(Auto)update libjingle 63738002-> 63773382

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5787 4adac7df-926f-26a2-2b94-8c16560cd09d

talk/app/webrtc/mediaconstraintsinterface.cc

@@ -77,6 +77,14 @@ const char MediaConstraintsInterface::kCpuUnderuseThreshold[] =
 const char MediaConstraintsInterface::kCpuOveruseThreshold[] =
     "googCpuOveruseThreshold";
 
+// Google-specific PeerConnection constraint keys.
+const char MediaConstraintsInterface::kImprovedWifiBwe[] =
+    "googImprovedWifiBwe";
+const char MediaConstraintsInterface::kScreencastMinBitrate[] =
+    "googScreencastMinBitrate";
+const char MediaConstraintsInterface::kSkipEncodingUnusedStreams[] =
+    "googSkipEncodingUnusedStreams";
+
 // Constraint keys for CreateOffer / CreateAnswer defined in W3C specification.
 const char MediaConstraintsInterface::kOfferToReceiveAudio[] =
     "OfferToReceiveAudio";

talk/app/webrtc/mediaconstraintsinterface.h

@@ -87,12 +87,19 @@ class MediaConstraintsInterface {
   // Google-specific constraint keys for a local video source
   static const char kNoiseReduction[];  // googNoiseReduction
   static const char kLeakyBucket[];  // googLeakyBucket
+
   // googTemporalLayeredScreencast
   static const char kTemporalLayeredScreencast[];
   static const char kCpuOveruseDetection[];
   static const char kCpuUnderuseThreshold[];
   static const char kCpuOveruseThreshold[];
 
+  // Google-specific PeerConnection constraint keys.
+  static const char kImprovedWifiBwe[];  // googImprovedWifiBwe
+  static const char kScreencastMinBitrate[];  // googScreencastMinBitrate
+  static const char kSkipEncodingUnusedStreams[];
+    // googSkipEncodingUnusedStreams
+
   // Constraint keys for CreateOffer / CreateAnswer
   // Specified by the W3C PeerConnection spec
   static const char kOfferToReceiveVideo[];  // OfferToReceiveVideo

talk/app/webrtc/webrtcsession.cc

@@ -517,6 +517,34 @@ bool WebRtcSession::Initialize(
     video_options_.suspend_below_min_bitrate.Set(value);
   }
 
+  if (FindConstraint(
+      constraints,
+      MediaConstraintsInterface::kSkipEncodingUnusedStreams,
+      &value,
+      NULL)) {
+    video_options_.skip_encoding_unused_streams.Set(value);
+  }
+
+  std::string string_value;
+  if (constraints &&
+      constraints->GetOptional().FindFirst(
+        MediaConstraintsInterface::kScreencastMinBitrate,
+        &string_value)) {
+    int bitrate;
+    if (talk_base::FromString(string_value, &bitrate)) {
+      video_options_.screencast_min_bitrate.Set(bitrate);
+    }
+  }
+
+  // Find improved wifi bwe constraint.
+  if (FindConstraint(
+        constraints,
+        MediaConstraintsInterface::kImprovedWifiBwe,
+        &value,
+        NULL)) {
+    video_options_.use_improved_wifi_bandwidth_estimator.Set(value);
+  }
+
   const cricket::VideoCodec default_codec(
       JsepSessionDescription::kDefaultVideoCodecId,
       JsepSessionDescription::kDefaultVideoCodecName,

talk/examples/ios/AppRTCDemo/APPRTCVideoView.h

@@ -1,43 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <UIKit/UIKit.h>
-
-@class RTCVideoTrack;
-
-// This class encapsulates VideoRenderIosView.
-@interface APPRTCVideoView : UIView
-
-// Property to get/set required video orientation.
-@property(nonatomic, assign) UIInterfaceOrientation videoOrientation;
-// Specifies whether the object represents a local or remote video stream.
-@property(nonatomic, assign) BOOL isRemote;
-
-// Sets up the underlying renderer and track objects.
-- (void)renderVideoTrackInterface:(RTCVideoTrack*)track;
-
-@end

