1] = static_cast<uint8_t>(rand() % 256);
+ media_packet->data[0] = random_.Rand<uint8_t>();
+ media_packet->data[1] = random_.Rand<uint8_t>();
// The first two bits are assumed to be 10 by the FEC encoder.
// In fact the FEC decoder will set the two first bits to 10 regardless of
@@ -929,7 +932,7 @@
// Generate random values for payload.
for (size_t j = 12; j < media_packet->length; ++j) {
- media_packet->data[j] = static_cast<uint8_t>(rand() % 256);
+ media_packet->data[j] = random_.Rand<uint8_t>();
}
sequence_number++;
}
@@ -940,5 +943,5 @@
}
int RtpFecTest::ConstructMediaPackets(int num_media_packets) {
- return ConstructMediaPacketsSeqNum(num_media_packets, rand());
+ return ConstructMediaPacketsSeqNum(num_media_packets, random_.Rand<int>());
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 152293a..4e91a29 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -34,6 +34,7 @@
namespace {
const size_t kRtpHeaderLength = 12;
+const uint16_t kMaxInitRtpSeqNumber = 32767; // 2^15 -1.
const char* FrameTypeToString(FrameType frame_type) {
switch (frame_type) {
@@ -126,6 +127,7 @@
// TickTime.
clock_delta_ms_(clock_->TimeInMilliseconds() -
TickTime::MillisecondTimestamp()),
+ random_(clock_->TimeInMicroseconds()),
bitrates_(new BitrateAggregator(bitrate_callback)),
total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()),
audio_configured_(audio),
@@ -183,8 +185,8 @@
ssrc_rtx_ = ssrc_db_.CreateSSRC(); // Can't be 0.
bitrates_->set_ssrc(ssrc_);
// Random start, 16 bits. Can't be 0.
- sequence_number_rtx_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
- sequence_number_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
+ sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
+ sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
RTPSender::~RTPSender() {
@@ -1656,8 +1658,7 @@
// Don't initialize seq number if SSRC passed externally.
if (!sequence_number_forced_ && !ssrc_forced_) {
// Generate a new sequence number.
- sequence_number_ =
- rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); // NOLINT
+ sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
}
}
@@ -1719,8 +1720,7 @@
ssrc_ = ssrc;
bitrates_->set_ssrc(ssrc_);
if (!sequence_number_forced_) {
- sequence_number_ =
- rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); // NOLINT
+ sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 30ac204..2aa7f4c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -16,6 +16,7 @@
#include <utility>
#include <vector>
+#include "webrtc/base/random.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -27,8 +28,6 @@
#include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
#include "webrtc/transport.h"
-#define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1.
-
namespace webrtc {
class BitrateAggregator;
@@ -387,6 +386,7 @@
Clock* clock_;
int64_t clock_delta_ms_;
+ Random random_ GUARDED_BY(send_critsect_);
rtc::scoped_ptr<BitrateAggregator> bitrates_;
Bitrate total_bitrate_sent_;
diff --git a/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc b/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
index 7f52a87..b164b7e 100644
--- a/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
+++ b/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
@@ -22,10 +22,10 @@
#include <list>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/random.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
-
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/test/testsupport/fileutils.h"
// #define VERBOSE_OUTPUT
@@ -40,30 +40,31 @@
void ReceivePackets(
ForwardErrorCorrection::ReceivedPacketList* toDecodeList,
ForwardErrorCorrection::ReceivedPacketList* receivedPacketList,
- uint32_t numPacketsToDecode,
+ size_t numPacketsToDecode,
float reorderRate,
- float duplicateRate) {
+ float duplicateRate,
+ Random* random) {
assert(toDecodeList->empty());
assert(numPacketsToDecode <= receivedPacketList->size());
ForwardErrorCorrection::ReceivedPacketList::iterator it;
- for (uint32_t i = 0; i < numPacketsToDecode; i++) {
+ for (size_t i = 0; i < numPacketsToDecode; i++) {
it = receivedPacketList->begin();
// Reorder packets.
- float randomVariable = static_cast<float>(rand()) / RAND_MAX;
+ float randomVariable = random->Rand<float>();
while (randomVariable < reorderRate) {
++it;
if (it == receivedPacketList->end()) {
--it;
break;
}
- randomVariable = static_cast<float>(rand()) / RAND_MAX;
+ randomVariable = random->Rand<float>();
}
ForwardErrorCorrection::ReceivedPacket* receivedPacket = *it;
toDecodeList->push_back(receivedPacket);
// Duplicate packets.
