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(Auto)update libjingle 78344087-> 78381351

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7515 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-10-23 21:36:17 +00:00
parent 7e19a11a71
commit fb5e9fc44e
2 changed files with 66 additions and 34 deletions
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78
talk/media/webrtc/webrtcvideoengine.cc
View File

@ -638,6 +638,9 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
int capture_id() const { return capture_id_; }
void set_sending(bool sending) { sending_ = sending; }
bool sending() const { return sending_; }
const Settable<CapturedFrameInfo>& last_captured_frame_info() const {
return last_captured_frame_info_;
}
void set_muted(bool on) {
// TODO(asapersson): add support.
// video_adapter_.SetBlackOutput(on);
@ -653,6 +656,23 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
return adapt_format_type() != kAdaptFormatTypeNone;
}
// Returns true if the last captured frame info changed.
void SetLastCapturedFrameInfo(
const VideoFrame* frame, bool screencast, bool* changed) {
CapturedFrameInfo last;
if (last_captured_frame_info_.Get(&last) &&
frame->GetWidth() == last.width &&
frame->GetHeight() == last.height &&
screencast == last.screencast) {
*changed = false;
return;
}
last_captured_frame_info_.Set(CapturedFrameInfo(
frame->GetWidth(), frame->GetHeight(), screencast));
*changed = true;
}
// Tells the video adapter to adapt down to a given format. The
// type indicates where the format came from, where different types
// have slightly different behavior and priority.
@ -866,6 +886,8 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
private:
int channel_id_;
int capture_id_;
// TODO(pthatcher): Merge CapturedFrameInfo and LocalStreamInfo.
Settable<CapturedFrameInfo> last_captured_frame_info_;
bool sending_;
bool muted_;
VideoCapturer* video_capturer_;
@ -3179,15 +3201,20 @@ bool WebRtcVideoMediaChannel::SendFrame(
return true;
}
// Checks if we need to reset vie send codec.
if (!MaybeResetVieSendCodec(send_channel,
static_cast<int>(frame->GetWidth()),
static_cast<int>(frame->GetHeight()),
is_screencast, NULL)) {
LOG(LS_ERROR) << "MaybeResetVieSendCodec failed with "
<< frame->GetWidth() << "x" << frame->GetHeight();
bool changed;
send_channel->SetLastCapturedFrameInfo(frame, is_screencast, &changed);
if (changed) {
// If the last captured frame info changed, then calling
// SendParams will update to the latest resolution.
if (!SetSendParams(send_channel, send_channel->stream_params())) {
LOG(LS_ERROR) << "SetSendParams from SendFrame failed with "
<< frame->GetWidth() << "x" << frame->GetHeight()
<< " screencast? " << is_screencast;
return false;
}
LogSendCodecChange("Captured frame size changed");
}
const VideoFrame* frame_out = frame;
rtc::scoped_ptr<VideoFrame> processed_frame;
// TODO(hellner): Remove the need for disabling mute when screencasting.
@ -3879,17 +3906,17 @@ int WebRtcVideoMediaChannel::GetRecvChannelId(uint32 ssrc) {
// only by the 'jec' logic.
// TODO(pthatcher): Get rid of this function, so we only ever set up
// codecs in a single place.
bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
bool WebRtcVideoMediaChannel::SetSendParams(
WebRtcVideoChannelSendInfo* send_channel,
int new_width,
int new_height,
bool is_screencast,
bool* reset) {
if (reset) {
*reset = false;
}
const StreamParams* send_params) {
ASSERT(send_codec_.get() != NULL);
CapturedFrameInfo frame;
if (!send_channel->last_captured_frame_info().Get(&frame)) {
// No captured frame yet, so nothing to set.
return true;
}
webrtc::VideoCodec target_codec = *send_codec_;
// TODO(pthatcher): We should rely on the adapter to adapt the
// resolution, and not do it here.
@ -3901,10 +3928,11 @@ bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
}
// Vie send codec size should not exceed target_codec.
int target_width = new_width;
int target_height = new_height;
if (!is_screencast &&
(new_width > target_codec.width || new_height > target_codec.height)) {
int target_width = static_cast<int>(frame.width);
int target_height = static_cast<int>(frame.height);
if (!frame.screencast &&
(target_width > target_codec.width ||
target_height > target_codec.height)) {
target_width = target_codec.width;
target_height = target_codec.height;
}
@ -3923,21 +3951,20 @@ bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
// automatic resize needs to be turned off when screencasting and on when
// not screencasting.
// Don't allow automatic resizing for screencasting.
bool automatic_resize = !is_screencast;
bool automatic_resize = !frame.screencast;
// Turn off VP8 frame dropping when screensharing as the current model does
// not work well at low fps.
bool vp8_frame_dropping = !is_screencast;
bool vp8_frame_dropping = !frame.screencast;
// TODO(pbos): Remove |video_noise_reduction| and enable it for all
// non-screencast.
bool enable_denoising =
options_.video_noise_reduction.GetWithDefaultIfUnset(true);
// Disable denoising for screencasting.
if (is_screencast) {
if (frame.screencast) {
enable_denoising = false;
}
int screencast_min_bitrate =
options_.screencast_min_bitrate.GetWithDefaultIfUnset(0);
StreamParams* send_params = send_channel->stream_params();
bool reset_send_codec =
target_width != cur_width || target_height != cur_height;
if (vie_codec.codecType == webrtc::kVideoCodecVP8) {
@ -3968,7 +3995,7 @@ bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
return false;
}
if (is_screencast) {
if (frame.screencast) {
engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id,
screencast_min_bitrate);
} else {
@ -3984,9 +4011,6 @@ bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
return false;
}
}
if (reset) {
*reset = true;
}
LogSendCodecChange("Capture size changed");
}

20
talk/media/webrtc/webrtcvideoengine.h
View File

@ -234,6 +234,16 @@ class WebRtcVideoEngine : public sigslot::has_slots<>,
rtc::scoped_ptr<rtc::CpuMonitor> cpu_monitor_;
};
struct CapturedFrameInfo {
CapturedFrameInfo() : width(0), height(0), screencast(false) {}
CapturedFrameInfo(size_t width, size_t height, bool screencast) :
width(width), height(height), screencast(screencast) {}
size_t width;
size_t height;
bool screencast;
};
class WebRtcVideoMediaChannel : public rtc::MessageHandler,
public VideoMediaChannel,
public webrtc::Transport {
@ -363,6 +373,10 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler,
bool SetSendCodec(const webrtc::VideoCodec& codec);
bool SetSendCodec(WebRtcVideoChannelSendInfo* send_channel,
const webrtc::VideoCodec& codec);
// TODO(pthatcher): Include codec and VideoOptions as well as StreamParams.
bool SetSendParams(WebRtcVideoChannelSendInfo* send_channel,
const StreamParams* params);
// Prepares the channel with channel id |info->channel_id()| to receive all
// codecs in |receive_codecs_| and start receive packets.
bool SetReceiveCodecs(WebRtcVideoChannelRecvInfo* info);
@ -374,12 +388,6 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler,
bool MaybeRegisterExternalEncoder(
WebRtcVideoChannelSendInfo* send_channel,
const webrtc::VideoCodec& codec);
// Given captured video frame size, checks if we need to reset vie send codec.
// |reset| is set to whether resetting has happened on vie or not.
// Returns false on error.
bool MaybeResetVieSendCodec(WebRtcVideoChannelSendInfo* send_channel,
int new_width, int new_height, bool is_screencast,
bool* reset);
// Helper function for starting the sending of media on all channels or
// |channel_id|. Note that these two function do not change |sending_|.
bool StartSend();