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webrtcvideoengine_unittest: Use RtpHeaderParser class for parsing rtp header.

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Removes parsing code in ParseRtpPacket.

Bug: none
Change-Id: I190b3f8abd5f1881cf19e14b7f31cc8f85b5f156
Reviewed-on: https://webrtc-review.googlesource.com/86823
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23829}
This commit is contained in:
Åsa Persson 2018-07-03 10:40:40 +02:00 committed by Commit Bot
parent b3d2119af7
commit 23cd45ac2e
2 changed files with 26 additions and 89 deletions
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1
media/BUILD.gn
View File

@ -488,6 +488,7 @@ if (rtc_include_tests) {
":rtc_data",
"../api/video:video_frame_i420",
"../modules/audio_processing:mocks",
"../modules/rtp_rtcp",
"../modules/video_coding:video_codec_interface",
"../modules/video_coding:webrtc_vp8",
"../pc:rtc_pc",

114
media/engine/webrtcvideoengine_unittest.cc
View File

@ -38,6 +38,7 @@
#include "media/engine/simulcast.h"
#include "media/engine/webrtcvideoengine.h"
#include "media/engine/webrtcvoiceengine.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/gunit.h"
#include "rtc_base/numerics/safe_conversions.h"
@ -1272,82 +1273,16 @@ class WebRtcVideoChannelBaseTest : public testing::Test {
return network_interface_.GetRtpPacket(index);
}
static int GetPayloadType(const rtc::CopyOnWriteBuffer* p) {
int pt = -1;
ParseRtpPacket(p, NULL, &pt, NULL, NULL, NULL, NULL);
return pt;
webrtc::RTPHeader header;
EXPECT_TRUE(ParseRtpPacket(p, &header));
return header.payloadType;
}
static bool ParseRtpPacket(const rtc::CopyOnWriteBuffer* p,
bool* x,
int* pt,
int* seqnum,
uint32_t* tstamp,
uint32_t* ssrc,
std::string* payload) {
rtc::ByteBufferReader buf(p->data<char>(), p->size());
uint8_t u08 = 0;
uint16_t u16 = 0;
uint32_t u32 = 0;
// Read X and CC fields.
if (!buf.ReadUInt8(&u08))
return false;
bool extension = ((u08 & 0x10) != 0);
uint8_t cc = (u08 & 0x0F);
if (x)
*x = extension;
// Read PT field.
if (!buf.ReadUInt8(&u08))
return false;
if (pt)
*pt = (u08 & 0x7F);
// Read Sequence Number field.
if (!buf.ReadUInt16(&u16))
return false;
if (seqnum)
*seqnum = u16;
// Read Timestamp field.
if (!buf.ReadUInt32(&u32))
return false;
if (tstamp)
*tstamp = u32;
// Read SSRC field.
if (!buf.ReadUInt32(&u32))
return false;
if (ssrc)
*ssrc = u32;
// Skip CSRCs.
for (uint8_t i = 0; i < cc; ++i) {
if (!buf.ReadUInt32(&u32))
return false;
}
// Skip extension header.
if (extension) {
// Read Profile-specific extension header ID
if (!buf.ReadUInt16(&u16))
return false;
// Read Extension header length
if (!buf.ReadUInt16(&u16))
return false;
uint16_t ext_header_len = u16;
// Read Extension header
for (uint16_t i = 0; i < ext_header_len; ++i) {
if (!buf.ReadUInt32(&u32))
return false;
}
}
if (payload) {
return buf.ReadString(payload, buf.Length());
}
return true;
webrtc::RTPHeader* header) {
std::unique_ptr<webrtc::RtpHeaderParser> parser(
webrtc::RtpHeaderParser::Create());
return parser->Parse(p->cdata(), p->size(), header);
}
// Tests that we can send and receive frames.
@ -1649,14 +1584,15 @@ TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrc) {
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(SendFrame());
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
uint32_t ssrc = 0;
webrtc::RTPHeader header;
std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
EXPECT_EQ(kSsrc, ssrc);
EXPECT_TRUE(ParseRtpPacket(p.get(), &header));
EXPECT_EQ(kSsrc, header.ssrc);
// Packets are being paced out, so these can mismatch between the first and
// second call to NumRtpPackets until pending packets are paced out.
EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(ssrc), kTimeout);
EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(ssrc), kTimeout);
EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(header.ssrc), kTimeout);
EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(header.ssrc), kTimeout);
EXPECT_EQ(1, NumSentSsrcs());
EXPECT_EQ(0, NumRtpPackets(kSsrc - 1));
EXPECT_EQ(0, NumRtpBytes(kSsrc - 1));
@ -1673,14 +1609,14 @@ TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrcAfterSetCodecs) {
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(WaitAndSendFrame(0));
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
uint32_t ssrc = 0;
webrtc::RTPHeader header;
std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
EXPECT_EQ(999u, ssrc);
EXPECT_TRUE(ParseRtpPacket(p.get(), &header));
EXPECT_EQ(999u, header.ssrc);
// Packets are being paced out, so these can mismatch between the first and
// second call to NumRtpPackets until pending packets are paced out.
EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(ssrc), kTimeout);
EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(ssrc), kTimeout);
EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(header.ssrc), kTimeout);
EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(header.ssrc), kTimeout);
EXPECT_EQ(1, NumSentSsrcs());
EXPECT_EQ(0, NumRtpPackets(kSsrc));
EXPECT_EQ(0, NumRtpBytes(kSsrc));
@ -1712,12 +1648,12 @@ TEST_F(WebRtcVideoChannelBaseTest, AddRemoveSendStreams) {
EXPECT_TRUE(SendFrame());
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_GT(NumRtpPackets(), 0);
uint32_t ssrc = 0;
webrtc::RTPHeader header;
size_t last_packet = NumRtpPackets() - 1;
std::unique_ptr<const rtc::CopyOnWriteBuffer> p(
GetRtpPacket(static_cast<int>(last_packet)));
ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
EXPECT_EQ(kSsrc, ssrc);
EXPECT_TRUE(ParseRtpPacket(p.get(), &header));
EXPECT_EQ(kSsrc, header.ssrc);
// Remove the send stream that was added during Setup.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
@ -1733,8 +1669,8 @@ TEST_F(WebRtcVideoChannelBaseTest, AddRemoveSendStreams) {
last_packet = NumRtpPackets() - 1;
p.reset(GetRtpPacket(static_cast<int>(last_packet)));
ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
EXPECT_EQ(789u, ssrc);
EXPECT_TRUE(ParseRtpPacket(p.get(), &header));
EXPECT_EQ(789u, header.ssrc);
}
// Tests the behavior of incoming streams in a conference scenario.