RTP video playback tool using Call APIs.
Plays back rtpdump files from Wireshark in realtime as well as save the
resulting raw video to file. Unlike the RTP playback tool it doesn't
support faster-than-realtime playback/rendering, but it instead utilizes
the same path as production code and also contains support for playing
back FEC.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6838 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/test/rtp_file_reader_unittest.cc b/webrtc/test/rtp_file_reader_unittest.cc
new file mode 100644
index 0000000..b5fa260
--- /dev/null
+++ b/webrtc/test/rtp_file_reader_unittest.cc
@@ -0,0 +1,111 @@
+/*
+ * Copyright (c) 2013 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 <map>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/rtp_file_reader.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+class TestRtpFileReader : public ::testing::Test {
+ public:
+ void Init(const std::string& filename) {
+ std::string filepath =
+ test::ResourcePath("video_coding/" + filename, "rtp");
+ rtp_packet_source_.reset(
+ test::RtpFileReader::Create(test::RtpFileReader::kRtpDump, filepath));
+ ASSERT_TRUE(rtp_packet_source_.get() != NULL);
+ }
+
+ int CountRtpPackets() {
+ test::RtpFileReader::Packet packet;
+ int c = 0;
+ while (rtp_packet_source_->NextPacket(&packet))
+ c++;
+ return c;
+ }
+
+ private:
+ scoped_ptr<test::RtpFileReader> rtp_packet_source_;
+};
+
+TEST_F(TestRtpFileReader, Test60Packets) {
+ Init("pltype103");
+ EXPECT_EQ(60, CountRtpPackets());
+}
+
+typedef std::map<uint32_t, int> PacketsPerSsrc;
+
+class TestPcapFileReader : public ::testing::Test {
+ public:
+ void Init(const std::string& filename) {
+ std::string filepath =
+ test::ResourcePath("video_coding/" + filename, "pcap");
+ rtp_packet_source_.reset(
+ test::RtpFileReader::Create(test::RtpFileReader::kPcap, filepath));
+ ASSERT_TRUE(rtp_packet_source_.get() != NULL);
+ }
+
+ int CountRtpPackets() {
+ int c = 0;
+ test::RtpFileReader::Packet packet;
+ while (rtp_packet_source_->NextPacket(&packet))
+ c++;
+ return c;
+ }
+
+ PacketsPerSsrc CountRtpPacketsPerSsrc() {
+ PacketsPerSsrc pps;
+ test::RtpFileReader::Packet packet;
+ while (rtp_packet_source_->NextPacket(&packet)) {
+ RtpUtility::RtpHeaderParser rtp_header_parser(packet.data, packet.length);
+ webrtc::RTPHeader header;
+ if (!rtp_header_parser.RTCP() && rtp_header_parser.Parse(header, NULL)) {
+ pps[header.ssrc]++;
+ }
+ }
+ return pps;
+ }
+
+ private:
+ scoped_ptr<test::RtpFileReader> rtp_packet_source_;
+};
+
+TEST_F(TestPcapFileReader, TestEthernetIIFrame) {
+ Init("frame-ethernet-ii");
+ EXPECT_EQ(368, CountRtpPackets());
+}
+
+TEST_F(TestPcapFileReader, TestLoopbackFrame) {
+ Init("frame-loopback");
+ EXPECT_EQ(491, CountRtpPackets());
+}
+
+TEST_F(TestPcapFileReader, TestTwoSsrc) {
+ Init("ssrcs-2");
+ PacketsPerSsrc pps = CountRtpPacketsPerSsrc();
+ EXPECT_EQ(2UL, pps.size());
+ EXPECT_EQ(370, pps[0x78d48f61]);
+ EXPECT_EQ(60, pps[0xae94130b]);
+}
+
+TEST_F(TestPcapFileReader, TestThreeSsrc) {
+ Init("ssrcs-3");
+ PacketsPerSsrc pps = CountRtpPacketsPerSsrc();
+ EXPECT_EQ(3UL, pps.size());
+ EXPECT_EQ(162, pps[0x938c5eaa]);
+ EXPECT_EQ(113, pps[0x59fe6ef0]);
+ EXPECT_EQ(61, pps[0xed2bd2ac]);
+}
+} // namespace webrtc