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Add BWE tools for parsing RTP files.

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bwe_rtp_play feeds packets from an RTP file into the BWE and prints the estimates.
bwe_rtp_to_text parses an RTP file and outputs the result to a text file.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7689006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5466 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2014-01-31 09:15:48 +00:00
parent 668a23b402
commit 1dd9b4d98e
6 changed files with 359 additions and 2 deletions
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5
webrtc/modules/video_coding/main/test/rtp_player.cc
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@ -203,7 +203,7 @@ class SsrcHandlers {
}
}
int RegisterSsrc(uint32_t ssrc, LostPackets* lost_packets) {
int RegisterSsrc(uint32_t ssrc, LostPackets* lost_packets, Clock* clock) {
if (handlers_.count(ssrc) > 0) {
return 0;
}
@ -217,6 +217,7 @@ class SsrcHandlers {
}
RtpRtcp::Configuration configuration;
configuration.clock = clock;
configuration.id = 1;
configuration.audio = false;
handler->rtp_module_.reset(RtpReceiver::CreateVideoReceiver(
@ -434,7 +435,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
return -1;
}
uint32_t ssrc = header.ssrc;
if (ssrc_handlers_.RegisterSsrc(ssrc, &lost_packets_) < 0) {
if (ssrc_handlers_.RegisterSsrc(ssrc, &lost_packets_, clock_) < 0) {
DEBUG_LOG1("Unable to register ssrc: %d", ssrc);
return -1;
}