talk/examples/ios/AppRTCDemo/APPRTCVideoView.m

@@ -1,82 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-/*
- * This APPRTCVideoView must be initialzed and added to a View to get
- * either the local or remote video stream rendered.
- * It is a view itself and it encapsulates
- * an object of VideoRenderIosView and UIActivityIndicatorView.
- * Both of the views will get resized as per the frame of their parent.
- */
-
-#import "APPRTCVideoView.h"
-
-#import "RTCVideoRenderer.h"
-#import "RTCVideoTrack.h"
-
-@interface APPRTCVideoView () {
-  RTCVideoTrack* _track;
-  RTCVideoRenderer* _renderer;
-}
-
-@property(nonatomic, weak) UIView* renderView;
-@property(nonatomic, weak) UIActivityIndicatorView* activityView;
-
-@end
-
-@implementation APPRTCVideoView
-
-@synthesize videoOrientation = _videoOrientation;
-
-- (void)layoutSubviews {
-  [super layoutSubviews];
-  if (!_renderer) {
-    // Left-right (mirror) flip the remote view.
-    CGAffineTransform xform =
-        CGAffineTransformMakeScale(self.isRemote ? -1 : 1, 1);
-    // TODO(fischman): why is this rotate (vertical+horizontal flip) needed?!?
-    xform = CGAffineTransformRotate(xform, M_PI);
-    // TODO(fischman): ensure back-camera flip is correct in all orientations,
-    // when back-camera support is added.
-    [self setTransform:xform];
-    _renderer = [[RTCVideoRenderer alloc] initWithView:self];
-  }
-}
-
-- (void)renderVideoTrackInterface:(RTCVideoTrack*)videoTrack {
-  [_track removeRenderer:_renderer];
-  [_renderer stop];
-
-  _track = videoTrack;
-
-  if (_track) {
-    [_track addRenderer:_renderer];
-    [_renderer start];
-  }
-}
-
-@end

talk/media/base/mediachannel.h

@@ -329,6 +329,10 @@ struct VideoOptions {
     suspend_below_min_bitrate.SetFrom(change.suspend_below_min_bitrate);
     unsignalled_recv_stream_limit.SetFrom(change.unsignalled_recv_stream_limit);
     use_simulcast_adapter.SetFrom(change.use_simulcast_adapter);
+    skip_encoding_unused_streams.SetFrom(change.skip_encoding_unused_streams);
+    screencast_min_bitrate.SetFrom(change.screencast_min_bitrate);
+    use_improved_wifi_bandwidth_estimator.SetFrom(
+        change.use_improved_wifi_bandwidth_estimator);
   }
 
   bool operator==(const VideoOptions& o) const {
@@ -359,7 +363,11 @@ struct VideoOptions {
         dscp == o.dscp &&
         suspend_below_min_bitrate == o.suspend_below_min_bitrate &&
         unsignalled_recv_stream_limit == o.unsignalled_recv_stream_limit &&
-        use_simulcast_adapter == o.use_simulcast_adapter;
+        use_simulcast_adapter == o.use_simulcast_adapter &&
+        skip_encoding_unused_streams == o.skip_encoding_unused_streams &&
+        screencast_min_bitrate == o.screencast_min_bitrate;
+        use_improved_wifi_bandwidth_estimator ==
+            o.use_improved_wifi_bandwidth_estimator;
   }
 
   std::string ToString() const {
@@ -395,6 +403,11 @@ struct VideoOptions {
     ost << ToStringIfSet("num channels for early receive",
                          unsignalled_recv_stream_limit);
     ost << ToStringIfSet("use simulcast adapter", use_simulcast_adapter);
+    ost << ToStringIfSet("skip encoding unused streams",
+                         skip_encoding_unused_streams);
+    ost << ToStringIfSet("screencast min bitrate", screencast_min_bitrate);
+    ost << ToStringIfSet("improved wifi bwe",
+                         use_improved_wifi_bandwidth_estimator);
     ost << "}";
     return ost.str();
   }
@@ -454,6 +467,13 @@ struct VideoOptions {
   Settable<int> unsignalled_recv_stream_limit;
   // Enable use of simulcast adapter.
   Settable<bool> use_simulcast_adapter;
+  // Enables the encoder to skip encoding stream not actually sent due to too
+  // low available bit rate.
+  Settable<bool> skip_encoding_unused_streams;
+  // Force screencast to use a minimum bitrate
+  Settable<int> screencast_min_bitrate;
+  // Enable improved bandwidth estiamtor on wifi.
+  Settable<bool> use_improved_wifi_bandwidth_estimator;
 };
 
 // A class for playing out soundclips.

