| /* |
| * 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 "testing/gtest/include/gtest/gtest.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
| #include "webrtc/modules/video_coding/main/test/rtp_file_reader.h" |
| #include "webrtc/modules/video_coding/main/test/rtp_player.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| #include "webrtc/test/testsupport/fileutils.h" |
| |
| namespace webrtc { |
| namespace rtpplayer { |
| |
| 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(CreateRtpFileReader(filepath)); |
| ASSERT_TRUE(rtp_packet_source_.get() != NULL); |
| } |
| |
| int CountRtpPackets() { |
| const uint32_t kBufferSize = 4096; |
| uint8_t data[kBufferSize]; |
| uint32_t length = kBufferSize; |
| uint32_t dummy_time_ms = 0; |
| int c = 0; |
| while (rtp_packet_source_->NextPacket(data, &length, &dummy_time_ms) == 0) { |
| EXPECT_GE(kBufferSize, length); |
| length = kBufferSize; |
| c++; |
| } |
| return c; |
| } |
| |
| private: |
| scoped_ptr<RtpPacketSourceInterface> rtp_packet_source_; |
| }; |
| |
| TEST_F(TestRtpFileReader, Test60Packets) { |
| Init("pltype103"); |
| EXPECT_EQ(60, CountRtpPackets()); |
| } |
| |
| } // namespace rtpplayer |
| } // namespace webrtc |