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
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
index 6667afc..794c983 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -43,7 +43,7 @@
Packet* RtpFileSource::NextPacket() {
while (true) {
- RtpFileReader::Packet temp_packet;
+ RtpPacket temp_packet;
if (!rtp_reader_->NextPacket(&temp_packet)) {
return NULL;
}
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
index 45a03f9..b8c379a 100644
--- a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
+++ b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
@@ -69,7 +69,7 @@
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;
}
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
index a85bca4..12deb6c 100644
--- a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
+++ b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
@@ -43,7 +43,7 @@
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);
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.cc b/webrtc/modules/video_coding/main/test/rtp_player.cc
index 81295ab..02ae7c2 100644
--- a/webrtc/modules/video_coding/main/test/rtp_player.cc
+++ b/webrtc/modules/video_coding/main/test/rtp_player.cc
@@ -450,7 +450,7 @@
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_;
diff --git a/webrtc/test/rtp_file_reader.cc b/webrtc/test/rtp_file_reader.cc
index fd3116e..19531ed 100644
--- a/webrtc/test/rtp_file_reader.cc
+++ b/webrtc/test/rtp_file_reader.cc
@@ -100,9 +100,9 @@
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 @@
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);
diff --git a/webrtc/test/rtp_file_reader.h b/webrtc/test/rtp_file_reader.h
index 095ce76..f309380 100644
--- a/webrtc/test/rtp_file_reader.h
+++ b/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 @@
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
diff --git a/webrtc/test/rtp_file_reader_unittest.cc b/webrtc/test/rtp_file_reader_unittest.cc
index 54fb874..713597d 100644
--- a/webrtc/test/rtp_file_reader_unittest.cc
+++ b/webrtc/test/rtp_file_reader_unittest.cc
@@ -30,7 +30,7 @@
}
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 @@
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 @@
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;
diff --git a/webrtc/video/replay.cc b/webrtc/video/replay.cc
index 5cfb06f..ee05c4f 100644
--- a/webrtc/video/replay.cc
+++ b/webrtc/video/replay.cc
@@ -238,7 +238,7 @@
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;