talk/media/webrtc/fakewebrtcvideoengine.h

@@ -244,10 +244,6 @@ class FakeWebRtcVideoEncoderFactory : public WebRtcVideoEncoderFactory {
       (*it)->OnCodecsAvailable();
   }
 
-  int GetNumCreatedEncoders() {
-    return num_created_encoders_;
-  }
-
   const std::vector<FakeWebRtcVideoEncoder*>& encoders() {
     return encoders_;
   }

talk/media/webrtc/webrtcvideoengine.cc

@@ -2837,6 +2837,13 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
     conference_mode_turned_off = true;
   }
 
+#ifdef USE_WEBRTC_DEV_BRANCH
+  bool improved_wifi_bwe_changed =
+      options.use_improved_wifi_bandwidth_estimator.IsSet() &&
+      options_.use_improved_wifi_bandwidth_estimator !=
+          options.use_improved_wifi_bandwidth_estimator;
+
+#endif
 
   // Save the options, to be interpreted where appropriate.
   // Use options_.SetAll() instead of assignment so that unset value in options
@@ -2961,6 +2968,19 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
       LOG(LS_WARNING) << "Cannot disable video suspension once it is enabled";
     }
   }
+#ifdef USE_WEBRTC_DEV_BRANCH
+  if (improved_wifi_bwe_changed) {
+    webrtc::Config config;
+    config.Set(new webrtc::AimdRemoteRateControl(
+        options_.use_improved_wifi_bandwidth_estimator
+          .GetWithDefaultIfUnset(false)));
+    for (SendChannelMap::iterator it = send_channels_.begin();
+            it != send_channels_.end(); ++it) {
+      engine()->vie()->network()->SetBandwidthEstimationConfig(
+          it->second->channel_id(), config);
+    }
+  }
+#endif
   return true;
 }
 
@@ -3753,6 +3773,11 @@ bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
   // Disable denoising for screencasting.
   bool enable_denoising =
       options_.video_noise_reduction.GetWithDefaultIfUnset(false);
+#ifdef USE_WEBRTC_DEV_BRANCH
+  int screencast_min_bitrate =
+      options_.screencast_min_bitrate.GetWithDefaultIfUnset(0);
+  bool leaky_bucket = options_.video_leaky_bucket.GetWithDefaultIfUnset(false);
+#endif
   bool denoising = !is_screencast && enable_denoising;
   bool reset_send_codec =
       target_width != cur_width || target_height != cur_height ||
@@ -3786,6 +3811,24 @@ bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
       LOG_RTCERR1(SetSendCodec, channel_id);
       return false;
     }
+
+#ifdef USE_WEBRTC_DEV_BRANCH
+    if (is_screencast) {
+      engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id,
+                                                    screencast_min_bitrate);
+      // If screencast and min bitrate set, force enable pacer.
+      if (screencast_min_bitrate > 0) {
+        engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id,
+                                                               true);
+      }
+    } else {
+      // In case of switching from screencast to regular capture, set
+      // min bitrate padding and pacer back to defaults.
+      engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id, 0);
+      engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id,
+                                                             leaky_bucket);
+    }
+#endif
     if (reset) {
       *reset = true;
     }

talk/media/webrtc/webrtcvideoengine_unittest.cc

@@ -1662,11 +1662,9 @@ TEST_F(WebRtcVideoEngineTestFake, DontRegisterEncoderMultipleTimes) {
 
   EXPECT_TRUE(vie_.ExternalEncoderRegistered(channel_num, 100));
   EXPECT_EQ(1, vie_.GetNumExternalEncoderRegistered(channel_num));
-  EXPECT_EQ(1, encoder_factory_.GetNumCreatedEncoders());
 
   EXPECT_TRUE(channel_->SetSendCodecs(codecs));
   EXPECT_EQ(1, vie_.GetNumExternalEncoderRegistered(channel_num));
-  EXPECT_EQ(1, encoder_factory_.GetNumCreatedEncoders());
 
   // Remove stream previously added to free the external encoder instance.
   EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
@@ -1798,7 +1796,6 @@ TEST_F(WebRtcVideoEngineTestFake, UpdateEncoderCodecsAfterSetFactory) {
 
   EXPECT_TRUE(vie_.ExternalEncoderRegistered(channel_num, 100));
   EXPECT_EQ(1, vie_.GetNumExternalEncoderRegistered(channel_num));
-  EXPECT_EQ(1, encoder_factory_.GetNumCreatedEncoders());
 
   // Remove stream previously added to free the external encoder instance.
   EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));