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gstDecoder.cpp
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/*
* Copyright (c) 2018, NVIDIA CORPORATION. All rights reserved.
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*/
#include "gstDecoder.h"
#include "gstWebRTC.h"
#include "cudaColorspace.h"
#include "filesystem.h"
#include "logging.h"
#include <gst/app/gstappsink.h>
#include <gst/pbutils/pbutils.h>
#include <sstream>
#include <unistd.h>
#include <string.h>
#include <strings.h>
//
// RTP test source pipeline:
// (from PC) $ gst-launch-1.0 -v videotestsrc ! video/x-raw,width=300,height=300,framerate=30/1 ! x264enc ! rtph264pay ! udpsink host=127.0.0.1 port=5000
// (from Jetson) $ ./video-viewer test.mkv rtp://@:5000
//
// RTP test recieve pipeline:
// (from PC) $ gst-launch-1.0 -v udpsrc port=5000 caps = "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, payload=(int)96" ! rtph264depay ! decodebin ! videoconvert ! autovideosink
// (from Jetson) $ ./video-viewer rtp://@:5000
// (from VLC) SDP file contents:
// c=IN IP4 127.0.0.1
// m=video 5000 RTP/AVP 96
// a=rtpmap:96 H264/90000
//
// RSTP test server installation:
// $ git clone https://github.com/GStreamer/gst-rtsp-server && cd gst-rtsp-server
// $ git checkout 1.14.5
// $ ./autogen.sh --noconfigure && ./configure && make
// $ cd examples && ./test-launch "( videotestsrc ! x264enc ! rtph264pay name=pay0 pt=96 )"
// > rtsp://127.0.0.1:8554/test
//
// RTSP authentication test:
// $ ./test-auth
// > rtsp://user:[email protected]:8554/test
// > rtsp://admin:[email protected]:8554/test
//
// supported image file extensions
const char* gstDecoder::SupportedExtensions[] = { "mkv", "mp4", "qt",
"flv", "avi", "h264",
"h265", "mov", "webm", NULL };
bool gstDecoder::IsSupportedExtension( const char* ext )
{
if( !ext )
return false;
uint32_t extCount = 0;
while(true)
{
if( !SupportedExtensions[extCount] )
break;
if( strcasecmp(SupportedExtensions[extCount], ext) == 0 )
return true;
extCount++;
}
return false;
}
// constructor
gstDecoder::gstDecoder( const videoOptions& options ) : videoSource(options)
{
mAppSink = NULL;
mBus = NULL;
mPipeline = NULL;
mCustomSize = false;
mCustomRate = false;
mEOS = false;
mLoopCount = 1;
mBufferManager = new gstBufferManager(&mOptions);
mWebRTCServer = NULL;
mWebRTCConnected = false;
}
// destructor
gstDecoder::~gstDecoder()
{
Close();
if( mWebRTCServer != NULL )
{
mWebRTCServer->Release();
mWebRTCServer = NULL;
}
destroyPipeline();
SAFE_DELETE(mBufferManager);
}
// destroyPipeline
void gstDecoder::destroyPipeline()
{
if( mAppSink != NULL )
{
gst_object_unref(mAppSink);
mAppSink = NULL;
}
if( mBus != NULL )
{
gst_object_unref(mBus);
mBus = NULL;
}
if( mPipeline != NULL )
{
gst_element_set_state(mPipeline, GST_STATE_NULL);
gst_object_unref(mPipeline);
mPipeline = NULL;
}
mEOS = false;
mStreaming = false;
mWebRTCConnected = false;
}
// Create
gstDecoder* gstDecoder::Create( const videoOptions& options )
{
gstDecoder* dec = new gstDecoder(options);
if( !dec )
return NULL;
if( !dec->init() )
{
LogError(LOG_GSTREAMER "gstDecoder -- failed to create decoder for %s\n", dec->mOptions.resource.string.c_str());
return NULL;
}
return dec;
}
// Create
gstDecoder* gstDecoder::Create( const URI& resource, videoOptions::Codec codec )
{
videoOptions opt;
opt.resource = resource;
opt.codec = codec;
