Expose the original packet length in in the RTP play tools.
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6198 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/video_coding/main/test/rtp_file_reader.cc b/webrtc/modules/video_coding/main/test/rtp_file_reader.cc
index 447eb9b..2ab5460 100644
--- a/webrtc/modules/video_coding/main/test/rtp_file_reader.cc
+++ b/webrtc/modules/video_coding/main/test/rtp_file_reader.cc
@@ -103,10 +103,11 @@
}
virtual int NextPacket(uint8_t* rtp_data, uint32_t* length,
- uint32_t* time_ms) {
+ uint32_t* time_ms, uint32_t* original_length) {
assert(rtp_data);
assert(length);
assert(time_ms);
+ assert(original_length);
uint16_t len;
uint16_t plen;
@@ -126,6 +127,7 @@
*length = len;
*time_ms = offset;
+ *original_length = plen;
return kResultSuccess;
}
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.h b/webrtc/modules/video_coding/main/test/rtp_player.h
index 24e62b6..c945b5c 100644
--- a/webrtc/modules/video_coding/main/test/rtp_player.h
+++ b/webrtc/modules/video_coding/main/test/rtp_player.h
@@ -53,9 +53,10 @@
// Read next RTP packet into buffer pointed to by rtp_data. On call, 'length'
// field must be filled in with the size of the buffer. The actual size of
// the packet is available in 'length' upon returning. Time in milliseconds
- // from start of stream is returned in 'time_ms'.
+ // from start of stream is returned in 'time_ms'. The original full length
+ // of the packet is returned in 'original_length'.
virtual int NextPacket(uint8_t* rtp_data, uint32_t* length,
- uint32_t* time_ms) = 0;
+ uint32_t* time_ms, uint32_t* original_length) = 0;
};
// Implemented by RtpPlayer and given to client as a means to retrieve
