Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(252)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc

Issue 2528343002: H.264 packetization mode 0 (try 3) (Closed)
Patch Set: Changed from enum to enum class Created 4 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc ('K') | « webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc ('k') | webrtc/modules/video_coding/BUILD.gn » ('j') | webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc » ('J')
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index 894415da8a7960d70bad206f115182e67e35f235..82d569ac94d1867134f7129be53de1e60d797dbc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -43,6 +43,27 @@ enum NalDefs { kFBit = 0x80, kNriMask = 0x60, kTypeMask = 0x1F };
// Bit masks for FU (A and B) headers.
enum FuDefs { kSBit = 0x80, kEBit = 0x40, kRBit = 0x20 };
+void CreateThreeFragments(RTPFragmentationHeader* fragmentation,
+ size_t frameSize,
+ size_t payloadOffset) {
+ fragmentation->VerifyAndAllocateFragmentationHeader(3);
+ fragmentation->fragmentationOffset[0] = 0;
+ fragmentation->fragmentationLength[0] = 2;
+ fragmentation->fragmentationOffset[1] = 2;
+ fragmentation->fragmentationLength[1] = 2;
+ fragmentation->fragmentationOffset[2] = 4;
+ fragmentation->fragmentationLength[2] =
+ kNalHeaderSize + frameSize - payloadOffset;
+}
+
+RtpPacketizer* CreateH264Packetizer(H264PacketizationMode mode,
+ size_t max_payload_size) {
+ RTPVideoTypeHeader type_header;
+ type_header.H264.packetization_mode = mode;
+ return RtpPacketizer::Create(kRtpVideoH264, max_payload_size, &type_header,
+ kEmptyFrame);
+}
+
void VerifyFua(size_t fua_index,
const uint8_t* expected_payload,
int offset,
@@ -84,8 +105,8 @@ void TestFua(size_t frame_size,
fragmentation.VerifyAndAllocateFragmentationHeader(1);
fragmentation.fragmentationOffset[0] = 0;
fragmentation.fragmentationLength[0] = frame_size;
- std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
- kRtpVideoH264, max_payload_size, NULL, kEmptyFrame));
+ std::unique_ptr<RtpPacketizer> packetizer(CreateH264Packetizer(
+ H264PacketizationMode::NonInterleaved, max_payload_size));
packetizer->SetPayloadData(frame.get(), frame_size, &fragmentation);
std::unique_ptr<uint8_t[]> packet(new uint8_t[max_payload_size]);
@@ -152,14 +173,19 @@ void VerifySingleNaluPayload(const RTPFragmentationHeader& fragmentation,
}
} // namespace
-TEST(RtpPacketizerH264Test, TestSingleNalu) {
+// Tests that should work with both packetization mode 0 and
+// packetization mode 1.
+class RtpPacketizerH264ModeTest
+ : public ::testing::TestWithParam<H264PacketizationMode> {};
+
+TEST_P(RtpPacketizerH264ModeTest, TestSingleNalu) {
const uint8_t frame[2] = {0x05, 0xFF}; // F=0, NRI=0, Type=5.