- randomVariable = static_cast<float>(rand()) / RAND_MAX;
+ randomVariable = random->Rand<float>();
while (randomVariable < duplicateRate) {
ForwardErrorCorrection::ReceivedPacket* duplicatePacket =
new ForwardErrorCorrection::ReceivedPacket;
@@ -74,7 +75,7 @@
duplicatePacket->pkt->length = receivedPacket->pkt->length;
toDecodeList->push_back(duplicatePacket);
- randomVariable = static_cast<float>(rand()) / RAND_MAX;
+ randomVariable = random->Rand<float>();
}
receivedPacketList->erase(it);
}
@@ -125,7 +126,7 @@
// Seed the random number generator, storing the seed to file in order to
// reproduce past results.
const unsigned int randomSeed = static_cast<unsigned int>(time(NULL));
- srand(randomSeed);
+ Random random(randomSeed);
std::string filename = webrtc::test::OutputPath() + "randomSeedLog.txt";
FILE* randomSeedFile = fopen(filename.c_str(), "a");
fprintf(randomSeedFile, "%u\n", randomSeed);
@@ -133,8 +134,8 @@
randomSeedFile = NULL;
uint16_t seqNum = 0;
- uint32_t timeStamp = static_cast<uint32_t>(rand());
- const uint32_t ssrc = static_cast<uint32_t>(rand());
+ uint32_t timeStamp = random.Rand<uint32_t>();
+ const uint32_t ssrc = random.Rand(1u, 0xfffffffe);
// Loop over the mask types: random and bursty.
for (int mask_type_idx = 0; mask_type_idx < kNumFecMaskTypes;
@@ -227,16 +228,15 @@
for (uint32_t i = 0; i < numMediaPackets; ++i) {
mediaPacket = new ForwardErrorCorrection::Packet;
mediaPacketList.push_back(mediaPacket);
- mediaPacket->length = static_cast<size_t>(
- (static_cast<float>(rand()) / RAND_MAX) *
- (IP_PACKET_SIZE - 12 - 28 -
- ForwardErrorCorrection::PacketOverhead()));
- if (mediaPacket->length < 12) {
- mediaPacket->length = 12;
- }
+ const uint32_t kMinPacketSize = 12;
+ const uint32_t kMaxPacketSize = static_cast<uint32_t>(
+ IP_PACKET_SIZE - 12 - 28 -
+ ForwardErrorCorrection::PacketOverhead());
+ mediaPacket->length = random.Rand(kMinPacketSize, kMaxPacketSize);
+
// Generate random values for the first 2 bytes.
- mediaPacket->data[0] = static_cast<uint8_t>(rand() % 256);
- mediaPacket->data[1] = static_cast<uint8_t>(rand() % 256);
+ mediaPacket->data[0] = random.Rand<uint8_t>();
+ mediaPacket->data[1] = random.Rand<uint8_t>();
// The first two bits are assumed to be 10 by the
// FEC encoder. In fact the FEC decoder will set the
@@ -261,7 +261,7 @@
ByteWriter<uint32_t>::WriteBigEndian(&mediaPacket->data[8], ssrc);
// Generate random values for payload
for (size_t j = 12; j < mediaPacket->length; ++j) {
- mediaPacket->data[j] = static_cast<uint8_t>(rand() % 256);
+ mediaPacket->data[j] = random.Rand<uint8_t>();
}
seqNum++;
}
@@ -284,8 +284,7 @@
while (mediaPacketListItem != mediaPacketList.end()) {
mediaPacket = *mediaPacketListItem;
// We want a value between 0 and 1.
- const float lossRandomVariable =
- (static_cast<float>(rand()) / (RAND_MAX));
+ const float lossRandomVariable = random.Rand<float>();
if (lossRandomVariable >= lossRate[lossRateIdx]) {
mediaLossMask[mediaPacketIdx] = 1;
@@ -310,8 +309,7 @@
uint32_t fecPacketIdx = 0;
while (fecPacketListItem != fecPacketList.end()) {
fecPacket = *fecPacketListItem;
- const float lossRandomVariable =
- (static_cast<float>(rand()) / (RAND_MAX));
+ const float lossRandomVariable = random.Rand<float>();
if (lossRandomVariable >= lossRate[lossRateIdx]) {
fecLossMask[fecPacketIdx] = 1;
receivedPacket = new ForwardErrorCorrection::ReceivedPacket;
@@ -382,15 +380,11 @@
// For error-checking frame completion.
bool fecPacketReceived = false;
while (!receivedPacketList.empty()) {
- uint32_t numPacketsToDecode = static_cast<uint32_t>(
- (static_cast<float>(rand()) / RAND_MAX) *
- receivedPacketList.size() +
- 0.5);
- if (numPacketsToDecode < 1) {
- numPacketsToDecode = 1;
- }
+ size_t numPacketsToDecode = random.Rand(
+ 1u, static_cast<uint32_t>(receivedPacketList.size()));
ReceivePackets(&toDecodeList, &receivedPacketList,
- numPacketsToDecode, reorderRate, duplicateRate);
+ numPacketsToDecode, reorderRate, duplicateRate,
+ &random);
if (fecPacketReceived == false) {
ForwardErrorCorrection::ReceivedPacketList::iterator