opt.ioType = videoOptions::INPUT;
return Create(opt);
}
// init
bool gstDecoder::init()
{
if( !gstreamerInit() )
{
LogError(LOG_GSTREAMER "failed to initialize gstreamer API\n");
return NULL;
}
// first, check that the file exists
if( mOptions.resource.protocol == "file" )
{
if( !fileExists(mOptions.resource.location) )
{
LogError(LOG_GSTREAMER "gstDecoder -- couldn't find file '%s'\n", mOptions.resource.location.c_str());
return false;
}
}
LogInfo(LOG_GSTREAMER "gstDecoder -- creating decoder for %s\n", mOptions.resource.location.c_str());
// flag if the user wants a specific resolution and framerate
if( mOptions.width != 0 || mOptions.height != 0 )
mCustomSize = true;
if( mOptions.frameRate != 0 )
mCustomRate = true;
// discover resource stats
if( !discover() )
{
if( mOptions.resource.protocol == "rtp" || mOptions.resource.protocol == "webrtc" )
{
LogWarning(LOG_GSTREAMER "gstDecoder -- resource discovery not supported for RTP/WebRTC streams\n");
if( mOptions.codec == videoOptions::CODEC_UNKNOWN )
{
LogWarning(LOG_GSTREAMER "gstDecoder -- defaulting to H264 codec (you can change this with the --input-codec option)\n");
mOptions.codec = videoOptions::CODEC_H264;
}
}
else
LogError(LOG_GSTREAMER "gstDecoder -- resource discovery and auto-negotiation failed\n");
if( mOptions.codec == videoOptions::CODEC_UNKNOWN )
{
LogError(LOG_GSTREAMER "gstDecoder -- try manually setting the codec with the --input-codec option\n");
return false;
}
}
// build pipeline string
if( !buildLaunchStr() )
{
LogError(LOG_GSTREAMER "gstDecoder -- failed to build pipeline string\n");
return false;
}
// create pipeline
if( !initPipeline() )
{
LogError(LOG_GSTREAMER "failed to create decoder pipeline\n");
return false;
}
// create server for WebRTC streams
if( mOptions.resource.protocol == "webrtc" )
{
// create WebRTC server
mWebRTCServer = WebRTCServer::Create(mOptions.resource.port, mOptions.stunServer.c_str(),
mOptions.sslCert.c_str(), mOptions.sslKey.c_str());
if( !mWebRTCServer )
return false;
mWebRTCServer->AddRoute(mOptions.resource.path.c_str(), onWebsocketMessage, this, WEBRTC_VIDEO|WEBRTC_RECEIVE|WEBRTC_PUBLIC);
}
return true;
}
// initPipeline
bool gstDecoder::initPipeline()
{
GError* err = NULL;
mPipeline = gst_parse_launch(mLaunchStr.c_str(), &err);
if( err != NULL )
{
LogError(LOG_GSTREAMER "gstDecoder -- failed to create pipeline\n");
LogError(LOG_GSTREAMER " (%s)\n", err->message);
g_error_free(err);
return false;
}
GstPipeline* pipeline = GST_PIPELINE(mPipeline);
if( !pipeline )
{
LogError(LOG_GSTREAMER "gstDecoder -- failed to cast GstElement into GstPipeline\n");
return false;
}
// retrieve pipeline bus
mBus = gst_pipeline_get_bus(pipeline);
if( !mBus )
{
LogError(LOG_GSTREAMER "gstDecoder -- failed to retrieve GstBus from pipeline\n");
return false;
}
// add watch for messages (disabled when we poll the bus ourselves, instead of gmainloop)
//gst_bus_add_watch(mBus, (GstBusFunc)gst_message_print, NULL);
// get the appsrc
GstElement* appsinkElement = gst_bin_get_by_name(GST_BIN(pipeline), "mysink");
GstAppSink* appsink = GST_APP_SINK(appsinkElement);
if( !appsinkElement || !appsink)
{
LogError(LOG_GSTREAMER "gstDecoder -- failed to retrieve AppSink element from pipeline\n");
return false;
}
mAppSink = appsink;
// setup callbacks
GstAppSinkCallbacks cb;
memset(&cb, 0, sizeof(GstAppSinkCallbacks));
