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(Auto)update libjingle 75854833-> 75865376

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7233 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-09-19 02:01:09 +00:00
parent ccb3e3f3db
commit d7acf11e8d
2 changed files with 67 additions and 24 deletions
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78
talk/media/webrtc/webrtcvideoengine.cc
View File

@ -1890,18 +1890,9 @@ bool WebRtcVideoMediaChannel::AddSendStream(const StreamParams& sp) {
}
WebRtcVideoChannelSendInfo* send_channel = GetSendChannelBySsrcKey(ssrc_key);
// Set the send (local) SSRC.
// If there are multiple send SSRCs, we can only set the first one here, and
// the rest of the SSRC(s) need to be set after SetSendCodec has been called
// (with a codec requires multiple SSRC(s)).
if (engine()->vie()->rtp()->SetLocalSSRC(channel_id,
sp.first_ssrc()) != 0) {
LOG_RTCERR2(SetLocalSSRC, channel_id, sp.first_ssrc());
return false;
}
// Set the corresponding RTX SSRC.
if (!SetLocalRtxSsrc(channel_id, sp, sp.first_ssrc(), 0)) {
if (!SetLimitedNumberOfSendSsrcs(channel_id, sp, 1)) {
return false;
}
@ -3673,6 +3664,7 @@ bool WebRtcVideoMediaChannel::SetSendCodec(
LOG(LS_INFO) << "0x0 resolution selected. Captured frames will be dropped "
<< "for ssrc: " << ssrc << ".";
} else {
StreamParams* send_params = send_channel->stream_params();
MaybeChangeBitrates(channel_id, &target_codec);
webrtc::VideoCodec current_codec;
if (!engine()->vie()->codec()->GetSendCodec(channel_id, current_codec)) {
@ -3688,6 +3680,11 @@ bool WebRtcVideoMediaChannel::SetSendCodec(
return false;
}
if (send_params) {
if (!SetSendSsrcs(channel_id, *send_params, target_codec)) {
return false;
}
}
// NOTE: SetRtxSendPayloadType must be called after all simulcast SSRCs
// are configured. Otherwise ssrc's configured after this point will use
// the primary PT for RTX.
@ -3976,6 +3973,13 @@ bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id,
leaky_bucket);
}
// TODO(sriniv): SetSendCodec already sets ssrc's like below.
// Consider removing.
if (send_params) {
if (!SetSendSsrcs(channel_id, *send_params, target_codec)) {
return false;
}
}
if (reset) {
*reset = true;
}
@ -4131,21 +4135,55 @@ bool WebRtcVideoMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter,
return SetHeaderExtension(setter, channel_id, extension);
}
bool WebRtcVideoMediaChannel::SetLocalRtxSsrc(int channel_id,
const StreamParams& send_params,
uint32 primary_ssrc,
int stream_idx) {
uint32 rtx_ssrc = 0;
bool has_rtx = send_params.GetFidSsrc(primary_ssrc, &rtx_ssrc);
if (has_rtx && engine()->vie()->rtp()->SetLocalSSRC(
channel_id, rtx_ssrc, webrtc::kViEStreamTypeRtx, stream_idx) != 0) {
LOG_RTCERR4(SetLocalSSRC, channel_id, rtx_ssrc,
webrtc::kViEStreamTypeRtx, stream_idx);
bool WebRtcVideoMediaChannel::SetPrimaryAndRtxSsrcs(
int channel_id, int idx, uint32 primary_ssrc,
const StreamParams& send_params) {
LOG(LS_INFO) << "Set primary ssrc " << primary_ssrc
<< " on channel " << channel_id << " idx " << idx;
if (engine()->vie()->rtp()->SetLocalSSRC(
channel_id, primary_ssrc, webrtc::kViEStreamTypeNormal, idx) != 0) {
LOG_RTCERR4(SetLocalSSRC,
channel_id, primary_ssrc, webrtc::kViEStreamTypeNormal, idx);
return false;
}
uint32 rtx_ssrc = 0;
if (send_params.GetFidSsrc(primary_ssrc, &rtx_ssrc)) {
LOG(LS_INFO) << "Set rtx ssrc " << rtx_ssrc
<< " on channel " << channel_id << " idx " << idx;
if (engine()->vie()->rtp()->SetLocalSSRC(
channel_id, rtx_ssrc, webrtc::kViEStreamTypeRtx, idx) != 0) {
LOG_RTCERR4(SetLocalSSRC,
channel_id, rtx_ssrc, webrtc::kViEStreamTypeRtx, idx);
return false;
}
}
return true;
}
bool WebRtcVideoMediaChannel::SetLimitedNumberOfSendSsrcs(
int channel_id, const StreamParams& sp, size_t limit) {
const SsrcGroup* sim_group = sp.get_ssrc_group(kSimSsrcGroupSemantics);
if (!sim_group || limit == 1) {
return SetPrimaryAndRtxSsrcs(channel_id, 0, sp.first_ssrc(), sp);
}
std::vector<uint32> ssrcs = sim_group->ssrcs;
for (size_t i = 0; i < ssrcs.size() && i < limit; ++i) {
if (!SetPrimaryAndRtxSsrcs(channel_id, i, ssrcs[i], sp)) {
return false;
}
}
return true;
}
bool WebRtcVideoMediaChannel::SetSendSsrcs(
int channel_id, const StreamParams& sp,
const webrtc::VideoCodec& codec) {
// TODO(pthatcher): Support more than one primary SSRC per stream.
return SetLimitedNumberOfSendSsrcs(channel_id, sp, 1);
}
void WebRtcVideoMediaChannel::MaybeConnectCapturer(VideoCapturer* capturer) {
if (capturer && GetSendChannelNum(capturer) == 1) {
capturer->SignalVideoFrame.connect(this,

13
talk/media/webrtc/webrtcvideoengine.h
View File

@ -310,6 +310,15 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler,
virtual int SendPacket(int channel, const void* data, int len);
virtual int SendRTCPPacket(int channel, const void* data, int len);
bool SetPrimaryAndRtxSsrcs(
int channel_id, int idx, uint32 primary_ssrc,
const StreamParams& send_params);
bool SetLimitedNumberOfSendSsrcs(
int channel_id, const StreamParams& send_params, size_t limit);
virtual bool SetSendSsrcs(
int channel_id, const StreamParams& send_params,
const webrtc::VideoCodec& codec);
private:
typedef std::map<uint32, WebRtcVideoChannelRecvInfo*> RecvChannelMap;
typedef std::map<uint32, WebRtcVideoChannelSendInfo*> SendChannelMap;
@ -413,10 +422,6 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler,
// Signal when cpu adaptation has no further scope to adapt.
void OnCpuAdaptationUnable();
// Set the local (send-side) RTX SSRC corresponding to primary_ssrc.
bool SetLocalRtxSsrc(int channel_id, const StreamParams& send_params,
uint32 primary_ssrc, int stream_idx);
// Connect |capturer| to WebRtcVideoMediaChannel if it is only registered
// to one send channel, i.e. the first send channel.
void MaybeConnectCapturer(VideoCapturer* capturer);