RTPFragmentationHeader fragmentation;
fragmentation.VerifyAndAllocateFragmentationHeader(1);
fragmentation.fragmentationOffset[0] = 0;
fragmentation.fragmentationLength[0] = sizeof(frame);
std::unique_ptr<RtpPacketizer> packetizer(
- RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
+ CreateH264Packetizer(GetParam(), kMaxPayloadSize));
packetizer->SetPayloadData(frame, sizeof(frame), &fragmentation);
uint8_t packet[kMaxPayloadSize] = {0};
size_t length = 0;
@@ -167,12 +193,12 @@ TEST(RtpPacketizerH264Test, TestSingleNalu) {
ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last));
EXPECT_EQ(2u, length);
EXPECT_TRUE(last);
- VerifySingleNaluPayload(
- fragmentation, 0, frame, sizeof(frame), packet, length);
+ VerifySingleNaluPayload(fragmentation, 0, frame, sizeof(frame), packet,
+ length);
EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last));
}
-TEST(RtpPacketizerH264Test, TestSingleNaluTwoPackets) {
+TEST_P(RtpPacketizerH264ModeTest, TestSingleNaluTwoPackets) {
const size_t kFrameSize = kMaxPayloadSize + 100;
uint8_t frame[kFrameSize] = {0};
for (size_t i = 0; i < kFrameSize; ++i)
@@ -188,7 +214,7 @@ TEST(RtpPacketizerH264Test, TestSingleNaluTwoPackets) {
frame[fragmentation.fragmentationOffset[1]] = 0x01;
std::unique_ptr<RtpPacketizer> packetizer(
- RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
+ CreateH264Packetizer(GetParam(), kMaxPayloadSize));
packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
uint8_t packet[kMaxPayloadSize] = {0};
@@ -206,6 +232,12 @@ TEST(RtpPacketizerH264Test, TestSingleNaluTwoPackets) {
EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last));
}
+INSTANTIATE_TEST_CASE_P(
+ PacketMode,
+ RtpPacketizerH264ModeTest,
+ ::testing::Values(H264PacketizationMode::SingleNalUnit,
+ H264PacketizationMode::NonInterleaved));
+
TEST(RtpPacketizerH264Test, TestStapA) {
const size_t kFrameSize =
kMaxPayloadSize - 3 * kLengthFieldLength - kNalHeaderSize;
@@ -216,16 +248,9 @@ TEST(RtpPacketizerH264Test, TestStapA) {
for (size_t i = 0; i < kFrameSize - kPayloadOffset; ++i)
frame[i + kPayloadOffset] = i;
RTPFragmentationHeader fragmentation;
- fragmentation.VerifyAndAllocateFragmentationHeader(3);
- fragmentation.fragmentationOffset[0] = 0;
- fragmentation.fragmentationLength[0] = 2;
- fragmentation.fragmentationOffset[1] = 2;
- fragmentation.fragmentationLength[1] = 2;
- fragmentation.fragmentationOffset[2] = 4;
- fragmentation.fragmentationLength[2] =
- kNalHeaderSize + kFrameSize - kPayloadOffset;
- std::unique_ptr<RtpPacketizer> packetizer(
- RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
+ CreateThreeFragments(&fragmentation, kFrameSize, kPayloadOffset);
+ std::unique_ptr<RtpPacketizer> packetizer(CreateH264Packetizer(
+ H264PacketizationMode::NonInterleaved, kMaxPayloadSize));
packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
uint8_t packet[kMaxPayloadSize] = {0};
@@ -242,6 +267,32 @@ TEST(RtpPacketizerH264Test, TestStapA) {
EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last));
}
+TEST(RtpPacketizerH264Test, TestMode0HasNoStapA) {
hbos 2016/12/02 10:21:26 nit/optional: Naming difference "mode" vs enum nam
hta-webrtc 2016/12/02 11:09:01 Done.
+ // This is the same setup as for the TestStapA test.
+ const size_t kFrameSize =
+ kMaxPayloadSize - 3 * kLengthFieldLength - kNalHeaderSize;
+ uint8_t frame[kFrameSize] = {0x07, 0xFF, // F=0, NRI=0, Type=7 (SPS).
+ 0x08, 0xFF, // F=0, NRI=0, Type=8 (PPS).
+ 0x05}; // F=0, NRI=0, Type=5 (IDR).
+ const size_t kPayloadOffset = 5;
+ for (size_t i = 0; i < kFrameSize - kPayloadOffset; ++i)
+ frame[i + kPayloadOffset] = i;
+ RTPFragmentationHeader fragmentation;
+ CreateThreeFragments(&fragmentation, kFrameSize, kPayloadOffset);
+ std::unique_ptr<RtpPacketizer> packetizer(CreateH264Packetizer(
+ H264PacketizationMode::SingleNalUnit, kMaxPayloadSize));
+ packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
+
+ uint8_t packet[kMaxPayloadSize] = {0};
+ size_t length = 0;
+ bool last = false;
+ // The three fragments should be returned as three packets.
+ ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last));
+ ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last));
+ ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last));
+ EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last));
+}
+
TEST(RtpPacketizerH264Test, TestTooSmallForStapAHeaders) {
const size_t kFrameSize = kMaxPayloadSize - 1;
uint8_t frame[kFrameSize] = {0x07, 0xFF, // F=0, NRI=0, Type=7.