cb.eos = onEOS;
cb.new_preroll = onPreroll; // disabled b/c preroll sometimes occurs during Close() and crashes
#if GST_CHECK_VERSION(1,0,0)
cb.new_sample = onBuffer;
#else
cb.new_buffer = onBuffer;
#endif
gst_app_sink_set_callbacks(mAppSink, &cb, (void*)this, NULL);
return true;
}
// findVideoStreamInfo
static GstDiscovererVideoInfo* findVideoStreamInfo( GstDiscovererStreamInfo* info )
{
if( !info )
return NULL;
//printf("stream type -- %s\n", gst_discoverer_stream_info_get_stream_type_nick(info));
if( GST_IS_DISCOVERER_VIDEO_INFO(info) )
{
return GST_DISCOVERER_VIDEO_INFO(info);
}
else if( GST_IS_DISCOVERER_CONTAINER_INFO(info) )
{
GstDiscovererContainerInfo* containerInfo = GST_DISCOVERER_CONTAINER_INFO(info);
if( !containerInfo )
return NULL;
GList* containerStreams = gst_discoverer_container_info_get_streams(containerInfo);
for( GList* n=containerStreams; n; n = n->next )
{
GstDiscovererVideoInfo* videoStream = findVideoStreamInfo(GST_DISCOVERER_STREAM_INFO(n->data));
if( videoStream != NULL )
return videoStream;
}
}
return findVideoStreamInfo(gst_discoverer_stream_info_get_next(info));
}
// discover
bool gstDecoder::discover()
{
// RTP streams and WebRTC connections can't be discovered
if( mOptions.resource.protocol == "rtp" || mOptions.resource.protocol == "webrtc" )
return false;
// create a new discovery interface
GError* err = NULL;
GstDiscoverer* discoverer = gst_discoverer_new(5 * GST_SECOND, &err);
if( !discoverer )
{
LogError(LOG_GSTREAMER "gstDecoder -- failed to create gstreamer discovery instance: %s\n", err->message);
return false;
}
GstDiscovererInfo* info = gst_discoverer_discover_uri(discoverer,
mOptions.resource.string.c_str(), &err);
if( !info || err != NULL )
{
LogError(LOG_GSTREAMER "gstDecoder -- %s\n", err->message);
return false;
}
GstDiscovererStreamInfo* rootStream = gst_discoverer_info_get_stream_info(info);
if( !rootStream )
{
LogError(LOG_GSTREAMER "gstDecoder -- failed to discover stream info\n");
return false;
}
GstDiscovererVideoInfo* videoInfo = findVideoStreamInfo(rootStream);
GstDiscovererStreamInfo* streamInfo = GST_DISCOVERER_STREAM_INFO(videoInfo);
if( !videoInfo )
{
LogError(LOG_GSTREAMER "gstDecoder -- failed to discover any video streams\n");
return false;
}
// retrieve video resolution
guint width = gst_discoverer_video_info_get_width(videoInfo);
guint height = gst_discoverer_video_info_get_height(videoInfo);
if( mOptions.flipMethod == videoOptions::FLIP_CLOCKWISE || mOptions.flipMethod == videoOptions::FLIP_COUNTERCLOCKWISE
|| mOptions.flipMethod == videoOptions::FLIP_UPPER_LEFT_DIAGONAL || mOptions.flipMethod == videoOptions::FLIP_UPPER_RIGHT_DIAGONAL )
{
const guint prevWidth = width;
width = height;
height = prevWidth;
}
const float framerate_num = gst_discoverer_video_info_get_framerate_num(videoInfo);
const float framerate_denom = gst_discoverer_video_info_get_framerate_denom(videoInfo);
const float framerate = framerate_num / framerate_denom;
LogVerbose(LOG_GSTREAMER "gstDecoder -- discovered video resolution: %ux%u (framerate %f Hz)\n", width, height, framerate);
// disable re-scaling if the user's custom size matches the feed's
if( mCustomSize && mOptions.width == width && mOptions.height == height )
mCustomSize = false;
if( mOptions.width == 0 )
mOptions.width = width;
if( mOptions.height == 0 )
mOptions.height = height;
