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(Auto)update libjingle 66798415-> 66813165

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6110 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-05-12 18:42:49 +00:00
parent 0df2ea064f
commit 17911dca80
12 changed files with 13 additions and 3391 deletions
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26
talk/app/webrtc/peerconnectionfactory.cc
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@ -111,9 +111,9 @@ enum {
namespace webrtc {
talk_base::scoped_refptr<PeerConnectionFactoryInterface>
scoped_refptr<PeerConnectionFactoryInterface>
CreatePeerConnectionFactory() {
talk_base::scoped_refptr<PeerConnectionFactory> pc_factory(
scoped_refptr<PeerConnectionFactory> pc_factory(
new talk_base::RefCountedObject<PeerConnectionFactory>());
if (!pc_factory->Initialize()) {
@ -122,19 +122,17 @@ CreatePeerConnectionFactory() {
return pc_factory;
}
talk_base::scoped_refptr<PeerConnectionFactoryInterface>
scoped_refptr<PeerConnectionFactoryInterface>
CreatePeerConnectionFactory(
talk_base::Thread* worker_thread,
talk_base::Thread* signaling_thread,
AudioDeviceModule* default_adm,
cricket::WebRtcVideoEncoderFactory* encoder_factory,
cricket::WebRtcVideoDecoderFactory* decoder_factory) {
talk_base::scoped_refptr<PeerConnectionFactory> pc_factory(
new talk_base::RefCountedObject<PeerConnectionFactory>(worker_thread,
signaling_thread,
default_adm,
encoder_factory,
decoder_factory));
scoped_refptr<PeerConnectionFactory> pc_factory(
new talk_base::RefCountedObject<PeerConnectionFactory>(
worker_thread, signaling_thread, default_adm,
encoder_factory, decoder_factory));
if (!pc_factory->Initialize()) {
return NULL;
}
@ -280,7 +278,7 @@ bool PeerConnectionFactory::StartAecDump_s(talk_base::PlatformFile file) {
return channel_manager_->StartAecDump(file);
}
talk_base::scoped_refptr<PeerConnectionInterface>
scoped_refptr<PeerConnectionInterface>
PeerConnectionFactory::CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
const MediaConstraintsInterface* constraints,
@ -316,7 +314,7 @@ PeerConnectionFactory::CreatePeerConnection_s(
return PeerConnectionProxy::Create(signaling_thread(), pc);
}
talk_base::scoped_refptr<MediaStreamInterface>
scoped_refptr<MediaStreamInterface>
PeerConnectionFactory::CreateLocalMediaStream(const std::string& label) {
return MediaStreamProxy::Create(signaling_thread_,
MediaStream::Create(label));
@ -350,9 +348,9 @@ PeerConnectionFactory::CreateVideoTrack(
return VideoTrackProxy::Create(signaling_thread_, track);
}
talk_base::scoped_refptr<AudioTrackInterface>
PeerConnectionFactory::CreateAudioTrack(const std::string& id,
AudioSourceInterface* source) {
scoped_refptr<AudioTrackInterface> PeerConnectionFactory::CreateAudioTrack(
const std::string& id,
AudioSourceInterface* source) {
talk_base::scoped_refptr<AudioTrackInterface> track(
AudioTrack::Create(id, source));
return AudioTrackProxy::Create(signaling_thread_, track);

4
talk/libjingle.gyp
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@ -806,7 +806,7 @@
'<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib',
'<(webrtc_root)/modules/modules.gyp:video_capture_module',
'<(webrtc_root)/modules/modules.gyp:video_render_module',
'<(webrtc_root)/webrtc.gyp:webrtc',
'<(webrtc_root)/video_engine/video_engine.gyp:video_engine_core',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'libjingle',
@ -889,8 +889,6 @@
'media/webrtc/webrtcvideoencoderfactory.h',
'media/webrtc/webrtcvideoengine.cc',
'media/webrtc/webrtcvideoengine.h',
'media/webrtc/webrtcvideoengine2.cc',
'media/webrtc/webrtcvideoengine2.h',
'media/webrtc/webrtcvideoframe.cc',
'media/webrtc/webrtcvideoframe.h',
'media/webrtc/webrtcvie.h',

1
talk/libjingle_tests.gyp
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@ -300,7 +300,6 @@
# TODO(ronghuawu): Reenable these tests.
# 'media/devices/devicemanager_unittest.cc',
'media/webrtc/webrtcvideoengine_unittest.cc',
'media/webrtc/webrtcvideoengine2_unittest.cc',
'media/webrtc/webrtcvoiceengine_unittest.cc',
],
'conditions': [

50
talk/media/base/codec.cc
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@ -31,7 +31,6 @@
#include <sstream>
#include "talk/base/common.h"
#include "talk/base/logging.h"
#include "talk/base/stringencode.h"
#include "talk/base/stringutils.h"
@ -161,55 +160,6 @@ std::string VideoCodec::ToString() const {
return os.str();
}
VideoCodec VideoCodec::CreateRtxCodec(int rtx_payload_type,
int associated_payload_type) {
VideoCodec rtx_codec(rtx_payload_type, kRtxCodecName, 0, 0, 0, 0);
rtx_codec.SetParam(kCodecParamAssociatedPayloadType, associated_payload_type);
return rtx_codec;
}
VideoCodec::CodecType VideoCodec::GetCodecType() const {
const char* payload_name = name.c_str();
if (_stricmp(payload_name, kRedCodecName) == 0) {
return CODEC_RED;
}
if (_stricmp(payload_name, kUlpfecCodecName) == 0) {
return CODEC_ULPFEC;
}
if (_stricmp(payload_name, kRtxCodecName) == 0) {
return CODEC_RTX;
}
return CODEC_VIDEO;
}
bool VideoCodec::ValidateCodecFormat() const {
if (id < 0 || id > 127) {
LOG(LS_ERROR) << "Codec with invalid payload type: " << ToString();
return false;
}
if (GetCodecType() != CODEC_VIDEO) {
return true;
}
// Video validation from here on.
if (width <= 0 || height <= 0) {
LOG(LS_ERROR) << "Codec with invalid dimensions: " << ToString();
return false;
}
int min_bitrate;
int max_bitrate;
if (GetParam(kCodecParamMinBitrate, &min_bitrate) &&
GetParam(kCodecParamMaxBitrate, &max_bitrate)) {
if (max_bitrate < min_bitrate) {
LOG(LS_ERROR) << "Codec with max < min bitrate: " << ToString();
return false;
}
}
return true;
}
std::string DataCodec::ToString() const {
std::ostringstream os;
os << "DataCodec[" << id << ":" << name << "]";

16
talk/media/base/codec.h
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@ -246,22 +246,6 @@ struct VideoCodec : public Codec {
bool operator!=(const VideoCodec& c) const {
return !(*this == c);
}
static VideoCodec CreateRtxCodec(int rtx_payload_type,
int associated_payload_type);
enum CodecType {
CODEC_VIDEO,
CODEC_RED,
CODEC_ULPFEC,
CODEC_RTX,
};
CodecType GetCodecType() const;
// Validates a VideoCodec's payload type, dimensions and bitrates etc. If they
// don't make sense (such as max < min bitrate), and error is logged and
// ValidateCodecFormat returns false.
bool ValidateCodecFormat() const;
};
struct DataCodec : public Codec {

81
talk/media/base/codec_unittest.cc
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@ -34,9 +34,6 @@ using cricket::DataCodec;
using cricket::FeedbackParam;
using cricket::VideoCodec;
using cricket::VideoEncoderConfig;
using cricket::kCodecParamAssociatedPayloadType;
using cricket::kCodecParamMaxBitrate;
using cricket::kCodecParamMinBitrate;
class CodecTest : public testing::Test {
public:
@ -315,81 +312,3 @@ TEST_F(CodecTest, TestIntersectFeedbackParams) {
EXPECT_FALSE(c1.HasFeedbackParam(b2));
EXPECT_FALSE(c1.HasFeedbackParam(c3));
}
TEST_F(CodecTest, TestGetCodecType) {
// Codec type comparison should be case insenstive on names.
const VideoCodec codec(96, "V", 320, 200, 30, 3);
const VideoCodec rtx_codec(96, "rTx", 320, 200, 30, 3);
const VideoCodec ulpfec_codec(96, "ulpFeC", 320, 200, 30, 3);
const VideoCodec red_codec(96, "ReD", 320, 200, 30, 3);
EXPECT_EQ(VideoCodec::CODEC_VIDEO, codec.GetCodecType());
EXPECT_EQ(VideoCodec::CODEC_RTX, rtx_codec.GetCodecType());
EXPECT_EQ(VideoCodec::CODEC_ULPFEC, ulpfec_codec.GetCodecType());
EXPECT_EQ(VideoCodec::CODEC_RED, red_codec.GetCodecType());
}
TEST_F(CodecTest, TestCreateRtxCodec) {
VideoCodec rtx_codec = VideoCodec::CreateRtxCodec(96, 120);
EXPECT_EQ(96, rtx_codec.id);
EXPECT_EQ(VideoCodec::CODEC_RTX, rtx_codec.GetCodecType());
int associated_payload_type;
ASSERT_TRUE(rtx_codec.GetParam(kCodecParamAssociatedPayloadType,
&associated_payload_type));
EXPECT_EQ(120, associated_payload_type);
}
TEST_F(CodecTest, TestValidateCodecFormat) {
const VideoCodec codec(96, "V", 320, 200, 30, 3);
ASSERT_TRUE(codec.ValidateCodecFormat());
// Accept 0-127 as payload types.
VideoCodec low_payload_type = codec;
low_payload_type.id = 0;
VideoCodec high_payload_type = codec;
high_payload_type.id = 127;
ASSERT_TRUE(low_payload_type.ValidateCodecFormat());
EXPECT_TRUE(high_payload_type.ValidateCodecFormat());
// Reject negative payloads.
VideoCodec negative_payload_type = codec;
negative_payload_type.id = -1;
EXPECT_FALSE(negative_payload_type.ValidateCodecFormat());
// Reject too-high payloads.
VideoCodec too_high_payload_type = codec;
too_high_payload_type.id = 128;
EXPECT_FALSE(too_high_payload_type.ValidateCodecFormat());
// Reject zero-width codecs.
VideoCodec zero_width = codec;
zero_width.width = 0;
EXPECT_FALSE(zero_width.ValidateCodecFormat());
// Reject zero-height codecs.
VideoCodec zero_height = codec;
zero_height.height = 0;
EXPECT_FALSE(zero_height.ValidateCodecFormat());
// Accept non-video codecs with zero dimensions.
VideoCodec zero_width_rtx_codec = VideoCodec::CreateRtxCodec(96, 120);
zero_width_rtx_codec.width = 0;
EXPECT_TRUE(zero_width_rtx_codec.ValidateCodecFormat());
// Reject codecs with min bitrate > max bitrate.
VideoCodec incorrect_bitrates = codec;
incorrect_bitrates.params[kCodecParamMinBitrate] = "100";
incorrect_bitrates.params[kCodecParamMaxBitrate] = "80";
EXPECT_FALSE(incorrect_bitrates.ValidateCodecFormat());
// Accept min bitrate == max bitrate.
VideoCodec equal_bitrates = codec;
equal_bitrates.params[kCodecParamMinBitrate] = "100";
equal_bitrates.params[kCodecParamMaxBitrate] = "100";
EXPECT_TRUE(equal_bitrates.ValidateCodecFormat());
// Accept min bitrate < max bitrate.
VideoCodec different_bitrates = codec;
different_bitrates.params[kCodecParamMinBitrate] = "99";
different_bitrates.params[kCodecParamMaxBitrate] = "100";
EXPECT_TRUE(different_bitrates.ValidateCodecFormat());
}

2
talk/media/base/constants.cc
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@ -40,8 +40,6 @@ const float kLowSystemCpuThreshold = 0.65f;
const float kProcessCpuThreshold = 0.10f;
const char kRtxCodecName[] = "rtx";
const char kRedCodecName[] = "red";
const char kUlpfecCodecName[] = "ulpfec";
// RTP payload type is in the 0-127 range. Use 128 to indicate "all" payload
// types.

3
talk/media/base/constants.h
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@ -44,9 +44,6 @@ extern const float kLowSystemCpuThreshold;
extern const float kProcessCpuThreshold;
extern const char kRtxCodecName[];
extern const char kRedCodecName[];
extern const char kUlpfecCodecName[];
// Codec parameters
extern const int kWildcardPayloadType;

44
talk/media/webrtc/webrtcvideochannelfactory.h
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@ -1,44 +0,0 @@
/*
* libjingle
* Copyright 2004 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.
*/
#ifndef TALK_MEDIA_WEBRTC_WEBRTCVIDEOCHANNEL_H_
#define TALK_MEDIA_WEBRTC_WEBRTCVIDEOCHANNEL_H_
namespace cricket {
class VoiceMediaChannel;
class WebRtcVideoEngine2;
class WebRtcVideoChannel2;
class WebRtcVideoChannelFactory {
public:
virtual ~WebRtcVideoChannelFactory() {}
virtual WebRtcVideoChannel2* Create(WebRtcVideoEngine2* engine,
VoiceMediaChannel* voice_channel) = 0;
};
} // namespace cricket
#endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOCHANNEL_H_

1658
talk/media/webrtc/webrtcvideoengine2.cc
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332
talk/media/webrtc/webrtcvideoengine2.h
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@ -1,332 +0,0 @@
/*
* libjingle
* Copyright 2014 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.
*/
#ifndef TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_
#define TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_
#include <map>
#include <vector>
#include <string>
#include "talk/base/cpumonitor.h"
#include "talk/base/scoped_ptr.h"
#include "talk/media/base/mediaengine.h"
#include "talk/media/webrtc/webrtcvideochannelfactory.h"
#include "webrtc/common_video/interface/i420_video_frame.h"
#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/transport.h"
#include "webrtc/video_renderer.h"
#include "webrtc/video_send_stream.h"
namespace webrtc {
class Call;
class VideoCaptureModule;
class VideoDecoder;
class VideoEncoder;
class VideoRender;
class VideoSendStreamInput;
class VideoReceiveStream;
}
namespace talk_base {
class CpuMonitor;
class Thread;
} // namespace talk_base
namespace cricket {
class VideoCapturer;
class VideoFrame;
class VideoProcessor;
class VideoRenderer;
class VoiceMediaChannel;
class WebRtcVideoChannel2;
class WebRtcDecoderObserver;
class WebRtcEncoderObserver;
class WebRtcLocalStreamInfo;
class WebRtcRenderAdapter;
class WebRtcVideoChannelRecvInfo;
class WebRtcVideoChannelSendInfo;
class WebRtcVideoDecoderFactory;
class WebRtcVoiceEngine;
struct CapturedFrame;
struct Device;
class WebRtcVideoEngine2;
class WebRtcVideoChannel2;
class WebRtcVideoEncoderFactory2 {
public:
virtual bool CreateEncoderSettings(
webrtc::VideoSendStream::Config::EncoderSettings* encoder_settings,
const VideoOptions& options,
const cricket::VideoCodec& codec,
size_t num_streams) = 0;
virtual bool SupportsCodec(const cricket::VideoCodec& codec) = 0;
};
// WebRtcVideoEngine2 is used for the new native WebRTC Video API (webrtc:1667).
class WebRtcVideoEngine2 : public sigslot::has_slots<> {
public:
// Creates the WebRtcVideoEngine2 with internal VideoCaptureModule.
WebRtcVideoEngine2();
// Custom WebRtcVideoChannelFactory for testing purposes.
explicit WebRtcVideoEngine2(WebRtcVideoChannelFactory* channel_factory);
~WebRtcVideoEngine2();
// Basic video engine implementation.
bool Init(talk_base::Thread* worker_thread);
void Terminate();
int GetCapabilities();
bool SetOptions(const VideoOptions& options);
bool SetDefaultEncoderConfig(const VideoEncoderConfig& config);
VideoEncoderConfig GetDefaultEncoderConfig() const;
WebRtcVideoChannel2* CreateChannel(VoiceMediaChannel* voice_channel);
const std::vector<VideoCodec>& codecs() const;
const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
void SetLogging(int min_sev, const char* filter);
bool EnableTimedRender();
// No-op, never used.
bool SetLocalRenderer(VideoRenderer* renderer);
// This is currently ignored.
sigslot::repeater2<VideoCapturer*, CaptureState> SignalCaptureStateChange;
// Set the VoiceEngine for A/V sync. This can only be called before Init.
bool SetVoiceEngine(WebRtcVoiceEngine* voice_engine);
// Functions called by WebRtcVideoChannel2.
const VideoFormat& default_codec_format() const {
return default_codec_format_;
}
bool FindCodec(const VideoCodec& in);
bool CanSendCodec(const VideoCodec& in,
const VideoCodec& current,
VideoCodec* out);
// Check whether the supplied trace should be ignored.
bool ShouldIgnoreTrace(const std::string& trace);
VideoFormat GetStartCaptureFormat() const { return default_codec_format_; }
talk_base::CpuMonitor* cpu_monitor() { return cpu_monitor_.get(); }
virtual WebRtcVideoEncoderFactory2* GetDefaultVideoEncoderFactory() const;
private:
void Construct(WebRtcVideoChannelFactory* channel_factory,
WebRtcVoiceEngine* voice_engine,
talk_base::CpuMonitor* cpu_monitor);
talk_base::Thread* worker_thread_;
WebRtcVoiceEngine* voice_engine_;
std::vector<VideoCodec> video_codecs_;
std::vector<RtpHeaderExtension> rtp_header_extensions_;
VideoFormat default_codec_format_;
bool initialized_;
bool capture_started_;
// Critical section to protect the media processor register/unregister
// while processing a frame
talk_base::CriticalSection signal_media_critical_;
talk_base::scoped_ptr<talk_base::CpuMonitor> cpu_monitor_;
WebRtcVideoChannelFactory* channel_factory_;
};
// Adapter between webrtc::VideoRenderer and cricket::VideoRenderer.
// The webrtc::VideoRenderer is set once, whereas the cricket::VideoRenderer can
// be set after initialization. This adapter will also convert the incoming
// webrtc::I420VideoFrame to a frame type that cricket::VideoRenderer can
// render.
class WebRtcVideoRenderer : public webrtc::VideoRenderer {
public:
WebRtcVideoRenderer();
virtual void RenderFrame(const webrtc::I420VideoFrame& frame,
int time_to_render_ms) OVERRIDE;
void SetRenderer(cricket::VideoRenderer* renderer);
cricket::VideoRenderer* GetRenderer();
private:
void SetSize(int width, int height);
int last_width_;
int last_height_;
talk_base::CriticalSection lock_;
cricket::VideoRenderer* renderer_ GUARDED_BY(lock_);
};
class WebRtcVideoChannel2 : public talk_base::MessageHandler,
public VideoMediaChannel,
public webrtc::newapi::Transport {
public:
WebRtcVideoChannel2(WebRtcVideoEngine2* engine,
VoiceMediaChannel* voice_channel,
WebRtcVideoEncoderFactory2* encoder_factory);
// For testing purposes insert a pre-constructed call to verify that
// WebRtcVideoChannel2 calls the correct corresponding methods.
WebRtcVideoChannel2(webrtc::Call* call,
WebRtcVideoEngine2* engine,
WebRtcVideoEncoderFactory2* encoder_factory);
~WebRtcVideoChannel2();
bool Init();
// VideoMediaChannel implementation
virtual bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) OVERRIDE;
virtual bool SetSendCodecs(const std::vector<VideoCodec>& codecs) OVERRIDE;
virtual bool GetSendCodec(VideoCodec* send_codec) OVERRIDE;
virtual bool SetSendStreamFormat(uint32 ssrc,
const VideoFormat& format) OVERRIDE;
virtual bool SetRender(bool render) OVERRIDE;
virtual bool SetSend(bool send) OVERRIDE;
virtual bool AddSendStream(const StreamParams& sp) OVERRIDE;
virtual bool RemoveSendStream(uint32 ssrc) OVERRIDE;
virtual bool AddRecvStream(const StreamParams& sp) OVERRIDE;
virtual bool RemoveRecvStream(uint32 ssrc) OVERRIDE;
virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer) OVERRIDE;
virtual bool GetStats(const StatsOptions& options,
VideoMediaInfo* info) OVERRIDE;
virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer) OVERRIDE;
virtual bool SendIntraFrame() OVERRIDE;
virtual bool RequestIntraFrame() OVERRIDE;
virtual void OnPacketReceived(talk_base::Buffer* packet,
const talk_base::PacketTime& packet_time)
OVERRIDE;
virtual void OnRtcpReceived(talk_base::Buffer* packet,
const talk_base::PacketTime& packet_time)
OVERRIDE;
virtual void OnReadyToSend(bool ready) OVERRIDE;
virtual bool MuteStream(uint32 ssrc, bool mute) OVERRIDE;
virtual bool SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) OVERRIDE;
virtual bool SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) OVERRIDE;
virtual bool SetStartSendBandwidth(int bps) OVERRIDE;
virtual bool SetMaxSendBandwidth(int bps) OVERRIDE;
virtual bool SetOptions(const VideoOptions& options) OVERRIDE;
virtual bool GetOptions(VideoOptions* options) const OVERRIDE {
*options = options_;
return true;
}
virtual void SetInterface(NetworkInterface* iface) OVERRIDE;
virtual void UpdateAspectRatio(int ratio_w, int ratio_h) OVERRIDE;
virtual void OnMessage(talk_base::Message* msg) OVERRIDE;
// Implemented for VideoMediaChannelTest.
bool sending() const { return sending_; }
uint32 GetDefaultChannelSsrc() { return default_send_ssrc_; }
bool GetRenderer(uint32 ssrc, VideoRenderer** renderer);
private:
struct VideoCodecSettings {
VideoCodecSettings();
cricket::VideoCodec codec;
webrtc::FecConfig fec;
int rtx_payload_type;
};
class WebRtcVideoSendStream : public sigslot::has_slots<> {
public:
WebRtcVideoSendStream(webrtc::Call* call,
const webrtc::VideoSendStream::Config& config,
WebRtcVideoEncoderFactory2* encoder_factory);
~WebRtcVideoSendStream();
void SetCodec(const VideoOptions& options, const VideoCodecSettings& codec);
void InputFrame(VideoCapturer* capturer, const VideoFrame* frame);
bool SetCapturer(VideoCapturer* capturer);
bool SetVideoFormat(const VideoFormat& format);
bool MuteStream(bool mute);
bool DisconnectCapturer();
void Start();
void Stop();
private:
void RecreateWebRtcStream();
void SetDimensions(int width, int height);
webrtc::Call* const call_;
WebRtcVideoEncoderFactory2* const encoder_factory_;
talk_base::CriticalSection lock_;
webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
webrtc::VideoSendStream::Config config_ GUARDED_BY(lock_);
VideoCapturer* capturer_ GUARDED_BY(lock_);
bool sending_ GUARDED_BY(lock_);
bool muted_ GUARDED_BY(lock_);
VideoFormat format_ GUARDED_BY(lock_);
talk_base::CriticalSection frame_lock_;
webrtc::I420VideoFrame video_frame_ GUARDED_BY(frame_lock_);
};
void Construct(webrtc::Call* call, WebRtcVideoEngine2* engine);
virtual bool SendRtp(const uint8_t* data, size_t len) OVERRIDE;
virtual bool SendRtcp(const uint8_t* data, size_t len) OVERRIDE;
void StartAllSendStreams();
void StopAllSendStreams();
void SetCodecForAllSendStreams(const VideoCodecSettings& codec);
static std::vector<VideoCodecSettings> MapCodecs(
const std::vector<VideoCodec>& codecs);
std::vector<VideoCodecSettings> FilterSupportedCodecs(
const std::vector<VideoCodecSettings>& mapped_codecs);
uint32_t rtcp_receiver_report_ssrc_;
bool sending_;
talk_base::scoped_ptr<webrtc::Call> call_;
std::map<uint32, WebRtcVideoRenderer*> renderers_;
VideoRenderer* default_renderer_;
uint32_t default_send_ssrc_;
uint32_t default_recv_ssrc_;
// Using primary-ssrc (first ssrc) as key.
std::map<uint32, WebRtcVideoSendStream*> send_streams_;
std::map<uint32, webrtc::VideoReceiveStream*> receive_streams_;
Settable<VideoCodecSettings> send_codec_;
WebRtcVideoEncoderFactory2* const encoder_factory_;
std::vector<VideoCodecSettings> recv_codecs_;
VideoOptions options_;
};
} // namespace cricket
#endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_

1187
talk/media/webrtc/webrtcvideoengine2_unittest.cc
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