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(Auto)update libjingle 68689052-> 68689059

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6350 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-06-06 10:54:08 +00:00
parent 6ae48c6609
commit 7b6cbb3aa0
3 changed files with 70 additions and 19 deletions
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46
talk/media/webrtc/webrtcvideoengine.cc
View File

@ -2050,6 +2050,9 @@ bool WebRtcVideoMediaChannel::AddRecvStream(const StreamParams& sp) {
<< " reuse default channel #"
<< vie_channel_;
first_receive_ssrc_ = sp.first_ssrc();
if (!MaybeSetRtxSsrc(sp, vie_channel_)) {
return false;
}
if (render_started_) {
if (engine()->vie()->render()->StartRender(vie_channel_) !=0) {
LOG_RTCERR1(StartRender, vie_channel_);
@ -2088,19 +2091,8 @@ bool WebRtcVideoMediaChannel::AddRecvStream(const StreamParams& sp) {
}
channel_iterator = recv_channels_.find(sp.first_ssrc());
// Set the corresponding RTX SSRC.
uint32 rtx_ssrc;
bool has_rtx = sp.GetFidSsrc(sp.first_ssrc(), &rtx_ssrc);
if (has_rtx) {
LOG(LS_INFO) << "Setting rtx ssrc " << rtx_ssrc << " for stream "
<< sp.first_ssrc();
if (engine()->vie()->rtp()->SetRemoteSSRCType(
channel_id, webrtc::kViEStreamTypeRtx, rtx_ssrc) != 0) {
LOG_RTCERR3(SetRemoteSSRCType, channel_id, webrtc::kViEStreamTypeRtx,
rtx_ssrc);
return false;
}
rtx_to_primary_ssrc_[rtx_ssrc] = sp.first_ssrc();
if (!MaybeSetRtxSsrc(sp, channel_id)) {
return false;
}
// Get the default renderer.
@ -2129,6 +2121,24 @@ bool WebRtcVideoMediaChannel::AddRecvStream(const StreamParams& sp) {
return true;
}
bool WebRtcVideoMediaChannel::MaybeSetRtxSsrc(const StreamParams& sp,
int channel_id) {
uint32 rtx_ssrc;
bool has_rtx = sp.GetFidSsrc(sp.first_ssrc(), &rtx_ssrc);
if (has_rtx) {
LOG(LS_INFO) << "Setting rtx ssrc " << rtx_ssrc << " for stream "
<< sp.first_ssrc();
if (engine()->vie()->rtp()->SetRemoteSSRCType(
channel_id, webrtc::kViEStreamTypeRtx, rtx_ssrc) != 0) {
LOG_RTCERR3(SetRemoteSSRCType, channel_id, webrtc::kViEStreamTypeRtx,
rtx_ssrc);
return false;
}
rtx_to_primary_ssrc_[rtx_ssrc] = sp.first_ssrc();
}
return true;
}
bool WebRtcVideoMediaChannel::RemoveRecvStream(uint32 ssrc) {
if (ssrc == 0) {
LOG(LS_ERROR) << "RemoveRecvStream with 0 ssrc is not supported.";
@ -3974,9 +3984,13 @@ int WebRtcVideoMediaChannel::GetRecvChannelNum(uint32 ssrc) {
// Check if we have an RTX stream registered on this SSRC.
SsrcMap::iterator rtx_it = rtx_to_primary_ssrc_.find(ssrc);
if (rtx_it != rtx_to_primary_ssrc_.end()) {
it = recv_channels_.find(rtx_it->second);
assert(it != recv_channels_.end());
recv_channel = it->second->channel_id();
if (rtx_it->second == first_receive_ssrc_) {
recv_channel = vie_channel_;
} else {
it = recv_channels_.find(rtx_it->second);
assert(it != recv_channels_.end());
recv_channel = it->second->channel_id();
}
}
} else {
recv_channel = it->second->channel_id();

1
talk/media/webrtc/webrtcvideoengine.h
View File

@ -350,6 +350,7 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler,
bool SetReceiveCodecs(WebRtcVideoChannelRecvInfo* info);
// Returns the channel number that receives the stream with SSRC |ssrc|.
int GetRecvChannelNum(uint32 ssrc);
bool MaybeSetRtxSsrc(const StreamParams& sp, int channel_id);
// Given captured video frame size, checks if we need to reset vie send codec.
// |reset| is set to whether resetting has happened on vie or not.
// Returns false on error.

42
talk/media/webrtc/webrtcvideoengine_unittest.cc
View File

@ -74,7 +74,7 @@ static const unsigned int kMaxBandwidthKbps = 2000;
static const uint32 kSsrcs1[] = {1};
static const uint32 kSsrcs2[] = {1, 2};
static const uint32 kSsrcs3[] = {1, 2, 3};
static const uint32 kRtxSsrc1[] = {4};
static const uint32 kRtxSsrcs1[] = {4};
static const uint32 kRtxSsrcs3[] = {4, 5, 6};
@ -750,8 +750,44 @@ TEST_F(WebRtcVideoEngineTestFake, SetRecvCodecsWithRtx) {
EXPECT_EQ(rtx_codec.id, vie_.GetRtxRecvPayloadType(channel_num));
}
// Test that RTX packets are routed to the default video channel if
// there's only one recv stream.
TEST_F(WebRtcVideoEngineTestFake, TestReceiveRtxOneStream) {
EXPECT_TRUE(SetupEngine());
// Setup one channel with an associated RTX stream.
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]);
EXPECT_TRUE(channel_->AddRecvStream(params));
int channel_num = vie_.GetLastChannel();
EXPECT_EQ(static_cast<int>(kRtxSsrcs1[0]),
vie_.GetRemoteRtxSsrc(channel_num));
// Register codecs.
std::vector<cricket::VideoCodec> codec_list;
codec_list.push_back(kVP8Codec720p);
cricket::VideoCodec rtx_codec(96, "rtx", 0, 0, 0, 0);
rtx_codec.SetParam("apt", kVP8Codec.id);
codec_list.push_back(rtx_codec);
EXPECT_TRUE(channel_->SetRecvCodecs(codec_list));
// Construct a fake RTX packet and verify that it is passed to the
// right WebRTC channel.
const size_t kDataLength = 12;
uint8_t data[kDataLength];
memset(data, 0, sizeof(data));
data[0] = 0x80;
data[1] = rtx_codec.id;
talk_base::SetBE32(&data[8], kRtxSsrcs1[0]);
talk_base::Buffer packet(data, kDataLength);
talk_base::PacketTime packet_time;
channel_->OnPacketReceived(&packet, packet_time);
EXPECT_EQ(rtx_codec.id, vie_.GetLastRecvdPayloadType(channel_num));
}
// Test that RTX packets are routed to the correct video channel.
TEST_F(WebRtcVideoEngineTestFake, TestReceiveRtx) {
TEST_F(WebRtcVideoEngineTestFake, TestReceiveRtxThreeStreams) {
EXPECT_TRUE(SetupEngine());
// Setup three channels with associated RTX streams.
@ -881,7 +917,7 @@ TEST_F(WebRtcVideoEngineTestFake, RecvStreamWithRtx) {
EXPECT_TRUE(channel_->AddRecvStream(
cricket::CreateSimWithRtxStreamParams("cname",
MAKE_VECTOR(kSsrcs1),
MAKE_VECTOR(kRtxSsrc1))));
MAKE_VECTOR(kRtxSsrcs1))));
int new_channel_num = vie_.GetLastChannel();
EXPECT_NE(default_channel, new_channel_num);
EXPECT_EQ(4, vie_.GetRemoteRtxSsrc(new_channel_num));