New flag is_screencast in VideoOptions.

webrtc/media/engine/webrtcvideoengine2.cc

Index: webrtc/media/engine/webrtcvideoengine2.cc
diff --git a/webrtc/media/engine/webrtcvideoengine2.cc b/webrtc/media/engine/webrtcvideoengine2.cc
index c6bdafee5a336666a4fd46f5aab78c0590525105..a3a32fb3504a963f0649ea719d82690dece64d24 100644
--- a/webrtc/media/engine/webrtcvideoengine2.cc
+++ b/webrtc/media/engine/webrtcvideoengine2.cc
@@ -410,9 +410,8 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoStreams(
 }
 
 void* WebRtcVideoChannel2::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
-    const VideoCodec& codec,
-    const VideoOptions& options,
-    bool is_screencast) {
+    const VideoCodec& codec) {
+  bool is_screencast = parameters_.options.is_screencast.value_or(false);
   // No automatic resizing when using simulcast or screencast.
   bool automatic_resize =
       !is_screencast && parameters_.config.rtp.ssrcs.size() == 1;
@@ -423,8 +422,8 @@ void* WebRtcVideoChannel2::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
     denoising = false;
   } else {
     // Use codec default if video_noise_reduction is unset.
-    codec_default_denoising = !options.video_noise_reduction;
-    denoising = options.video_noise_reduction.value_or(false);
+    codec_default_denoising = !parameters_.options.video_noise_reduction;
+    denoising = parameters_.options.video_noise_reduction.value_or(false);
   }
 
   if (CodecNamesEq(codec.name, kH264CodecName)) {
@@ -1528,7 +1527,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
                                          : webrtc::RtcpMode::kCompound;
 
   if (codec_settings) {
-    SetCodecAndOptions(*codec_settings, parameters_.options);
+    SetCodecAndOptions(*codec_settings);
   }
 }
 
@@ -1588,8 +1587,7 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame(
   last_frame_timestamp_ms_ = first_frame_timestamp_ms_ + frame_delta_ms;
   video_frame.set_render_time_ms(last_frame_timestamp_ms_);
   // Reconfigure codec if necessary.
-  SetDimensions(video_frame.width(), video_frame.height(),
-                capturer_->IsScreencast());
+  SetDimensions(video_frame.width(), video_frame.height());
   last_rotation_ = video_frame.rotation();
 
   stream_->Input()->IncomingCapturedFrame(video_frame);
@@ -1668,12 +1666,12 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const {
 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions(
     const VideoOptions& options) {
   rtc::CritScope cs(&lock_);
+  // TODO(nisse): Use .SetAll method?
+  parameters_.options = options;

[pthatcher1, 2016/02/20 08:33:00]

If you don't, then the options that come from SetVideoSend will blow away the
ones that come from SetSendParameters.  In fact, I think that's what's happening
right now, which is bad.  I think that means suspend_below_min_bitrate and
screencast_min_bitrate_kbps are being blown away every time the track is disable
and then re-enable or the track is changed via RtpSender.SetTrack. 

Since VideoOptions is now just 2 things from PeerConnection constraints and 2
things from the VideoSource, is it time to break it up into something more
simple?    

[nisse-webrtc, 2016/02/22 07:52:34]

I'll change to SetAll. I think splitting VideoOptions is best left for a
different cl, ok?

[nisse-webrtc, 2016/02/23 13:13:59]

Done.

   if (parameters_.codec_settings) {
     LOG(LS_INFO) << "SetCodecAndOptions because of SetOptions; options="
                  << options.ToString();
-    SetCodecAndOptions(*parameters_.codec_settings, options);
-  } else {
-    parameters_.options = options;
+    SetCodecAndOptions(*parameters_.codec_settings);
   }
 }
 
@@ -1732,8 +1730,7 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::DestroyVideoEncoder(
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodecAndOptions(
-    const VideoCodecSettings& codec_settings,
-    const VideoOptions& options) {
+    const VideoCodecSettings& codec_settings) {
   parameters_.encoder_config =
       CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
   RTC_DCHECK(!parameters_.encoder_config.streams.empty());
@@ -1765,15 +1762,14 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodecAndOptions(
       HasNack(codec_settings.codec) ? kNackHistoryMs : 0;
 
   parameters_.config.suspend_below_min_bitrate =
-      options.suspend_below_min_bitrate.value_or(false);
+      parameters_.options.suspend_below_min_bitrate.value_or(false);
 
   parameters_.codec_settings =
       rtc::Optional<WebRtcVideoChannel2::VideoCodecSettings>(codec_settings);
-  parameters_.options = options;
 
