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      1 /*
      2  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
      3  *
      4  *  Use of this source code is governed by a BSD-style license
      5  *  that can be found in the LICENSE file in the root of the source
      6  *  tree. An additional intellectual property rights grant can be found
      7  *  in the file PATENTS.  All contributing project authors may
      8  *  be found in the AUTHORS file in the root of the source tree.
      9  */
     10 
     11 #include "testing/gtest/include/gtest/gtest.h"
     12 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
     13 #include "webrtc/modules/video_coding/main/test/rtp_file_reader.h"
     14 #include "webrtc/modules/video_coding/main/test/rtp_player.h"
     15 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
     16 #include "webrtc/test/testsupport/fileutils.h"
     17 
     18 namespace webrtc {
     19 namespace rtpplayer {
     20 
     21 class TestRtpFileReader : public ::testing::Test {
     22  public:
     23   void Init(const std::string& filename) {
     24     std::string filepath =
     25         test::ResourcePath("video_coding/" + filename, "rtp");
     26     rtp_packet_source_.reset(CreateRtpFileReader(filepath));
     27     ASSERT_TRUE(rtp_packet_source_.get() != NULL);
     28   }
     29 
     30   int CountRtpPackets() {
     31     const uint32_t kBufferSize = 4096;
     32     uint8_t data[kBufferSize];
     33     uint32_t length = kBufferSize;
     34     uint32_t dummy_time_ms = 0;
     35     int c = 0;
     36     while (rtp_packet_source_->NextPacket(data, &length, &dummy_time_ms) == 0) {
     37       EXPECT_GE(kBufferSize, length);
     38       length = kBufferSize;
     39       c++;
     40     }
     41     return c;
     42   }
     43 
     44  private:
     45   scoped_ptr<RtpPacketSourceInterface> rtp_packet_source_;
     46 };
     47 
     48 TEST_F(TestRtpFileReader, Test60Packets) {
     49   Init("pltype103");
     50   EXPECT_EQ(60, CountRtpPackets());
     51 }
     52 
     53 }  // namespace rtpplayer
     54 }  // namespace webrtc
     55