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
This commit is contained in:
stefan@webrtc.org 2014-05-20 13:09:16 +00:00
parent be4ab99a53
commit e208458643
2 changed files with 6 additions and 3 deletions

View File

@ -103,10 +103,11 @@ class RtpFileReaderImpl : public RtpPacketSourceInterface {
}
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 @@ class RtpFileReaderImpl : public RtpPacketSourceInterface {
*length = len;
*time_ms = offset;
*original_length = plen;
return kResultSuccess;
}

View File

@ -53,9 +53,10 @@ class RtpPacketSourceInterface {
// 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