// confirm the desired framerate against what the stream provides
if( mCustomRate )
{
// disable rate-limiting if the user's custom rate matches the feed's
if( mOptions.frameRate == framerate )
mCustomRate = false;
}
else
{
// otherwise adopt the feed's framerate
mOptions.frameRate = framerate;
}
// retrieve video caps
GstCaps* caps = gst_discoverer_stream_info_get_caps(streamInfo);
if( !caps )
{
LogError(LOG_GSTREAMER "gstDecoder -- failed to discover video caps\n");
return false;
}
const std::string videoCaps = gst_caps_to_string(caps);
LogVerbose(LOG_GSTREAMER "gstDecoder -- discovered video caps: %s\n", videoCaps.c_str());
// parse codec
if( videoCaps.find("video/x-h264") != std::string::npos )
mOptions.codec = videoOptions::CODEC_H264;
else if( videoCaps.find("video/x-h265") != std::string::npos )
mOptions.codec = videoOptions::CODEC_H265;
else if( videoCaps.find("video/x-vp8") != std::string::npos )
mOptions.codec = videoOptions::CODEC_VP8;
else if( videoCaps.find("video/x-vp9") != std::string::npos )
mOptions.codec = videoOptions::CODEC_VP9;
else if( videoCaps.find("image/jpeg") != std::string::npos )
mOptions.codec = videoOptions::CODEC_MJPEG;
else if( videoCaps.find("video/mpeg") != std::string::npos )
{
if( videoCaps.find("mpegversion=(int)4") != std::string::npos )
mOptions.codec = videoOptions::CODEC_MPEG4;
else if( videoCaps.find("mpegversion=(int)2") != std::string::npos )
mOptions.codec = videoOptions::CODEC_MPEG2;
}
if( mOptions.codec == videoOptions::CODEC_UNKNOWN )
{
LogError(LOG_GSTREAMER "gstDecoder -- unsupported codec, supported codecs are:\n");
LogError(LOG_GSTREAMER " * h264\n");
LogError(LOG_GSTREAMER " * h265\n");
LogError(LOG_GSTREAMER " * vp8\n");
LogError(LOG_GSTREAMER " * vp9\n");
LogError(LOG_GSTREAMER " * mpeg2\n");
LogError(LOG_GSTREAMER " * mpeg4\n");
LogError(LOG_GSTREAMER " * mjpeg\n");
return false;
}
// TODO free other resources
//g_free(discoverer);
return true;
}
// buildLaunchStr
bool gstDecoder::buildLaunchStr()
{
std::ostringstream ss;
#if defined(ENABLE_NVMM)
const bool enable_nvmm = true;
#else
const bool enable_nvmm = false;
#endif
// select the decoder
const char* decoder = gst_select_decoder(mOptions.codec, mOptions.codecType);
if( !decoder )
{
LogError(LOG_GSTREAMER "gstDecoder -- unsupported codec requested (%s)\n", videoOptions::CodecToStr(mOptions.codec));
LogError(LOG_GSTREAMER " supported decoder codecs are:\n");
LogError(LOG_GSTREAMER " * h264\n");
LogError(LOG_GSTREAMER " * h265\n");
LogError(LOG_GSTREAMER " * vp8\n");
LogError(LOG_GSTREAMER " * vp9\n");
LogError(LOG_GSTREAMER " * mpeg2\n");
LogError(LOG_GSTREAMER " * mpeg4\n");
LogError(LOG_GSTREAMER " * mjpeg\n");
return false;
}
// select the parser (if needed)
const char* parser = "";
if( mOptions.codec == videoOptions::CODEC_H264 )
parser = "h264parse ! ";
else if( mOptions.codec == videoOptions::CODEC_H265 )
parser = "h265parse ! ";
else if( mOptions.codec == videoOptions::CODEC_MPEG2 )
parser = "mpegvideoparse ! ";
else if( mOptions.codec == videoOptions::CODEC_MPEG4 )
parser = "mpeg4videoparse ! ";
// determine the requested protocol to use
const URI& uri = GetResource();
if( uri.protocol == "file" )
{
mOptions.deviceType = videoOptions::DEVICE_FILE;
ss << "filesrc location=" << mOptions.resource.location << " ! ";
if( uri.extension == "mkv" || uri.extension == "webm" )
ss << "matroskademux ! ";
else if( uri.extension == "mp4" || uri.extension == "qt" || uri.extension == "mov" )
ss << "qtdemux ! ";
else if( uri.extension == "flv" )
ss << "flvdemux ! ";
else if( uri.extension == "avi" )
ss << "avidemux ! ";
else if( uri.extension != "h264" && uri.extension != "h265" )
{
LogError(LOG_GSTREAMER "gstDecoder -- unsupported video file extension (%s)\n", uri.extension.c_str());
LogError(LOG_GSTREAMER " supported video extensions are:\n");
LogError(LOG_GSTREAMER " * mkv, webm\n");
LogError(LOG_GSTREAMER " * mp4, qt, mov\n");
LogError(LOG_GSTREAMER " * flv\n");
LogError(LOG_GSTREAMER " * avi\n");
LogError(LOG_GSTREAMER " * h264, h265\n");
return false;
}
ss << "queue ! " << parser;
}
else if( uri.protocol == "rtp" )
{
mOptions.deviceType = videoOptions::DEVICE_IP;
if( uri.port <= 0 )
{
LogError(LOG_GSTREAMER "gstDecoder -- invalid RTP port (%i)\n", uri.port);
return false;
}
ss << "udpsrc port=" << uri.port;
ss << " multicast-group=" << uri.location << " auto-multicast=true";
ss << " caps=\"" << "application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)";
if( mOptions.codec == videoOptions::CODEC_H264 )
ss << "H264\" ! rtph264depay ! ";
else if( mOptions.codec == videoOptions::CODEC_H265 )
ss << "H265\" ! rtph265depay ! ";
else if( mOptions.codec == videoOptions::CODEC_VP8 )
ss << "VP8\" ! rtpvp8depay ! ";
else if( mOptions.codec == videoOptions::CODEC_VP9 )
ss << "VP9\" ! rtpvp9depay ! ";
else if( mOptions.codec == videoOptions::CODEC_MPEG2 )
ss << "MP2T\" ! rtpmp2tdepay ! "; // MP2T-ES
else if( mOptions.codec == videoOptions::CODEC_MPEG4 )
ss << "MP4V-ES\" ! rtpmp4vdepay ! "; // MPEG4-GENERIC\" ! rtpmp4gdepay
else if( mOptions.codec == videoOptions::CODEC_MJPEG )
ss << "JPEG\" ! rtpjpegdepay ! ";
if( mOptions.codecType != videoOptions::CODEC_V4L2 )
ss << parser;
}
else if( uri.protocol == "rtsp" || uri.protocol == "webrtc" )
{
mOptions.deviceType = videoOptions::DEVICE_IP;
if( uri.protocol == "rtsp" )
{
ss << "rtspsrc location=" << uri.string;
ss << " latency=" << mOptions.latency;
ss << " ! queue ! ";
}
else
{
ss << "webrtcbin name=webrtcbin ";
ss << "stun-server=stun://" << WEBRTC_DEFAULT_STUN_SERVER;
ss << " ! queue ! ";
}
if( mOptions.codec == videoOptions::CODEC_H264 )
ss << "rtph264depay ! ";
else if( mOptions.codec == videoOptions::CODEC_H265 )
ss << "rtph265depay ! ";
else if( mOptions.codec == videoOptions::CODEC_VP8 )
ss << "rtpvp8depay ! ";
else if( mOptions.codec == videoOptions::CODEC_VP9 )
ss << "rtpvp9depay ! ";
else if( mOptions.codec == videoOptions::CODEC_MPEG2 )
ss << "rtpmp2tdepay ! "; // MP2T-ES
else if( mOptions.codec == videoOptions::CODEC_MPEG4 )
ss << "rtpmp4vdepay ! "; // rtpmp4gdepay
else if( mOptions.codec == videoOptions::CODEC_MJPEG )
ss << "rtpjpegdepay ! ";
if( mOptions.codecType != videoOptions::CODEC_V4L2 )
ss << parser;
}
else
{
LogError(LOG_GSTREAMER "gstDecoder -- unsupported protocol (%s)\n", uri.protocol.c_str());
LogError(LOG_GSTREAMER " supported protocols are:\n");
LogError(LOG_GSTREAMER " * file://\n");
LogError(LOG_GSTREAMER " * rtp://\n");
LogError(LOG_GSTREAMER " * rtsp://\n");
LogError(LOG_GSTREAMER " * webrtc://\n");
return false;
}
if( mOptions.save.path.length() > 0 )
{
ss << "tee name=savetee savetee. ! queue ! ";
if( !gst_build_filesink(mOptions.save, mOptions.codec, ss) )
return false;
ss << "savetee. ! queue ! ";
