Minor modifications to test::RtpFileReader
Adding original_length to the Packet struct. This is populated with
the plen value from the RTP dump file. In the case of reading a
pcap file, original_length will be equal to length.
Also increasing the maximum packet size to 3500 bytes. This is to
accomodate some test files that contain PCM16b audio encoding.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7333 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/resources/video_coding/pltype103_header_only.rtp.sha1 b/resources/video_coding/pltype103_header_only.rtp.sha1
new file mode 100644
index 0000000..eee5aa9
--- /dev/null
+++ b/resources/video_coding/pltype103_header_only.rtp.sha1
@@ -0,0 +1 @@
+2a77f0d030aa96f51f0a96e971b42c3b11fe006b
\ No newline at end of file
diff --git a/webrtc/test/rtp_file_reader.cc b/webrtc/test/rtp_file_reader.cc
index be8dc2b..fd3116e 100644
--- a/webrtc/test/rtp_file_reader.cc
+++ b/webrtc/test/rtp_file_reader.cc
@@ -16,6 +16,7 @@
#include <string>
#include <vector>
+#include "webrtc/base/checks.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
@@ -113,13 +114,17 @@
// Use 'len' here because a 'plen' of 0 specifies rtcp.
len -= kPacketHeaderSize;
if (packet->length < len) {
- return false;
+ FATAL() << "Packet is too large to fit: " << len << " bytes vs "
+ << packet->length
+ << " bytes allocated. Consider increasing the buffer "
+ "size";
}
if (fread(rtp_data, 1, len, file_) != len) {
return false;
}
packet->length = len;
+ packet->original_length = plen;
packet->time_ms = offset;
return true;
}
@@ -290,6 +295,7 @@
if (NextPcap(packet->data, &length, &packet->time_ms) != kResultSuccess)
return false;
packet->length = static_cast<size_t>(length);
+ packet->original_length = packet->length;
return true;
}
diff --git a/webrtc/test/rtp_file_reader.h b/webrtc/test/rtp_file_reader.h
index 379bf2d..095ce76 100644
--- a/webrtc/test/rtp_file_reader.h
+++ b/webrtc/test/rtp_file_reader.h
@@ -24,9 +24,14 @@
};
struct Packet {
- static const size_t kMaxPacketBufferSize = 1500;
+ // Accommodate for 50 ms packets of 32 kHz PCM16 samples (3200 bytes) plus
+ // some overhead.
+ static const size_t kMaxPacketBufferSize = 3500;
uint8_t data[kMaxPacketBufferSize];
size_t length;
+ // The length the packet had on wire. Will be different from |length| when
+ // reading a header-only RTP dump.
+ size_t original_length;
uint32_t time_ms;
};
diff --git a/webrtc/test/rtp_file_reader_unittest.cc b/webrtc/test/rtp_file_reader_unittest.cc
index b5fa260..54fb874 100644
--- a/webrtc/test/rtp_file_reader_unittest.cc
+++ b/webrtc/test/rtp_file_reader_unittest.cc
@@ -20,28 +20,40 @@
class TestRtpFileReader : public ::testing::Test {
public:
- void Init(const std::string& filename) {
+ void Init(const std::string& filename, bool headers_only_file) {
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);
+ headers_only_file_ = headers_only_file;
}
int CountRtpPackets() {
test::RtpFileReader::Packet packet;
int c = 0;
- while (rtp_packet_source_->NextPacket(&packet))
+ while (rtp_packet_source_->NextPacket(&packet)) {
+ if (headers_only_file_)
+ EXPECT_LT(packet.length, packet.original_length);
+ else
+ EXPECT_EQ(packet.length, packet.original_length);
c++;
+ }
return c;
}
private:
scoped_ptr<test::RtpFileReader> rtp_packet_source_;
+ bool headers_only_file_;
};
TEST_F(TestRtpFileReader, Test60Packets) {
- Init("pltype103");
+ Init("pltype103", false);
+ EXPECT_EQ(60, CountRtpPackets());
+}
+
+TEST_F(TestRtpFileReader, Test60PacketsHeaderOnly) {
+ Init("pltype103_header_only", true);
EXPECT_EQ(60, CountRtpPackets());
}
@@ -60,8 +72,10 @@
int CountRtpPackets() {
int c = 0;
test::RtpFileReader::Packet packet;
- while (rtp_packet_source_->NextPacket(&packet))
+ while (rtp_packet_source_->NextPacket(&packet)) {
+ EXPECT_EQ(packet.length, packet.original_length);
c++;
+ }
return c;
}
diff --git a/webrtc/test/webrtc_test_common.gyp b/webrtc/test/webrtc_test_common.gyp
index d422a4b..477ec0f 100644
--- a/webrtc/test/webrtc_test_common.gyp
+++ b/webrtc/test/webrtc_test_common.gyp
@@ -58,6 +58,7 @@
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+ '<(webrtc_root)/base/base.gyp:rtc_base',
'<(webrtc_root)/modules/modules.gyp:media_file',
'<(webrtc_root)/modules/modules.gyp:video_capture_module_impl',
'<(webrtc_root)/modules/modules.gyp:video_render_module_impl',
