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(Auto)update libjingle 63738002-> 63773382

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@5787 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrike@webrtc.org 2014-03-26 01:17:30 +00:00
parent 440fa23553
commit dce3feb0b0
11 changed files with 107 additions and 133 deletions
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8
talk/app/webrtc/mediaconstraintsinterface.cc
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@ -77,6 +77,14 @@ const char MediaConstraintsInterface::kCpuUnderuseThreshold[] =
const char MediaConstraintsInterface::kCpuOveruseThreshold[] =
"googCpuOveruseThreshold";
// Google-specific PeerConnection constraint keys.
const char MediaConstraintsInterface::kImprovedWifiBwe[] =
"googImprovedWifiBwe";
const char MediaConstraintsInterface::kScreencastMinBitrate[] =
"googScreencastMinBitrate";
const char MediaConstraintsInterface::kSkipEncodingUnusedStreams[] =
"googSkipEncodingUnusedStreams";
// Constraint keys for CreateOffer / CreateAnswer defined in W3C specification.
const char MediaConstraintsInterface::kOfferToReceiveAudio[] =
"OfferToReceiveAudio";

7
talk/app/webrtc/mediaconstraintsinterface.h
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@ -87,12 +87,19 @@ class MediaConstraintsInterface {
// Google-specific constraint keys for a local video source
static const char kNoiseReduction[]; // googNoiseReduction
static const char kLeakyBucket[]; // googLeakyBucket
// googTemporalLayeredScreencast
static const char kTemporalLayeredScreencast[];
static const char kCpuOveruseDetection[];
static const char kCpuUnderuseThreshold[];
static const char kCpuOveruseThreshold[];
// Google-specific PeerConnection constraint keys.
static const char kImprovedWifiBwe[]; // googImprovedWifiBwe
static const char kScreencastMinBitrate[]; // googScreencastMinBitrate
static const char kSkipEncodingUnusedStreams[];
// googSkipEncodingUnusedStreams
// Constraint keys for CreateOffer / CreateAnswer
// Specified by the W3C PeerConnection spec
static const char kOfferToReceiveVideo[]; // OfferToReceiveVideo

28
talk/app/webrtc/webrtcsession.cc
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@ -517,6 +517,34 @@ bool WebRtcSession::Initialize(
video_options_.suspend_below_min_bitrate.Set(value);
}
if (FindConstraint(
constraints,
MediaConstraintsInterface::kSkipEncodingUnusedStreams,
&value,
NULL)) {
video_options_.skip_encoding_unused_streams.Set(value);
}
std::string string_value;
if (constraints &&
constraints->GetOptional().FindFirst(
MediaConstraintsInterface::kScreencastMinBitrate,
&string_value)) {
int bitrate;
if (talk_base::FromString(string_value, &bitrate)) {
video_options_.screencast_min_bitrate.Set(bitrate);
}
}
// Find improved wifi bwe constraint.
if (FindConstraint(
constraints,
MediaConstraintsInterface::kImprovedWifiBwe,
&value,
NULL)) {
video_options_.use_improved_wifi_bandwidth_estimator.Set(value);
}
const cricket::VideoCodec default_codec(
JsepSessionDescription::kDefaultVideoCodecId,
JsepSessionDescription::kDefaultVideoCodecName,

43
talk/examples/ios/AppRTCDemo/APPRTCVideoView.h
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@ -1,43 +0,0 @@
/*
* libjingle
* Copyright 2013, Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#import <UIKit/UIKit.h>
@class RTCVideoTrack;
// This class encapsulates VideoRenderIosView.
@interface APPRTCVideoView : UIView
// Property to get/set required video orientation.
@property(nonatomic, assign) UIInterfaceOrientation videoOrientation;
// Specifies whether the object represents a local or remote video stream.
@property(nonatomic, assign) BOOL isRemote;
// Sets up the underlying renderer and track objects.
- (void)renderVideoTrackInterface:(RTCVideoTrack*)track;
@end

