Index: webrtc/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..129ae24d904486a11c9357c0dce4bd525c5bed5b |
--- /dev/null |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc |
@@ -0,0 +1,80 @@ |
+/* |
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
+ * |
+ * Use of this source code is governed by a BSD-style license |
+ * that can be found in the LICENSE file in the root of the source |
+ * tree. An additional intellectual property rights grant can be found |
+ * in the file PATENTS. All contributing project authors may |
+ * be found in the AUTHORS file in the root of the source tree. |
+ */ |
+ |
+#include <memory> |
+#include <vector> |
+ |
+#include "webrtc/base/array_view.h" |
+#include "webrtc/modules/include/module_common_types.h" |
+#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" |
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" |
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" |
+#include "webrtc/test/gmock.h" |
+#include "webrtc/test/gtest.h" |
+ |
+namespace webrtc { |
+namespace RtpFormatVideoGeneric { |
danilchap
2017/05/12 18:56:15
do not use this namespace
ilnik
2017/05/15 09:38:33
Done.
|
+namespace { |
+ |
+using ::testing::ElementsAreArray; |
+ |
+constexpr RtpPacketToSend::ExtensionManager* kNoExtensions = nullptr; |
+const size_t kMaxPayloadSize = 1200; |
+// const size_t kGenericHeaderLen = 1; |
+ |
+uint8_t test_payload_buffer[kMaxPayloadSize]; |
+ |
+void VerifyPacketsSizes(size_t payload_length, size_t max_payload_length, |
+ size_t extensions_length, |
+ std::vector<size_t> expected_sizes) { |
danilchap
2017/05/12 18:56:15
Likely it is more readable when helpers do not con
ilnik
2017/05/15 09:38:33
Done.
|
+ size_t total_expected_bytes = 0; |
+ for (size_t i = 0; i < expected_sizes.size(); i++) |
+ total_expected_bytes += expected_sizes[i] - 1; |
+ RTC_CHECK_EQ(total_expected_bytes, payload_length) << "Incorrect unittest. " |
+ "Sum of packets sizes, sans 1-byte headers, should be equal to payload " |
+ "length"; |
+ RtpPacketizerGeneric packetizer(kVideoFrameKey, |
+ max_payload_length, |
+ extensions_length); |
+ size_t num_packets = |
+ packetizer.SetPayloadData(test_payload_buffer, payload_length, nullptr); |
+ ASSERT_EQ(num_packets, expected_sizes.size()); |
+ RtpPacketToSend packet(kNoExtensions); |
danilchap
2017/05/12 18:56:15
alternative ways are
... packet(nullptr /* extensi
ilnik
2017/05/15 09:38:33
Done.
|
+ for (size_t i = 0; i < num_packets; i++) { |
+ EXPECT_TRUE(packetizer.NextPacket(&packet)); |
+ EXPECT_EQ(packet.payload_size(), expected_sizes[i]); |
+ } |
+ EXPECT_FALSE(packetizer.NextPacket(&packet)); |
+} |
+ |
+} // namespace |
+ |
+ |
+TEST(RtpPacketizerVideoGeneric, FourPackets) { |
+ VerifyPacketsSizes(12, 5, 2, {5, 5, 4, 2}); |
+} |
+ |
+TEST(RtpPacketizerVideoGeneric, NoExtensions) { |
+ VerifyPacketsSizes(12, 5, 0, {5, 5, 5}); |
+} |
+ |
+TEST(RtpPacketizerVideoGeneric, FitsInSinglePacket) { |
+ VerifyPacketsSizes(12, 16, 0, {13}); |
+} |
+ |
+TEST(RtpPacketizerVideoGeneric, ExtensionforsesSecondPacket) { |
+ VerifyPacketsSizes(12, 16, 4, {9, 5}); |
+} |
+ |
+ |
+ |
+} // namespace RtpFormatVideoGeneric |
+} // namespace webrtc |