   LOG(LS_INFO)
       << "RecreateWebRtcStream (send) because of SetCodecAndOptions; options="
-      << options.ToString();
+      << parameters_.options.ToString();
   RecreateWebRtcStream();
   if (allocated_encoder_.encoder != new_encoder.encoder) {
     DestroyVideoEncoder(&allocated_encoder_);
@@ -1810,21 +1806,20 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters(
     parameters_.conference_mode = *params.conference_mode;
   }
   // Set codecs and options.
+  if (params.options)
+    parameters_.options = *params.options;
   if (params.codec) {
-    SetCodecAndOptions(*params.codec,
-                       params.options ? *params.options : parameters_.options);
+    SetCodecAndOptions(*params.codec);
     return;
   } else if (params.options) {
     // Reconfigure if codecs are already set.
     if (parameters_.codec_settings) {
-      SetCodecAndOptions(*parameters_.codec_settings, *params.options);
+      SetCodecAndOptions(*parameters_.codec_settings);
       return;
-    } else {
-      parameters_.options = *params.options;
     }
   }
   else if (params.conference_mode && parameters_.codec_settings) {
-    SetCodecAndOptions(*parameters_.codec_settings, parameters_.options);
+    SetCodecAndOptions(*parameters_.codec_settings);
     return;
   }
   if (recreate_stream) {
@@ -1838,7 +1833,8 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
     const Dimensions& dimensions,
     const VideoCodec& codec) const {
   webrtc::VideoEncoderConfig encoder_config;
-  if (dimensions.is_screencast) {
+  bool is_screencast = parameters_.options.is_screencast.value_or(false);
+  if (is_screencast) {
     encoder_config.min_transmit_bitrate_bps =
         1000 * parameters_.options.screencast_min_bitrate_kbps.value_or(0);
     encoder_config.content_type =
@@ -1852,7 +1848,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
   // Restrict dimensions according to codec max.
   int width = dimensions.width;
   int height = dimensions.height;
-  if (!dimensions.is_screencast) {
+  if (!is_screencast) {
     if (codec.width < width)
       width = codec.width;
     if (codec.height < height)
@@ -1867,7 +1863,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
   // number of negotiated ssrcs. But if the codec is blacklisted for simulcast
   // or a screencast, only configure a single stream.
   size_t stream_count = parameters_.config.rtp.ssrcs.size();
-  if (IsCodecBlacklistedForSimulcast(codec.name) || dimensions.is_screencast) {
+  if (IsCodecBlacklistedForSimulcast(codec.name) || is_screencast) {
     stream_count = 1;
   }
 
@@ -1876,7 +1872,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
                          parameters_.max_bitrate_bps, stream_count);
 
   // Conference mode screencast uses 2 temporal layers split at 100kbit.
-  if (parameters_.conference_mode && dimensions.is_screencast &&
+  if (parameters_.conference_mode && is_screencast &&
       encoder_config.streams.size() == 1) {
     ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
 
@@ -1895,20 +1891,16 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
     int width,
-    int height,
-    bool is_screencast) {
+    int height) {
   if (last_dimensions_.width == width && last_dimensions_.height == height &&
-      last_dimensions_.is_screencast == is_screencast &&
       !pending_encoder_reconfiguration_) {
     // Configured using the same parameters, do not reconfigure.
     return;
   }
-  LOG(LS_INFO) << "SetDimensions: " << width << "x" << height
-               << (is_screencast ? " (screencast)" : " (not screencast)");
+  LOG(LS_INFO) << "SetDimensions: " << width << "x" << height;
 
   last_dimensions_.width = width;
   last_dimensions_.height = height;
-  last_dimensions_.is_screencast = is_screencast;
 
   RTC_DCHECK(!parameters_.encoder_config.streams.empty());
 
@@ -1919,7 +1911,7 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
       CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
 
   encoder_config.encoder_specific_settings = ConfigureVideoEncoderSettings(
-      codec_settings.codec, parameters_.options, is_screencast);
+      codec_settings.codec);
 
   bool stream_reconfigured = stream_->ReconfigureVideoEncoder(encoder_config);
 
@@ -2067,11 +2059,12 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() {
   }
 
   RTC_CHECK(parameters_.codec_settings);
+  RTC_DCHECK_EQ((parameters_.encoder_config.content_type ==
+                 webrtc::VideoEncoderConfig::ContentType::kScreen),
+                parameters_.options.is_screencast.value_or(false))
+      << "encoder content type inconsistent with screencast option";
   parameters_.encoder_config.encoder_specific_settings =
-      ConfigureVideoEncoderSettings(
-          parameters_.codec_settings->codec, parameters_.options,
-          parameters_.encoder_config.content_type ==
-              webrtc::VideoEncoderConfig::ContentType::kScreen);
+      ConfigureVideoEncoderSettings(parameters_.codec_settings->codec);
 
   webrtc::VideoSendStream::Config config = parameters_.config;
   if (!config.rtp.rtx.ssrcs.empty() && config.rtp.rtx.payload_type == -1) {