82
talk/examples/ios/AppRTCDemo/APPRTCVideoView.m
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@ -1,82 +0,0 @@
/*
* libjingle
* Copyright 2013, Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/*
* This APPRTCVideoView must be initialzed and added to a View to get
* either the local or remote video stream rendered.
* It is a view itself and it encapsulates
* an object of VideoRenderIosView and UIActivityIndicatorView.
* Both of the views will get resized as per the frame of their parent.
*/
#import "APPRTCVideoView.h"
#import "RTCVideoRenderer.h"
#import "RTCVideoTrack.h"
@interface APPRTCVideoView () {
RTCVideoTrack* _track;
RTCVideoRenderer* _renderer;
}
@property(nonatomic, weak) UIView* renderView;
@property(nonatomic, weak) UIActivityIndicatorView* activityView;
@end
@implementation APPRTCVideoView
@synthesize videoOrientation = _videoOrientation;
- (void)layoutSubviews {
[super layoutSubviews];
if (!_renderer) {
// Left-right (mirror) flip the remote view.
CGAffineTransform xform =
CGAffineTransformMakeScale(self.isRemote ? -1 : 1, 1);
// TODO(fischman): why is this rotate (vertical+horizontal flip) needed?!?
xform = CGAffineTransformRotate(xform, M_PI);
// TODO(fischman): ensure back-camera flip is correct in all orientations,
// when back-camera support is added.
[self setTransform:xform];
_renderer = [[RTCVideoRenderer alloc] initWithView:self];
}
}
- (void)renderVideoTrackInterface:(RTCVideoTrack*)videoTrack {
[_track removeRenderer:_renderer];
[_renderer stop];
_track = videoTrack;
if (_track) {
[_track addRenderer:_renderer];
[_renderer start];
}
}
@end

22
talk/media/base/mediachannel.h
View File

@ -329,6 +329,10 @@ struct VideoOptions {
suspend_below_min_bitrate.SetFrom(change.suspend_below_min_bitrate);
unsignalled_recv_stream_limit.SetFrom(change.unsignalled_recv_stream_limit);
use_simulcast_adapter.SetFrom(change.use_simulcast_adapter);
skip_encoding_unused_streams.SetFrom(change.skip_encoding_unused_streams);
screencast_min_bitrate.SetFrom(change.screencast_min_bitrate);
use_improved_wifi_bandwidth_estimator.SetFrom(
change.use_improved_wifi_bandwidth_estimator);
}
bool operator==(const VideoOptions& o) const {
@ -359,7 +363,11 @@ struct VideoOptions {
dscp == o.dscp &&
suspend_below_min_bitrate == o.suspend_below_min_bitrate &&
unsignalled_recv_stream_limit == o.unsignalled_recv_stream_limit &&
use_simulcast_adapter == o.use_simulcast_adapter;
use_simulcast_adapter == o.use_simulcast_adapter &&
skip_encoding_unused_streams == o.skip_encoding_unused_streams &&
screencast_min_bitrate == o.screencast_min_bitrate;
use_improved_wifi_bandwidth_estimator ==
o.use_improved_wifi_bandwidth_estimator;
}
std::string ToString() const {
@ -395,6 +403,11 @@ struct VideoOptions {
ost << ToStringIfSet("num channels for early receive",
unsignalled_recv_stream_limit);
ost << ToStringIfSet("use simulcast adapter", use_simulcast_adapter);
ost << ToStringIfSet("skip encoding unused streams",
skip_encoding_unused_streams);
ost << ToStringIfSet("screencast min bitrate", screencast_min_bitrate);
ost << ToStringIfSet("improved wifi bwe",
use_improved_wifi_bandwidth_estimator);
ost << "}";
return ost.str();
}
@ -454,6 +467,13 @@ struct VideoOptions {
Settable<int> unsignalled_recv_stream_limit;
// Enable use of simulcast adapter.
Settable<bool> use_simulcast_adapter;
// Enables the encoder to skip encoding stream not actually sent due to too
// low available bit rate.
Settable<bool> skip_encoding_unused_streams;
// Force screencast to use a minimum bitrate
Settable<int> screencast_min_bitrate;
// Enable improved bandwidth estiamtor on wifi.
Settable<bool> use_improved_wifi_bandwidth_estimator;
};
// A class for playing out soundclips.

