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

Issue 2871173008: Fix packetization logic to leave space for extensions in the last packet (Closed)
Patch Set: Implement generic video packetizer unittests Created 3 years, 7 months ago
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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

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