Avoid unnecessary HW video encoder reconfiguration

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 82d32b0e1865dff74586b16030a8731a6634a5ec..ea6012fc42f597b865eca72e2dc4c55196fe787e 100644
--- a/webrtc/media/engine/webrtcvideoengine2.cc
+++ b/webrtc/media/engine/webrtcvideoengine2.cc
@@ -748,7 +748,7 @@ bool WebRtcVideoChannel2::GetChangedSendParameters(
   // Handle RTP header extensions.
   std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
       params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true);
-  if (send_rtp_extensions_ != filtered_extensions) {
+  if (!send_rtp_extensions_ || (*send_rtp_extensions_ != filtered_extensions)) {
     changed_params->rtp_header_extensions =
         rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
   }
@@ -796,7 +796,7 @@ bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) {
   }
 
   if (changed_params.rtp_header_extensions) {
-    send_rtp_extensions_ = *changed_params.rtp_header_extensions;
+    send_rtp_extensions_ = changed_params.rtp_header_extensions;
   }
 
   if (changed_params.codec || changed_params.max_bandwidth_bps) {
@@ -1520,7 +1520,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
     bool enable_cpu_overuse_detection,
     int max_bitrate_bps,
     const rtc::Optional<VideoCodecSettings>& codec_settings,
-    const std::vector<webrtc::RtpExtension>& rtp_extensions,
+    const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
     // TODO(deadbeef): Don't duplicate information between send_params,
     // rtp_extensions, options, etc.
     const VideoSendParameters& send_params)
@@ -1546,15 +1546,19 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
   sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
                  &parameters_.config.rtp.rtx.ssrcs);
   parameters_.config.rtp.c_name = sp.cname;
-  parameters_.config.rtp.extensions = rtp_extensions;
+  if (rtp_extensions) {
+    parameters_.config.rtp.extensions = *rtp_extensions;
+  }
   parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
                                          ? webrtc::RtcpMode::kReducedSize
                                          : webrtc::RtcpMode::kCompound;
   parameters_.config.overuse_callback =
       enable_cpu_overuse_detection ? this : nullptr;
 
-  sink_wants_.rotation_applied = !ContainsHeaderExtension(
-      rtp_extensions, webrtc::RtpExtension::kVideoRotationUri);
+  sink_wants_.rotation_applied =
+      rtp_extensions &&
+      !ContainsHeaderExtension(*rtp_extensions,
+                               webrtc::RtpExtension::kVideoRotationUri);

[pthatcher1, 2016/06/15 20:40:23]

Can you leave a comment explaining that if it's not set, we want rotation to not
be applied, and why not?

[skvlad, 2016/06/15 22:10:36]

Added a comment.

 
   if (codec_settings) {
     SetCodec(*codec_settings);
@@ -1593,6 +1597,20 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame(
   webrtc::VideoFrame video_frame(frame.video_frame_buffer(), 0, 0,
                                  frame.rotation());
   rtc::CritScope cs(&lock_);
+
+  last_rotation_ = video_frame.rotation();
+  last_frame_is_texture_ =
+      video_frame.video_frame_buffer()->native_handle() != NULL;

[tommi, 2016/06/15 21:00:39]

nit: nullptr

[skvlad, 2016/06/15 22:10:36]

Done, and moved into a webrtc::VideoFrame class method.

+  if (video_frame.width() != last_dimensions_.width ||
+      video_frame.height() != last_dimensions_.height) {
+    last_dimensions_.width = video_frame.width();
+    last_dimensions_.height = video_frame.height();
+    pending_encoder_reconfiguration_ = true;
+    LOG(LS_INFO) << "Caching frame dimensions: " << last_dimensions_.width
+                 << "x" << last_dimensions_.height << ", r=" << last_rotation_
+                 << ", texture=" << last_frame_is_texture_;
+  }
+
   if (stream_ == NULL) {
     // Frame input before send codecs are configured, dropping frame.
     return;
@@ -1609,9 +1627,8 @@ 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());
-  last_rotation_ = video_frame.rotation();
+
+  ReconfigureEncoderIfNecessary();
 
   // Not sending, abort after reconfiguration. Reconfiguration should still
   // occur to permit sending this input as quickly as possible once we start
@@ -1758,8 +1775,8 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::DestroyVideoEncoder(
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec(
     const VideoCodecSettings& codec_settings) {
-  parameters_.encoder_config =
-      CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
+  parameters_.encoder_config = CreateVideoEncoderConfig(
+      last_dimensions_, last_frame_is_texture_, codec_settings.codec);
   RTC_DCHECK(!parameters_.encoder_config.streams.empty());
 
   AllocatedEncoder new_encoder = CreateVideoEncoder(codec_settings.codec);
@@ -1899,6 +1916,7 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::UpdateSendState() {
 webrtc::VideoEncoderConfig
 WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
     const Dimensions& dimensions,
+    bool encode_from_texture,

[pthatcher1, 2016/06/15 20:40:23]

Since these always come from last_dimentions_ and last_frame_is_texture_, why
pass them in?  Why not just use the member variables?

[skvlad, 2016/06/15 22:10:36]

Using member variables now.

     const VideoCodec& codec) const {
   webrtc::VideoEncoderConfig encoder_config;
   bool is_screencast = parameters_.options.is_screencast.value_or(false);
@@ -1940,6 +1958,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
                   parameters_.max_bitrate_bps);
   encoder_config.streams = CreateVideoStreams(
       clamped_codec, parameters_.options, stream_max_bitrate, stream_count);
+  encoder_config.encode_from_texture = encode_from_texture;
 
   // Conference mode screencast uses 2 temporal layers split at 100kbit.
   if (parameters_.conference_mode && is_screencast &&
@@ -1964,25 +1983,20 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
   return encoder_config;
 }
 
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
-    int width,
-    int height) {
-  if (last_dimensions_.width == width && last_dimensions_.height == height &&
-      !pending_encoder_reconfiguration_) {
+void WebRtcVideoChannel2::WebRtcVideoSendStream::
+    ReconfigureEncoderIfNecessary() {

[pthatcher1, 2016/06/15 20:40:23]

Why not just call this ReconfigureEncoder() and have the call site be:

if (pending_encoder_reconfiguration_) {
  ReconfigureEncoder();
  pending_encoder_reconfiguration_ = false;
}

?  

I think that would be more clear, especially since there is only one such call
site.  Maybe later if we have more than one, we could re-add
ReconfigureEncoderIfNecessary, but I don't think we will.

[skvlad, 2016/06/15 22:10:36]

Done.

+  if (!pending_encoder_reconfiguration_) {
     // Configured using the same parameters, do not reconfigure.
     return;
   }
 
-  last_dimensions_.width = width;
-  last_dimensions_.height = height;
-
   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);
+  webrtc::VideoEncoderConfig encoder_config = CreateVideoEncoderConfig(
+      last_dimensions_, last_frame_is_texture_, codec_settings.codec);
 
   encoder_config.encoder_specific_settings = ConfigureVideoEncoderSettings(
       codec_settings.codec);