(Auto)update libjingle 79104430-> 79104922
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7602 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/webrtcvideoengine.cc b/talk/media/webrtc/webrtcvideoengine.cc
index 451b84c..04092f3 100644
--- a/talk/media/webrtc/webrtcvideoengine.cc
+++ b/talk/media/webrtc/webrtcvideoengine.cc
@@ -3879,47 +3879,22 @@
// only needed because some unit tests bypass the VideoAdapter, and
// others expect behavior from the adapter different than what it
// actually does. We should fix the tests and remove this block.
- int frame_width = static_cast<int>(frame.width);
- int frame_height = static_cast<int>(frame.height);
VideoFormat max = send_channel->adapt_format();
+ size_t max_width = static_cast<size_t>(max.width);
+ size_t max_height = static_cast<size_t>(max.height);
if (!send_channel->last_captured_frame_info().IsSet() ||
(!frame.screencast &&
- (frame_width > max.width || frame_height > max.height))) {
- frame_width = max.width;
- frame_height = max.height;
+ (frame.width > max_width || frame.height > max_height))) {
+ frame.width = max_width;
+ frame.height = max_height;
}
- // Set the new codec on vie.
- webrtc::VideoCodec codec = send_params.codec;
-
- // Settings for both screencast and non-screencast
- codec.width = frame_width;
- codec.height = frame_height;
+ webrtc::VideoCodec codec;
+ ConfigureVieCodecFromSendParams(channel_id, send_params, frame, &codec);
+ // TODO(pthatcher): Figure out a clean way to configure the max
+ // framerate and sanitize the bitrates inside of
+ // ConfigureVieCodecFromSendParams.
codec.maxFramerate = max.framerate();
- codec.targetBitrate = 0;
- if (codec.codecType == webrtc::kVideoCodecVP8) {
- codec.codecSpecific.VP8.numberOfTemporalLayers =
- kDefaultNumberOfTemporalLayers;
- codec.codecSpecific.VP8.resilience = webrtc::kResilienceOff;
- }
- if (frame.screencast) {
- // Settings for screencast
- codec.mode = webrtc::kScreensharing;
- if (codec.codecType == webrtc::kVideoCodecVP8) {
- codec.codecSpecific.VP8.denoisingOn = false;
- codec.codecSpecific.VP8.automaticResizeOn = false;
- codec.codecSpecific.VP8.frameDroppingOn = false;
- }
- } else {
- // Settings for non-screencast
- codec.mode = webrtc::kRealtimeVideo;
- if (codec.codecType == webrtc::kVideoCodecVP8) {
- codec.codecSpecific.VP8.denoisingOn =
- options_.video_noise_reduction.GetWithDefaultIfUnset(true);
- codec.codecSpecific.VP8.automaticResizeOn = true;
- codec.codecSpecific.VP8.frameDroppingOn = true;
- }
- }
SanitizeBitrates(channel_id, &codec);
// Get current vie codec.
@@ -3958,6 +3933,44 @@
return true;
}
+bool WebRtcVideoMediaChannel::ConfigureVieCodecFromSendParams(
+ int channel_id,
+ const VideoSendParams& send_params,
+ const CapturedFrameInfo& last_captured_frame_info,
+ webrtc::VideoCodec* codec_out) {
+ webrtc::VideoCodec codec = send_params.codec;
+
+ codec.width = static_cast<int>(last_captured_frame_info.width);
+ codec.height = static_cast<int>(last_captured_frame_info.height);
+ codec.targetBitrate = 0;
+ if (codec.codecType == webrtc::kVideoCodecVP8) {
+ codec.codecSpecific.VP8.numberOfTemporalLayers =
+ kDefaultNumberOfTemporalLayers;
+ codec.codecSpecific.VP8.resilience = webrtc::kResilienceOff;
+ }
+
+ if (last_captured_frame_info.screencast) {
+ codec.mode = webrtc::kScreensharing;
+ if (codec.codecType == webrtc::kVideoCodecVP8) {
+ codec.codecSpecific.VP8.denoisingOn = false;
+ codec.codecSpecific.VP8.automaticResizeOn = false;
+ codec.codecSpecific.VP8.frameDroppingOn = false;
+ }
+ } else {
+ codec.mode = webrtc::kRealtimeVideo;
+ if (codec.codecType == webrtc::kVideoCodecVP8) {
+ // TODO(pthatcher): Pass in options in VideoSendParams.
+ codec.codecSpecific.VP8.denoisingOn =
+ options_.video_noise_reduction.GetWithDefaultIfUnset(true);
+ codec.codecSpecific.VP8.automaticResizeOn = true;
+ codec.codecSpecific.VP8.frameDroppingOn = true;
+ }
+ }
+
+ *codec_out = codec;
+ return true;
+}
+
void WebRtcVideoMediaChannel::SanitizeBitrates(
int channel_id, webrtc::VideoCodec* codec) {
codec->minBitrate = GetBitrate(codec->minBitrate, kMinVideoBitrate);
diff --git a/talk/media/webrtc/webrtcvideoengine.h b/talk/media/webrtc/webrtcvideoengine.h
index 26b0453..cc81ee9 100644
--- a/talk/media/webrtc/webrtcvideoengine.h
+++ b/talk/media/webrtc/webrtcvideoengine.h
@@ -336,6 +336,18 @@
return options_.conference_mode.GetWithDefaultIfUnset(false);
}
+ // We take lots of things as input from applications (packaged in
+ // params), but ViE wants lots of those packed instead as a
+ // webrtc::VideoCodec. This is where we convert between the inputs
+ // we get from the applications and the input to give to ViE. We
+ // also configure the codec differently depending on the latest
+ // frame that we have received (in particular, depending on the
+ // resolution and whether the it was a screencast frame or not).
+ virtual bool ConfigureVieCodecFromSendParams(
+ int channel_id,
+ const VideoSendParams& send_params,
+ const CapturedFrameInfo& last_captured_frame_info,
+ webrtc::VideoCodec* codec);
// Checks the current bitrate estimate and modifies the bitrates
// accordingly, including converting kAutoBandwidth to the correct defaults.
virtual void SanitizeBitrates(