0
talk/media/other/androidmediaengine.cc
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0
talk/media/other/androidmediaengine.h
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4
talk/media/webrtc/fakewebrtcvideoengine.h
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@ -244,10 +244,6 @@ class FakeWebRtcVideoEncoderFactory : public WebRtcVideoEncoderFactory {
(*it)->OnCodecsAvailable();
}
int GetNumCreatedEncoders() {
return num_created_encoders_;
}
const std::vector<FakeWebRtcVideoEncoder*>& encoders() {
return encoders_;
}

43
talk/media/webrtc/webrtcvideoengine.cc
View File

@ -2837,6 +2837,13 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
conference_mode_turned_off = true;
}
#ifdef USE_WEBRTC_DEV_BRANCH
bool improved_wifi_bwe_changed =
options.use_improved_wifi_bandwidth_estimator.IsSet() &&
options_.use_improved_wifi_bandwidth_estimator !=
options.use_improved_wifi_bandwidth_estimator;
#endif
// Save the options, to be interpreted where appropriate.
// Use options_.SetAll() instead of assignment so that unset value in options
@ -2961,6 +2968,19 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
LOG(LS_WARNING) << "Cannot disable video suspension once it is enabled";
}
}
#ifdef USE_WEBRTC_DEV_BRANCH
if (improved_wifi_bwe_changed) {
webrtc::Config config;
config.Set(new webrtc::AimdRemoteRateControl(
options_.use_improved_wifi_bandwidth_estimator
.GetWithDefaultIfUnset(false)));
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
engine()->vie()->network()->SetBandwidthEstimationConfig(
it->second->channel_id(), config);
}
}
#endif
return true;
}
@ -3753,6 +3773,11 @@ bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
// Disable denoising for screencasting.
bool enable_denoising =
options_.video_noise_reduction.GetWithDefaultIfUnset(false);
#ifdef USE_WEBRTC_DEV_BRANCH
int screencast_min_bitrate =
options_.screencast_min_bitrate.GetWithDefaultIfUnset(0);
bool leaky_bucket = options_.video_leaky_bucket.GetWithDefaultIfUnset(false);
#endif
bool denoising = !is_screencast && enable_denoising;
bool reset_send_codec =
target_width != cur_width || target_height != cur_height ||
@ -3786,6 +3811,24 @@ bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
LOG_RTCERR1(SetSendCodec, channel_id);
return false;
}
#ifdef USE_WEBRTC_DEV_BRANCH
if (is_screencast) {
engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id,
screencast_min_bitrate);
// If screencast and min bitrate set, force enable pacer.
if (screencast_min_bitrate > 0) {
engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id,
true);
}
} else {
// In case of switching from screencast to regular capture, set
// min bitrate padding and pacer back to defaults.
engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id, 0);
engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id,
leaky_bucket);
}
#endif
if (reset) {
*reset = true;
}

3
talk/media/webrtc/webrtcvideoengine_unittest.cc
View File

@ -1662,11 +1662,9 @@ TEST_F(WebRtcVideoEngineTestFake, DontRegisterEncoderMultipleTimes) {
EXPECT_TRUE(vie_.ExternalEncoderRegistered(channel_num, 100));
EXPECT_EQ(1, vie_.GetNumExternalEncoderRegistered(channel_num));
EXPECT_EQ(1, encoder_factory_.GetNumCreatedEncoders());
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
EXPECT_EQ(1, vie_.GetNumExternalEncoderRegistered(channel_num));
EXPECT_EQ(1, encoder_factory_.GetNumCreatedEncoders());
// Remove stream previously added to free the external encoder instance.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
@ -1798,7 +1796,6 @@ TEST_F(WebRtcVideoEngineTestFake, UpdateEncoderCodecsAfterSetFactory) {
EXPECT_TRUE(vie_.ExternalEncoderRegistered(channel_num, 100));
EXPECT_EQ(1, vie_.GetNumExternalEncoderRegistered(channel_num));
EXPECT_EQ(1, encoder_factory_.GetNumCreatedEncoders());
// Remove stream previously added to free the external encoder instance.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));