Implementing a packet router class, used to route RTP packets to the
sending RTP module for the specified simulcast layer a frame belongs to.
This CL also removes the corresponding functionality from the RTP RTCP
module and fixes lint warnings in the files touched.
BUG=769
TEST=New unittest and manual tests
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39629004
Cr-Commit-Position: refs/heads/master@{#8267}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8267 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index bfc8266..8bde0a8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -13,6 +13,8 @@
#include <assert.h>
#include <string.h>
+#include <set>
+
#include "webrtc/common_types.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/trace.h"
@@ -491,74 +493,21 @@
size_t payload_size,
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtp_video_hdr) {
- rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
+ assert(!IsDefaultModule());
- if (!IsDefaultModule()) {
- // Don't send RTCP from default module.
- if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
+ rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
+ if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
- }
- return rtp_sender_.SendOutgoingData(frame_type,
- payload_type,
- time_stamp,
- capture_time_ms,
- payload_data,
- payload_size,
- fragmentation,
- NULL,
- &(rtp_video_hdr->codecHeader));
}
- int32_t ret_val = -1;
- CriticalSectionScoped lock(critical_section_module_ptrs_.get());
- if (simulcast_) {
- if (rtp_video_hdr == NULL) {
- return -1;
- }
- int idx = 0;
- std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin();
- for (; idx < rtp_video_hdr->simulcastIdx; ++it) {
- if (it == child_modules_.end()) {
- return -1;
- }
- if ((*it)->SendingMedia()) {
- ++idx;
- }
- }
- for (; it != child_modules_.end(); ++it) {
- if ((*it)->SendingMedia()) {
- break;
- }
- ++idx;
- }
- if (it == child_modules_.end()) {
- return -1;
- }
- return (*it)->SendOutgoingData(frame_type,
- payload_type,
- time_stamp,
- capture_time_ms,
- payload_data,
- payload_size,
- fragmentation,
- rtp_video_hdr);
- } else {
- std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin();
- // Send to all "child" modules
- while (it != child_modules_.end()) {
- if ((*it)->SendingMedia()) {
- ret_val = (*it)->SendOutgoingData(frame_type,
- payload_type,
- time_stamp,
- capture_time_ms,
- payload_data,
- payload_size,
- fragmentation,
- rtp_video_hdr);
- }
- it++;
- }
- }
- return ret_val;
+ return rtp_sender_.SendOutgoingData(frame_type,
+ payload_type,
+ time_stamp,
+ capture_time_ms,
+ payload_data,
+ payload_size,
+ fragmentation,
+ NULL,
+ &(rtp_video_hdr->codecHeader));
}
bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
@@ -1346,4 +1295,4 @@
return !child_modules_.empty();
}
-} // Namespace webrtc
+} // namespace webrtc
