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Use size_t more consistently for packet/payload lengths.

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See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pkasting@chromium.org
2014-11-20 22:28:14 +00:00
parent edc6e57a92
commit 4591fbd09f
341 changed files with 2610 additions and 2613 deletions
Download Patch File Download Diff File Expand all files Collapse all files

18
webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
View File

@ -43,7 +43,7 @@ class LoopBackTransport : public webrtc::Transport {
void DropEveryNthPacket(int n) {
_packetLoss = n;
}
virtual int SendPacket(int channel, const void *data, int len) OVERRIDE {
virtual int SendPacket(int channel, const void *data, size_t len) OVERRIDE {
_count++;
if (_packetLoss > 0) {
if ((_count % _packetLoss) == 0) {
@ -52,9 +52,7 @@ class LoopBackTransport : public webrtc::Transport {
}
RTPHeader header;
scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
if (!parser->Parse(static_cast<const uint8_t*>(data),
static_cast<size_t>(len),
&header)) {
if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) {
return -1;
}
PayloadUnion payload_specific;
@ -70,11 +68,13 @@ class LoopBackTransport : public webrtc::Transport {
}
return len;
}
virtual int SendRTCPPacket(int channel, const void *data, int len) OVERRIDE {
virtual int SendRTCPPacket(int channel,
const void *data,
size_t len) OVERRIDE {
if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data, len) < 0) {
return -1;
}
return len;
return static_cast<int>(len);
}
private:
int _count;
@ -90,7 +90,7 @@ class TestRtpReceiver : public NullRtpData {
virtual int32_t OnReceivedPayloadData(
const uint8_t* payloadData,
const uint16_t payloadSize,
const size_t payloadSize,
const webrtc::WebRtcRTPHeader* rtpHeader) OVERRIDE {
EXPECT_LE(payloadSize, sizeof(_payloadData));
memcpy(_payloadData, payloadData, payloadSize);
@ -103,7 +103,7 @@ class TestRtpReceiver : public NullRtpData {
return _payloadData;
}
uint16_t payload_size() const {
size_t payload_size() const {
return _payloadSize;
}
@ -113,7 +113,7 @@ class TestRtpReceiver : public NullRtpData {
private:
uint8_t _payloadData[1500];
uint16_t _payloadSize;
size_t _payloadSize;
webrtc::WebRtcRTPHeader _rtpHeader;
};

8
webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
View File

@ -27,11 +27,11 @@ class VerifyingAudioReceiver : public NullRtpData {
public:
virtual int32_t OnReceivedPayloadData(
const uint8_t* payloadData,
const uint16_t payloadSize,
const size_t payloadSize,
const webrtc::WebRtcRTPHeader* rtpHeader) OVERRIDE {
if (rtpHeader->header.payloadType == 98 ||
rtpHeader->header.payloadType == 99) {
EXPECT_EQ(4, payloadSize);
EXPECT_EQ(4u, payloadSize);
char str[5];
memcpy(str, payloadData, payloadSize);
str[4] = 0;
@ -265,10 +265,10 @@ TEST_F(RtpRtcpAudioTest, RED) {
RTPFragmentationHeader fragmentation;
fragmentation.fragmentationVectorSize = 2;
fragmentation.fragmentationLength = new uint32_t[2];
fragmentation.fragmentationLength = new size_t[2];
fragmentation.fragmentationLength[0] = 4;
fragmentation.fragmentationLength[1] = 4;
fragmentation.fragmentationOffset = new uint32_t[2];
fragmentation.fragmentationOffset = new size_t[2];
fragmentation.fragmentationOffset[0] = 0;
fragmentation.fragmentationOffset[1] = 4;
fragmentation.fragmentationTimeDiff = new uint16_t[2];

