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(Auto)update libjingle 79200114-> 79205306

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7627 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-11-05 00:14:02 +00:00
parent 8125744a5f
commit 795d003770
1 changed files with 55 additions and 54 deletions
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109
talk/media/webrtc/webrtcvideoengine.cc
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@ -1987,14 +1987,21 @@ bool WebRtcVideoMediaChannel::AddSendStream(const StreamParams& sp) {
SetReceiverReportSsrc(sp.first_ssrc());
}
send_channel->set_stream_params(sp);
// Reset send codec after stream parameters changed.
if (send_codec_) {
if (!SetSendCodec(send_channel, *send_codec_)) {
send_channel->SetAdaptFormat(
VideoFormatFromVieCodec(*send_codec_),
WebRtcVideoChannelSendInfo::kAdaptFormatTypeCodec);
VideoSendParams send_params;
send_params.codec = *send_codec_;
send_params.stream = sp;
if (!SetSendParams(send_channel, send_params)) {
return false;
}
LogSendCodecChange("SetSendStreamFormat()");
LogSendCodecChange("AddStream()");
} else {
// Save the stream params for later, when we have a codec.
send_channel->set_stream_params(sp);
}
if (sending_) {
@ -2990,39 +2997,40 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
VideoOptions original = options_;
options_.SetAll(options);
// Set CPU options for all send channels.
// Set CPU options and codec options for all send channels.
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
send_channel->ApplyCpuOptions(options_);
if (send_codec_) {
VideoSendParams send_params = send_channel->send_params();
bool conference_mode_turned_off = (
original.conference_mode.IsSet() &&
options.conference_mode.IsSet() &&
original.conference_mode.GetWithDefaultIfUnset(false) &&
!options.conference_mode.GetWithDefaultIfUnset(false));
if (conference_mode_turned_off) {
// This is a special case for turning conference mode off.
// Max bitrate should go back to the default maximum value instead
// of the current maximum.
send_params.codec.maxBitrate = kAutoBandwidth;
}
// TODO(pthatcher): Remove this. We don't need 4 ways to set bitrates.
int new_start_bitrate;
if (options.video_start_bitrate.Get(&new_start_bitrate)) {
send_params.codec.startBitrate = new_start_bitrate;
}
if (!SetSendParams(send_channel, send_params)) {
return false;
}
LogSendCodecChange("SetOptions()");
}
}
if (send_codec_) {
webrtc::VideoCodec new_codec = *send_codec_;
bool conference_mode_turned_off = (
original.conference_mode.IsSet() &&
options.conference_mode.IsSet() &&
original.conference_mode.GetWithDefaultIfUnset(false) &&
!options.conference_mode.GetWithDefaultIfUnset(false));
if (conference_mode_turned_off) {
// This is a special case for turning conference mode off.
// Max bitrate should go back to the default maximum value instead
// of the current maximum.
new_codec.maxBitrate = kAutoBandwidth;
}
// TODO(pthatcher): Remove this. We don't need 4 ways to set bitrates.
int new_start_bitrate;
if (options.video_start_bitrate.Get(&new_start_bitrate)) {
new_codec.startBitrate = new_start_bitrate;
}
if (!SetSendCodec(new_codec)) {
return false;
}
LogSendCodecChange("SetOptions()");
}
int buffer_latency;
if (Changed(options.buffered_mode_latency,
@ -3676,26 +3684,9 @@ bool WebRtcVideoMediaChannel::SetSendCodec(
VideoFormatFromVieCodec(codec),
WebRtcVideoChannelSendInfo::kAdaptFormatTypeCodec);
MaybeRegisterExternalEncoder(send_channel, codec);
VideoSendParams send_params = send_channel->send_params();
send_params.codec = codec;
if (!SetSendParams(send_channel, send_params)) {
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.
const int channel_id = send_channel->channel_id();
if (send_rtx_type_ != -1 &&
engine()->vie()->rtp()->SetRtxSendPayloadType(channel_id,
send_rtx_type_) != 0) {
LOG_RTCERR2(SetRtxSendPayloadType, channel_id, send_rtx_type_);
return false;
}
return true;
return SetSendParams(send_channel, send_params);
}
static std::string ToString(webrtc::VideoCodecComplexity complexity) {
@ -3872,6 +3863,8 @@ bool WebRtcVideoMediaChannel::SetSendParams(
const VideoSendParams& send_params) {
const int channel_id = send_channel->channel_id();
MaybeRegisterExternalEncoder(send_channel, send_params.codec);
CapturedFrameInfo frame;
send_channel->last_captured_frame_info().Get(&frame);
@ -3923,10 +3916,18 @@ bool WebRtcVideoMediaChannel::SetSendParams(
}
engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id, true);
if (send_channel->IsActive()) {
if (!SetSendSsrcs(channel_id, send_params.stream, codec)) {
return false;
}
if (!SetSendSsrcs(channel_id, send_params.stream, codec)) {
return false;
}
// NOTE: SetRtxSendPayloadType must be called after all SSRCs are
// configured. Otherwise ssrc's configured after this point will use
// the primary PT for RTX.
if (send_rtx_type_ != -1 &&
engine()->vie()->rtp()->SetRtxSendPayloadType(channel_id,
send_rtx_type_) != 0) {
LOG_RTCERR2(SetRtxSendPayloadType, channel_id, send_rtx_type_);
return false;
}
send_channel->set_send_params(send_params);