(Auto)update libjingle 79200114-> 79205306
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7627 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/webrtcvideoengine.cc b/talk/media/webrtc/webrtcvideoengine.cc
index 04092f3..d0ed46d 100644
--- a/talk/media/webrtc/webrtcvideoengine.cc
+++ b/talk/media/webrtc/webrtcvideoengine.cc
@@ -1987,14 +1987,21 @@
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 @@
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 @@
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 @@
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 @@
}
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);