39
webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
View File

@ -73,43 +73,42 @@ class RtpRtcpVideoTest : public ::testing::Test {
payload_data_length_ = sizeof(video_frame_);
for (int n = 0; n < payload_data_length_; n++) {
for (size_t n = 0; n < payload_data_length_; n++) {
video_frame_[n] = n%10;
}
}
int32_t BuildRTPheader(uint8_t* dataBuffer,
uint32_t timestamp,
uint32_t sequence_number) {
size_t BuildRTPheader(uint8_t* dataBuffer,
uint32_t timestamp,
uint32_t sequence_number) {
dataBuffer[0] = static_cast<uint8_t>(0x80); // version 2
dataBuffer[1] = static_cast<uint8_t>(kPayloadType);
RtpUtility::AssignUWord16ToBuffer(dataBuffer + 2, sequence_number);
RtpUtility::AssignUWord32ToBuffer(dataBuffer + 4, timestamp);
RtpUtility::AssignUWord32ToBuffer(dataBuffer + 8, 0x1234); // SSRC.
int32_t rtpHeaderLength = 12;
size_t rtpHeaderLength = 12;
return rtpHeaderLength;
}
int PaddingPacket(uint8_t* buffer,
uint32_t timestamp,
uint32_t sequence_number,
int32_t bytes) {
size_t PaddingPacket(uint8_t* buffer,
uint32_t timestamp,
uint32_t sequence_number,
size_t bytes) {
// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
int max_length = 224;
size_t max_length = 224;
int padding_bytes_in_packet = max_length;
size_t padding_bytes_in_packet = max_length;
if (bytes < max_length) {
padding_bytes_in_packet = (bytes + 16) & 0xffe0; // Keep our modulus 32.
}
// Correct seq num, timestamp and payload type.
int header_length = BuildRTPheader(buffer, timestamp,
sequence_number);
size_t header_length = BuildRTPheader(buffer, timestamp, sequence_number);
buffer[0] |= 0x20; // Set padding bit.
int32_t* data =
reinterpret_cast<int32_t*>(&(buffer[header_length]));
// Fill data buffer with random data.
for (int j = 0; j < (padding_bytes_in_packet >> 2); j++) {
for (size_t j = 0; j < (padding_bytes_in_packet >> 2); j++) {
data[j] = rand(); // NOLINT
}
// Set number of padding bytes in the last byte of the packet.
@ -135,7 +134,7 @@ class RtpRtcpVideoTest : public ::testing::Test {
uint32_t test_timestamp_;
uint16_t test_sequence_number_;
uint8_t video_frame_[65000];
int payload_data_length_;
size_t payload_data_length_;
SimulatedClock fake_clock;
enum { kPayloadType = 100 };
};
@ -150,7 +149,7 @@ TEST_F(RtpRtcpVideoTest, BasicVideo) {
}
TEST_F(RtpRtcpVideoTest, PaddingOnlyFrames) {
const int kPadSize = 255;
const size_t kPadSize = 255;
uint8_t padding_packet[kPadSize];
uint32_t seq_num = 0;
uint32_t timestamp = 3000;
@ -165,8 +164,8 @@ TEST_F(RtpRtcpVideoTest, PaddingOnlyFrames) {
codec.maxBitrate));
for (int frame_idx = 0; frame_idx < 10; ++frame_idx) {
for (int packet_idx = 0; packet_idx < 5; ++packet_idx) {
int packet_size = PaddingPacket(padding_packet, timestamp, seq_num,
kPadSize);
size_t packet_size = PaddingPacket(padding_packet, timestamp, seq_num,
kPadSize);
++seq_num;
RTPHeader header;
scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
@ -175,11 +174,11 @@ TEST_F(RtpRtcpVideoTest, PaddingOnlyFrames) {
EXPECT_TRUE(rtp_payload_registry_.GetPayloadSpecifics(header.payloadType,
&payload_specific));
const uint8_t* payload = padding_packet + header.headerLength;
const int payload_length = packet_size - header.headerLength;
const size_t payload_length = packet_size - header.headerLength;
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, payload,
payload_length,
payload_specific, true));
EXPECT_EQ(0, receiver_->payload_size());
EXPECT_EQ(0u, receiver_->payload_size());
EXPECT_EQ(payload_length, receiver_->rtp_header().header.paddingLength);
}
timestamp += 3000;