Preserve RTP states for restarted VideoSendStreams.
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.
Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
index 4d32ab4..ff3fe50 100644
--- a/video/video_send_stream.cc
+++ b/video/video_send_stream.cc
@@ -108,18 +108,21 @@
}
namespace internal {
-VideoSendStream::VideoSendStream(newapi::Transport* transport,
- CpuOveruseObserver* overuse_observer,
- webrtc::VideoEngine* video_engine,
- const VideoSendStream::Config& config,
- const std::vector<VideoStream> video_streams,
- const void* encoder_settings,
- int base_channel,
- int start_bitrate_bps)
+VideoSendStream::VideoSendStream(
+ newapi::Transport* transport,
+ CpuOveruseObserver* overuse_observer,
+ webrtc::VideoEngine* video_engine,
+ const VideoSendStream::Config& config,
+ const std::vector<VideoStream> video_streams,
+ const void* encoder_settings,
+ const std::map<uint32_t, RtpState>& suspended_ssrcs,
+ int base_channel,
+ int start_bitrate_bps)
: transport_adapter_(transport),
encoded_frame_proxy_(config.post_encode_callback),
config_(config),
start_bitrate_bps_(start_bitrate_bps),
+ suspended_ssrcs_(suspended_ssrcs),
external_codec_(NULL),
channel_(-1),
stats_proxy_(new SendStatisticsProxy(config, this)) {
@@ -403,6 +406,9 @@
uint32_t ssrc = config_.rtp.ssrcs[i];
rtp_rtcp_->SetLocalSSRC(
channel_, ssrc, kViEStreamTypeNormal, static_cast<unsigned char>(i));
+ RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
+ if (it != suspended_ssrcs_.end())
+ rtp_rtcp_->SetRtpStateForSsrc(channel_, ssrc, it->second);
}
if (config_.rtp.rtx.ssrcs.empty()) {
@@ -412,11 +418,15 @@
// Set up RTX.
assert(config_.rtp.rtx.ssrcs.size() == config_.rtp.ssrcs.size());
- for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
+ for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
+ uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
rtp_rtcp_->SetLocalSSRC(channel_,
config_.rtp.rtx.ssrcs[i],
kViEStreamTypeRtx,
static_cast<unsigned char>(i));
+ RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
+ if (it != suspended_ssrcs_.end())
+ rtp_rtcp_->SetRtpStateForSsrc(channel_, ssrc, it->second);
}
if (config_.rtp.rtx.pad_with_redundant_payloads) {
@@ -427,5 +437,20 @@
rtp_rtcp_->SetRtxSendPayloadType(channel_, config_.rtp.rtx.payload_type);
}
+std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const {
+ std::map<uint32_t, RtpState> rtp_states;
+ for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
+ uint32_t ssrc = config_.rtp.ssrcs[i];
+ rtp_states[ssrc] = rtp_rtcp_->GetRtpStateForSsrc(channel_, ssrc);
+ }
+
+ for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
+ uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
+ rtp_states[ssrc] = rtp_rtcp_->GetRtpStateForSsrc(channel_, ssrc);
+ }
+
+ return rtp_states;
+}
+
} // namespace internal
} // namespace webrtc