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Remove TimeToSendPacket and TimeToSendPadding from the default module.

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Thie CL moves the default RTP module logic for TimeToSendPacket and
TimeToSendPadding to PayloadRouter class and asserts on usage of the
default module.

BUG=769
TEST=New unittest.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33319004

Cr-Commit-Position: refs/heads/master@{#8383}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8383 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org
2015-02-17 10:15:06 +00:00
parent c0fc4dd87c
commit 290cb56dca
7 changed files with 216 additions and 52 deletions
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23
webrtc/video_engine/payload_router.cc
View File

@ -71,6 +71,29 @@ bool PayloadRouter::RoutePayload(FrameType frame_type,
payload_length, fragmentation, rtp_video_hdr) == 0 ? true : false;
}
bool PayloadRouter::TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_timestamp,
bool retransmission) {
CriticalSectionScoped cs(crit_.get());
for (auto* rtp_module : rtp_modules_) {
if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) {
return rtp_module->TimeToSendPacket(ssrc, sequence_number,
capture_timestamp, retransmission);
}
}
return true;
}
size_t PayloadRouter::TimeToSendPadding(size_t bytes) {
CriticalSectionScoped cs(crit_.get());
for(auto* rtp_module : rtp_modules_) {
if (rtp_module->SendingMedia())
return rtp_module->TimeToSendPadding(bytes);
}
return 0;
}
size_t PayloadRouter::MaxPayloadLength() const {
size_t min_payload_length = DefaultMaxPayloadLength();
CriticalSectionScoped cs(crit_.get());