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 6667afc5bd..794c9836f6 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 @@ 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; } 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 45a03f9d9d..b8c379ae54 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 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; } 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 a85bca4f64..12deb6c8a3 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 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); diff --git a/webrtc/modules/video_coding/main/test/rtp_player.cc b/webrtc/modules/video_coding/main/test/rtp_player.cc index 81295abc09..02ae7c2974 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 @@ class RtpPlayerImpl : public RtpPlayerInterface { SsrcHandlers ssrc_handlers_; Clock* clock_; scoped_ptr 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 fd3116ebb3..19531ed48e 100644 --- a/webrtc/test/rtp_file_reader.cc +++ b/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(length); diff --git a/webrtc/test/rtp_file_reader.h b/webrtc/test/rtp_file_reader.h index 095ce76726..f309380fbf 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 @@ 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 diff --git a/webrtc/test/rtp_file_reader_unittest.cc b/webrtc/test/rtp_file_reader_unittest.cc index 54fb874b63..713597db6a 100644 --- a/webrtc/test/rtp_file_reader_unittest.cc +++ b/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; diff --git a/webrtc/video/replay.cc b/webrtc/video/replay.cc index 5cfb06f8b8..ee05c4fbd2 100644 --- a/webrtc/video/replay.cc +++ b/webrtc/video/replay.cc @@ -238,7 +238,7 @@ void RtpReplay() { int num_packets = 0; std::map unknown_packets; while (true) { - test::RtpFileReader::Packet packet; + test::RtpPacket packet; if (!rtp_reader->NextPacket(&packet)) break; ++num_packets;