Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader

This is in preparation for creating a new class RtpFileWriter which
will use the same RtpPacket struct.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7749 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc

@@ -43,7 +43,7 @@ bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
 
 Packet* RtpFileSource::NextPacket() {
   while (true) {
-    RtpFileReader::Packet temp_packet;
+    RtpPacket temp_packet;
     if (!rtp_reader_->NextPacket(&temp_packet)) {
       return NULL;
     }

webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc

@@ -69,7 +69,7 @@ int main(int argc, char** argv) {
   int non_zero_ts_offsets = 0;
   while (true) {
     if (next_rtp_time_ms <= clock.TimeInMilliseconds()) {
-      webrtc::test::RtpFileReader::Packet packet;
+      webrtc::test::RtpPacket packet;
       if (!rtp_reader->NextPacket(&packet)) {
         break;
       }

webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc

@@ -43,7 +43,7 @@ int main(int argc, char** argv) {
   int packet_counter = 0;
   int non_zero_abs_send_time = 0;
   int non_zero_ts_offsets = 0;
-  webrtc::test::RtpFileReader::Packet packet;
+  webrtc::test::RtpPacket packet;
   while (rtp_reader->NextPacket(&packet)) {
     webrtc::RTPHeader header;
     parser->Parse(packet.data, packet.length, &header);

webrtc/modules/video_coding/main/test/rtp_player.cc

@@ -450,7 +450,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
   SsrcHandlers ssrc_handlers_;
   Clock* clock_;
   scoped_ptr<test::RtpFileReader> packet_source_;
-  test::RtpFileReader::Packet next_packet_;
+  test::RtpPacket next_packet_;
   uint32_t next_rtp_time_;
   bool first_packet_;
   int64_t first_packet_rtp_time_;

webrtc/test/rtp_file_reader.cc

@@ -100,9 +100,9 @@ class RtpDumpReader : public RtpFileReaderImpl {
     return true;
   }
 
-  virtual bool NextPacket(Packet* packet) OVERRIDE {
+  virtual bool NextPacket(RtpPacket* packet) OVERRIDE {
     uint8_t* rtp_data = packet->data;
-    packet->length = Packet::kMaxPacketBufferSize;
+    packet->length = RtpPacket::kMaxPacketBufferSize;
 
     uint16_t len;
     uint16_t plen;
@@ -290,8 +290,8 @@ class PcapReader : public RtpFileReaderImpl {
     return kResultSuccess;
   }
 
-  virtual bool NextPacket(Packet* packet) OVERRIDE {
-    uint32_t length = Packet::kMaxPacketBufferSize;
+  virtual bool NextPacket(RtpPacket* packet) OVERRIDE {
+    uint32_t length = RtpPacket::kMaxPacketBufferSize;
     if (NextPcap(packet->data, &length, &packet->time_ms) != kResultSuccess)
       return false;
     packet->length = static_cast<size_t>(length);

webrtc/test/rtp_file_reader.h

@@ -16,6 +16,20 @@
 
 namespace webrtc {
 namespace test {
+
+struct RtpPacket {
+  // 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;
+};
+
 class RtpFileReader {
  public:
   enum FileFormat {
@@ -23,24 +37,11 @@ class RtpFileReader {
     kRtpDump,
   };
 
-  struct Packet {
-    // 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;
-  };
-
   virtual ~RtpFileReader() {}
   static RtpFileReader* Create(FileFormat format,
                                const std::string& filename);
 
-  virtual bool NextPacket(Packet* packet) = 0;
+  virtual bool NextPacket(RtpPacket* packet) = 0;
 };
 }  // namespace test
 }  // namespace webrtc

webrtc/test/rtp_file_reader_unittest.cc

@@ -30,7 +30,7 @@ class TestRtpFileReader : public ::testing::Test {
   }
 
   int CountRtpPackets() {
-    test::RtpFileReader::Packet packet;
+    test::RtpPacket packet;
     int c = 0;
     while (rtp_packet_source_->NextPacket(&packet)) {
       if (headers_only_file_)
@@ -71,7 +71,7 @@ class TestPcapFileReader : public ::testing::Test {
 
   int CountRtpPackets() {
     int c = 0;
-    test::RtpFileReader::Packet packet;
+    test::RtpPacket packet;
     while (rtp_packet_source_->NextPacket(&packet)) {
       EXPECT_EQ(packet.length, packet.original_length);
       c++;
@@ -81,7 +81,7 @@ class TestPcapFileReader : public ::testing::Test {
 
   PacketsPerSsrc CountRtpPacketsPerSsrc() {
     PacketsPerSsrc pps;
-    test::RtpFileReader::Packet packet;
+    test::RtpPacket packet;
     while (rtp_packet_source_->NextPacket(&packet)) {
       RtpUtility::RtpHeaderParser rtp_header_parser(packet.data, packet.length);
       webrtc::RTPHeader header;

webrtc/video/replay.cc

@@ -238,7 +238,7 @@ void RtpReplay() {
   int num_packets = 0;
   std::map<uint32_t, int> unknown_packets;
   while (true) {
-    test::RtpFileReader::Packet packet;
+    test::RtpPacket packet;
     if (!rtp_reader->NextPacket(&packet))
       break;
     ++num_packets;