New flag is_screencast in VideoOptions.

webrtc/media/engine/webrtcvideoengine2.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 392 matching lines @@
 
   int max_qp = kDefaultQpMax;
   codec.GetParam(kCodecParamMaxQuantization, &max_qp);
   stream.max_qp = max_qp;
   std::vector<webrtc::VideoStream> streams;
   streams.push_back(stream);
   return streams;
 }
 
 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;
   bool frame_dropping = !is_screencast;
   bool denoising;
   bool codec_default_denoising = false;
   if (is_screencast) {
     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)) {
     encoder_settings_.h264 = webrtc::VideoEncoder::GetDefaultH264Settings();
     encoder_settings_.h264.frameDroppingOn = frame_dropping;
     return &encoder_settings_.h264;
   }
   if (CodecNamesEq(codec.name, kVp8CodecName)) {
     encoder_settings_.vp8 = webrtc::VideoEncoder::GetDefaultVp8Settings();
     encoder_settings_.vp8.automaticResizeOn = automatic_resize;
@@ skipping 1083 matching lines @@
   sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
   sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
                  &parameters_.config.rtp.rtx.ssrcs);
   parameters_.config.rtp.c_name = sp.cname;
   parameters_.config.rtp.extensions = rtp_extensions;
   parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
                                          ? webrtc::RtcpMode::kReducedSize
                                          : webrtc::RtcpMode::kCompound;
 
   if (codec_settings) {
-    SetCodecAndOptions(*codec_settings, parameters_.options);
+    SetCodecAndOptions(*codec_settings);
   }
 }
 
 WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
   DisconnectCapturer();
   if (stream_ != NULL) {
     call_->DestroyVideoSendStream(stream_);
   }
   DestroyVideoEncoder(&allocated_encoder_);
 }
@@ skipping 39 matching lines @@
 
   int64_t frame_delta_ms = frame.GetTimeStamp() / rtc::kNumNanosecsPerMillisec;
   // frame->GetTimeStamp() is essentially a delta, align to webrtc time
   if (first_frame_timestamp_ms_ == 0) {
     first_frame_timestamp_ms_ = rtc::Time() - frame_delta_ms;
   }
 
   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);
 }
 
 bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
     VideoCapturer* capturer) {
   TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetCapturer");
   if (!DisconnectCapturer() && capturer == NULL) {
     return false;
@@ skipping 58 matching lines @@
 }
 
 const std::vector<uint32_t>&
 WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const {
   return ssrcs_;
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions(
     const VideoOptions& options) {
   rtc::CritScope cs(&lock_);
+  parameters_.options.SetAll(options);
   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);
   }
 }
 
 webrtc::VideoCodecType CodecTypeFromName(const std::string& name) {
   if (CodecNamesEq(name, kVp8CodecName)) {
     return webrtc::kVideoCodecVP8;
   } else if (CodecNamesEq(name, kVp9CodecName)) {
     return webrtc::kVideoCodecVP9;
   } else if (CodecNamesEq(name, kH264CodecName)) {
     return webrtc::kVideoCodecH264;
@@ skipping 37 matching lines @@
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::DestroyVideoEncoder(
     AllocatedEncoder* encoder) {
   if (encoder->external) {
     external_encoder_factory_->DestroyVideoEncoder(encoder->external_encoder);
   }
   delete encoder->encoder;
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodecAndOptions(

[pbos-webrtc, 2016/02/25 14:23:28]

Should this be SetCodecAndApplyOptions or something?

[nisse-webrtc, 2016/02/25 14:56:10]

Done.

-    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());
 
   AllocatedEncoder new_encoder = CreateVideoEncoder(codec_settings.codec);
   parameters_.config.encoder_settings.encoder = new_encoder.encoder;
   parameters_.config.encoder_settings.full_overuse_time = new_encoder.external;
   parameters_.config.encoder_settings.payload_name = codec_settings.codec.name;
   parameters_.config.encoder_settings.payload_type = codec_settings.codec.id;
   if (new_encoder.external) {
@@ skipping 11 matching lines @@
       parameters_.config.rtp.rtx.ssrcs.clear();
     } else {
       parameters_.config.rtp.rtx.payload_type = codec_settings.rtx_payload_type;
     }
   }
 
   parameters_.config.rtp.nack.rtp_history_ms =
       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_);
     allocated_encoder_ = new_encoder;
   }
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters(
     const ChangedSendParameters& params) {
   rtc::CritScope cs(&lock_);
@@ skipping 16 matching lines @@
   if (params.max_bandwidth_bps) {
     // Max bitrate has changed, reconfigure encoder settings on the next frame
     // or stream recreation.
     parameters_.max_bitrate_bps = *params.max_bandwidth_bps;
     pending_encoder_reconfiguration_ = true;
   }
   if (params.conference_mode) {
     parameters_.conference_mode = *params.conference_mode;
   }
   // Set codecs and options.
+  if (params.options)
+    parameters_.options = *params.options;