@@ -259,8 +310,8 @@ TEST(RtpPacketizerH264Test, TestTooSmallForStapAHeaders) {
fragmentation.fragmentationOffset[2] = 4;
fragmentation.fragmentationLength[2] =
kNalHeaderSize + kFrameSize - kPayloadOffset;
- std::unique_ptr<RtpPacketizer> packetizer(
- RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
+ std::unique_ptr<RtpPacketizer> packetizer(CreateH264Packetizer(
+ H264PacketizationMode::NonInterleaved, kMaxPayloadSize));
packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
uint8_t packet[kMaxPayloadSize] = {0};
@@ -307,8 +358,8 @@ TEST(RtpPacketizerH264Test, TestMixedStapA_FUA) {
frame[nalu_offset + j] = i + j;
}
}
- std::unique_ptr<RtpPacketizer> packetizer(
- RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
+ std::unique_ptr<RtpPacketizer> packetizer(CreateH264Packetizer(
+ H264PacketizationMode::NonInterleaved, kMaxPayloadSize));
packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
// First expecting two FU-A packets.
@@ -381,6 +432,28 @@ TEST(RtpPacketizerH264Test, TestFUABig) {
sizeof(kExpectedPayloadSizes) / sizeof(size_t)));
}
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+TEST(RtpPacketizerH264DeathTest, SendOverlongDataInPacketizationMode0) {
+ const size_t kFrameSize = kMaxPayloadSize + 100;
hbos 2016/12/02 10:21:26 nit: I prefer to test the edge case + 1
hta-webrtc 2016/12/02 11:09:01 Done.
+ uint8_t frame[kFrameSize] = {0};
+ for (size_t i = 0; i < kFrameSize; ++i)
+ frame[i] = i;
+ RTPFragmentationHeader fragmentation;
+ fragmentation.VerifyAndAllocateFragmentationHeader(1);
+ fragmentation.fragmentationOffset[0] = 0;
+ fragmentation.fragmentationLength[0] = kFrameSize;
+ // Set NAL headers.
+ frame[fragmentation.fragmentationOffset[0]] = 0x01;
+
+ std::unique_ptr<RtpPacketizer> packetizer(CreateH264Packetizer(
+ H264PacketizationMode::SingleNalUnit, kMaxPayloadSize));
+ EXPECT_DEATH(packetizer->SetPayloadData(frame, kFrameSize, &fragmentation),
+ "payload_size");
+}
+
+#endif // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
namespace {
const uint8_t kStartSequence[] = {0x00, 0x00, 0x00, 0x01};
const uint8_t kOriginalSps[] = {kSps, 0x00, 0x00, 0x03, 0x03,
@@ -421,9 +494,9 @@ TEST_F(RtpPacketizerH264TestSpsRewriting, FuASps) {
const size_t kHeaderOverhead = kFuAHeaderSize + 1;
// Set size to fragment SPS into two FU-A packets.
- packetizer_.reset(RtpPacketizer::Create(
- kRtpVideoH264, sizeof(kOriginalSps) - 2 + kHeaderOverhead, nullptr,
- kEmptyFrame));
+ packetizer_.reset(
+ CreateH264Packetizer(H264PacketizationMode::NonInterleaved,
+ sizeof(kOriginalSps) - 2 + kHeaderOverhead));
packetizer_->SetPayloadData(in_buffer_.data(), in_buffer_.size(),
&fragmentation_header_);
@@ -459,9 +532,8 @@ TEST_F(RtpPacketizerH264TestSpsRewriting, StapASps) {
sizeof(kIdrTwo) + (kLengthFieldLength * 3);
// Set size to include SPS and the rest of the packets in a Stap-A package.
- packetizer_.reset(RtpPacketizer::Create(kRtpVideoH264,
- kExpectedTotalSize + kHeaderOverhead,
- nullptr, kEmptyFrame));
+ packetizer_.reset(CreateH264Packetizer(H264PacketizationMode::NonInterleaved,
+ kExpectedTotalSize + kHeaderOverhead));
packetizer_->SetPayloadData(in_buffer_.data(), in_buffer_.size(),
&fragmentation_header_);
« webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc ('K') | « webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc ('k') | webrtc/modules/video_coding/BUILD.gn » ('j') | webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc » ('J')

Powered by Google App Engine
This is Rietveld 408576698