(Auto)update libjingle 64709629-> 64813990
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5897 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/webrtcvideoengine.cc b/talk/media/webrtc/webrtcvideoengine.cc
index 9ed49df..42d416c 100644
--- a/talk/media/webrtc/webrtcvideoengine.cc
+++ b/talk/media/webrtc/webrtcvideoengine.cc
@@ -238,8 +238,8 @@
class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
public:
- explicit WebRtcRenderAdapter(VideoRenderer* renderer)
- : renderer_(renderer), width_(0), height_(0) {
+ WebRtcRenderAdapter(VideoRenderer* renderer, int channel_id)
+ : renderer_(renderer), channel_id_(channel_id), width_(0), height_(0) {
}
virtual ~WebRtcRenderAdapter() {
@@ -256,7 +256,8 @@
if (width_ > 0 && height_ > 0 && renderer_ != NULL) {
if (!renderer_->SetSize(width_, height_, 0)) {
LOG(LS_ERROR)
- << "WebRtcRenderAdapter SetRenderer failed to SetSize to: "
+ << "WebRtcRenderAdapter (channel " << channel_id_
+ << ") SetRenderer failed to SetSize to: "
<< width_ << "x" << height_;
}
}
@@ -268,10 +269,12 @@
talk_base::CritScope cs(&crit_);
width_ = width;
height_ = height;
- LOG(LS_INFO) << "WebRtcRenderAdapter frame size changed to: "
+ LOG(LS_INFO) << "WebRtcRenderAdapter (channel " << channel_id_
+ << ") frame size changed to: "
<< width << "x" << height;
if (renderer_ == NULL) {
- LOG(LS_VERBOSE) << "WebRtcRenderAdapter the renderer has not been set. "
+ LOG(LS_VERBOSE) << "WebRtcRenderAdapter (channel " << channel_id_
+ << ") the renderer has not been set. "
<< "SetSize will be called later in SetRenderer.";
return 0;
}
@@ -313,7 +316,8 @@
// Sanity check on decoded frame size.
if (buffer_size != static_cast<int>(VideoFrame::SizeOf(width_, height_))) {
- LOG(LS_WARNING) << "WebRtcRenderAdapter received a strange frame size: "
+ LOG(LS_WARNING) << "WebRtcRenderAdapter (channel " << channel_id_
+ << ") received a strange frame size: "
<< buffer_size;
}
@@ -351,6 +355,7 @@
private:
talk_base::CriticalSection crit_;
VideoRenderer* renderer_;
+ int channel_id_;
unsigned int width_;
unsigned int height_;
talk_base::RateTracker frame_rate_tracker_;
@@ -539,7 +544,7 @@
typedef std::map<int, webrtc::VideoDecoder*> DecoderMap; // key: payload type
explicit WebRtcVideoChannelRecvInfo(int channel_id)
: channel_id_(channel_id),
- render_adapter_(NULL),
+ render_adapter_(NULL, channel_id),
decoder_observer_(channel_id) {
}
int channel_id() { return channel_id_; }
@@ -3008,6 +3013,8 @@
adjusted_min_bitrate || start_bitrate_changed);
+ LOG(LS_INFO) << "Reset send codec needed is enabled? "
+ << reset_send_codec_needed;
if (reset_send_codec_needed) {
// On success, SetSendCodec() will reset send_max_bitrate_ to
// expected_bitrate.
@@ -3023,7 +3030,7 @@
if (leaky_bucket_changed) {
bool enable_leaky_bucket =
options_.video_leaky_bucket.GetWithDefaultIfUnset(false);
- LOG(LS_INFO) << "Leaky bucket is enabled : " << enable_leaky_bucket;
+ LOG(LS_INFO) << "Leaky bucket is enabled? " << enable_leaky_bucket;
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
if (engine()->vie()->rtp()->SetTransmissionSmoothingStatus(
@@ -3037,6 +3044,7 @@
int buffer_latency =
options_.buffered_mode_latency.GetWithDefaultIfUnset(
cricket::kBufferedModeDisabled);
+ LOG(LS_INFO) << "Buffer latency is " << buffer_latency;
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
if (engine()->vie()->rtp()->SetSenderBufferingMode(
@@ -3057,6 +3065,8 @@
if (cpu_overuse_detection_changed) {
bool cpu_overuse_detection =
options_.cpu_overuse_detection.GetWithDefaultIfUnset(false);
+ LOG(LS_INFO) << "CPU overuse detection is enabled? "
+ << cpu_overuse_detection;
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
@@ -3067,12 +3077,14 @@
talk_base::DiffServCodePoint dscp = talk_base::DSCP_DEFAULT;
if (options_.dscp.GetWithDefaultIfUnset(false))
dscp = kVideoDscpValue;
+ LOG(LS_INFO) << "DSCP is " << dscp;
if (MediaChannel::SetDscp(dscp) != 0) {
LOG(LS_WARNING) << "Failed to set DSCP settings for video channel";
}
}
if (suspend_below_min_bitrate_changed) {
if (options_.suspend_below_min_bitrate.GetWithDefaultIfUnset(false)) {
+ LOG(LS_INFO) << "Suspend below min bitrate enabled.";
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
engine()->vie()->codec()->SuspendBelowMinBitrate(
@@ -3084,6 +3096,7 @@
}
#ifdef USE_WEBRTC_DEV_BRANCH
if (improved_wifi_bwe_changed) {
+ LOG(LS_INFO) << "Improved WIFI BWE called.";
webrtc::Config config;
config.Set(new webrtc::AimdRemoteRateControl(
options_.use_improved_wifi_bandwidth_estimator