[pbos-webrtc, 2016/02/25 14:23:28]

Should this be a SetAll?

[nisse-webrtc, 2016/02/25 14:56:10]

This is a bit confusing. There is a SetAll call in GetChangedSendParameters. So
the params.options we have here is the shared options from
WebRtcVideoChannel2::send_params_.options, with the caller's options applied on
top.

I think the problem to solve is to not override the "source" options passed in
via SetVideoSend and SetOptions. I'll change it to .SetAll, because I think that
gets us closer to right. I think it will be alright as long as options belonging
to the source are never passed to SetSendParameters.

But perhaps it would be even better to use SetAll here but not with the combined
options (the result of .SetAll in GetChangedSendParameters), but the unmodified
options passed to SetSendParameters?

   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) {
     LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetSendParameters";
     RecreateWebRtcStream();
   }
 }
 
 webrtc::VideoEncoderConfig
 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 =
         webrtc::VideoEncoderConfig::ContentType::kScreen;
   } else {
     encoder_config.min_transmit_bitrate_bps = 0;
     encoder_config.content_type =
         webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo;
   }
 
   // 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)
       height = codec.height;
   }
 
   VideoCodec clamped_codec = codec;
   clamped_codec.width = width;
   clamped_codec.height = height;
 
   // By default, the stream count for the codec configuration should match the
   // 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;
   }
 
   encoder_config.streams =
       CreateVideoStreams(clamped_codec, parameters_.options,
                          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();
 
     // For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked
     // on the VideoCodec struct as target and max bitrates, respectively.
     // See eg. webrtc::VP8EncoderImpl::SetRates().
     encoder_config.streams[0].target_bitrate_bps =
         config.tl0_bitrate_kbps * 1000;
     encoder_config.streams[0].max_bitrate_bps = config.tl1_bitrate_kbps * 1000;
     encoder_config.streams[0].temporal_layer_thresholds_bps.clear();
     encoder_config.streams[0].temporal_layer_thresholds_bps.push_back(
         config.tl0_bitrate_kbps * 1000);
   }
   return encoder_config;
 }
 
 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;

[pbos-webrtc, 2016/02/25 14:23:28]

Should we log is_screencast somewhere else?

[nisse-webrtc, 2016/02/25 14:56:10]

Unless you have a concrete suggestion on where to log it, I think we can omit
logging until need arises.

 
   last_dimensions_.width = width;
   last_dimensions_.height = height;
-  last_dimensions_.is_screencast = is_screencast;
 
   RTC_DCHECK(!parameters_.encoder_config.streams.empty());
 
   RTC_CHECK(parameters_.codec_settings);
   VideoCodecSettings codec_settings = *parameters_.codec_settings;
 
   webrtc::VideoEncoderConfig encoder_config =
       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);
 
   encoder_config.encoder_specific_settings = NULL;
   pending_encoder_reconfiguration_ = false;
 
   if (!stream_reconfigured) {
     LOG(LS_WARNING) << "Failed to reconfigure video encoder for dimensions: "
                     << width << "x" << height;
     return;
@@ skipping 127 matching lines @@
   bwe_info->target_enc_bitrate += stats.target_media_bitrate_bps;
   bwe_info->actual_enc_bitrate += stats.media_bitrate_bps;
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() {
   if (stream_ != NULL) {
     call_->DestroyVideoSendStream(stream_);
   }
 
   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) {
     LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX "
                        "payload type the set codec. Ignoring RTX.";
     config.rtp.rtx.ssrcs.clear();
   }
   stream_ = call_->CreateVideoSendStream(config, parameters_.encoder_config);
 
   parameters_.encoder_config.encoder_specific_settings = NULL;
@@ skipping 419 matching lines @@
         rtx_mapping[video_codecs[i].codec.id] !=
             fec_settings.red_payload_type) {
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket