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author | mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2014-07-24 16:41:25 +0000 |
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committer | mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2014-07-24 16:41:25 +0000 |
commit | 4a1b3e3a69d349b0d3e91f607f24e02d8b975688 (patch) | |
tree | da30959f40b1c31a881a2f5df8734c8e65432c32 /modules | |
parent | a4dc1ae57dff218db6e18471eee6dec2108189fe (diff) | |
download | webrtc-4a1b3e3a69d349b0d3e91f607f24e02d8b975688.tar.gz |
Make sure padding is sent on the first sending RTP module.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6774 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'modules')
-rw-r--r-- | modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 8 | ||||
-rw-r--r-- | modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc | 6 |
2 files changed, 5 insertions, 9 deletions
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc index a26b36a5..1a3b79cb 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc @@ -613,14 +613,10 @@ int ModuleRtpRtcpImpl::TimeToSendPadding(int bytes) { } } else { CriticalSectionScoped lock(critical_section_module_ptrs_.get()); - // Decide what media stream to pad on based on a round-robin scheme. for (size_t i = 0; i < child_modules_.size(); ++i) { - padding_index_ = (padding_index_ + 1) % child_modules_.size(); // Send padding on one of the modules sending media. - if (child_modules_[padding_index_]->SendingMedia() && - child_modules_[padding_index_]->rtp_sender_.GetTargetBitrate() > 0) { - return child_modules_[padding_index_]->rtp_sender_.TimeToSendPadding( - bytes); + if (child_modules_[i]->SendingMedia()) { + return child_modules_[i]->rtp_sender_.TimeToSendPadding(bytes); } } } diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc index e4507c34..3101a1fe 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc @@ -388,7 +388,7 @@ class RtpSendingTest : public ::testing::Test { VideoCodec codec_; }; -TEST_F(RtpSendingTest, RoundRobinPadding) { +TEST_F(RtpSendingTest, DISABLED_RoundRobinPadding) { // We have to send on an SSRC to be allowed to pad, since a marker bit must // be sent prior to padding packets. const uint8_t payload[200] = {0}; @@ -410,7 +410,7 @@ TEST_F(RtpSendingTest, RoundRobinPadding) { ExpectPadding(expected_padding); } -TEST_F(RtpSendingTest, RoundRobinPaddingRtx) { +TEST_F(RtpSendingTest, DISABLED_RoundRobinPaddingRtx) { // Enable RTX to allow padding to be sent prior to media. for (int i = 1; i < codec_.numberOfSimulcastStreams + 1; ++i) { // Abs-send-time is needed to be allowed to send padding prior to media, @@ -440,7 +440,7 @@ TEST_F(RtpSendingTest, RoundRobinPaddingRtx) { ExpectPadding(expected_padding); } -TEST_F(RtpSendingTest, RoundRobinPaddingRtxRedundantPayloads) { +TEST_F(RtpSendingTest, DISABLED_RoundRobinPaddingRtxRedundantPayloads) { for (int i = 1; i < codec_.numberOfSimulcastStreams + 1; ++i) { senders_[i]->SetRtxSendPayloadType(96); senders_[i]->SetRtxSsrc(kSenderRtxSsrc + i); |