}
// add the decoder
ss << decoder << " name=decoder ";
if( mOptions.codecType == videoOptions::CODEC_V4L2 && mOptions.codec != videoOptions::CODEC_MJPEG )
ss << "enable-max-performance=1 ";
ss << "! ";
// resize if requested
if( mCustomSize || mOptions.flipMethod != videoOptions::FLIP_NONE )
{
#if defined(__aarch64__)
ss << "nvvidconv name=vidconv";
if( mOptions.flipMethod != videoOptions::FLIP_NONE )
ss << " flip-method=" << (int)mOptions.flipMethod;
ss << " ! video/x-raw";
#elif defined(__x86_64__) || defined(__amd64__)
if( mOptions.flipMethod != videoOptions::FLIP_NONE )
ss << "videoflip method=" << videoOptions::FlipMethodToStr(mOptions.flipMethod) << " ! ";
if( mOptions.width != 0 && mOptions.height != 0 )
ss << "videoscale ! ";
ss << "video/x-raw";
#endif
if( enable_nvmm )
ss << "(" << GST_CAPS_FEATURE_MEMORY_NVMM << ")";
if( mOptions.width != 0 && mOptions.height != 0 )
ss << ", width=(int)" << mOptions.width << ", height=(int)" << mOptions.height << ", format=(string)NV12";
ss <<" ! ";
}
else
{
ss << "video/x-raw";
if( enable_nvmm || mOptions.codecType == videoOptions::CODEC_V4L2 )
{
// add NVMM caps when requested, or if using V4L2 codecs
ss << "(" << GST_CAPS_FEATURE_MEMORY_NVMM << ")";
// V4L2 codecs only output NVMM memory
// so if NVMM is disabled, put it through nvvidconv first
if( !enable_nvmm )
ss << " ! nvvidconv name=vidconv ! video/x-raw";
}
ss << " ! ";
}
// rate-limit if requested
if( mCustomRate )
ss << "videorate drop-only=true max-rate=" << (int)mOptions.frameRate << " ! ";
// add the app sink
ss << "appsink name=mysink";
if( uri.protocol != "file" )
ss << " sync=false"; // wait-on-eos=false; // this can improve realtime network streaming, but also causes videos to playback as fast as possible
mLaunchStr = ss.str();
LogInfo(LOG_GSTREAMER "gstDecoder -- pipeline string:\n");
LogInfo(LOG_GSTREAMER "%s\n", mLaunchStr.c_str());
return true;
}
// onEOS
void gstDecoder::onEOS( _GstAppSink* sink, void* user_data )
{
LogWarning(LOG_GSTREAMER "gstDecoder -- end of stream (EOS)\n");
if( !user_data )
return;
gstDecoder* dec = (gstDecoder*)user_data;
dec->mEOS = true;
dec->mStreaming = dec->isLooping();
}
// onPreroll
GstFlowReturn gstDecoder::onPreroll( _GstAppSink* sink, void* user_data )
{
LogVerbose(LOG_GSTREAMER "gstDecoder -- onPreroll()\n");
if( !user_data )
return GST_FLOW_OK;
gstDecoder* dec = (gstDecoder*)user_data;
#if GST_CHECK_VERSION(1,0,0)
// onPreroll gets called sometimes, just pull and free the buffer
// otherwise the pipeline may hang during shutdown
GstSample* gstSample = gst_app_sink_pull_preroll(dec->mAppSink);
if( !gstSample )
{
LogError(LOG_GSTREAMER "gstDecoder -- app_sink_pull_sample() returned NULL...\n");
return GST_FLOW_OK;
}
gst_sample_unref(gstSample);
#endif
dec->checkMsgBus();
return GST_FLOW_OK;
}
// onBuffer
GstFlowReturn gstDecoder::onBuffer( _GstAppSink* sink, void* user_data )
{
//printf(LOG_GSTREAMER "gstDecoder -- onBuffer()\n");
if( !user_data )
return GST_FLOW_OK;
gstDecoder* dec = (gstDecoder*)user_data;
dec->checkBuffer();
dec->checkMsgBus();
return GST_FLOW_OK;
}
#if GST_CHECK_VERSION(1,0,0)
#define release_return { gst_sample_unref(gstSample); return; }
#else
#define release_return { gst_buffer_unref(gstBuffer); return; }
#endif
// checkBuffer
void gstDecoder::checkBuffer()
{
if( !mAppSink )
return;
#if GST_CHECK_VERSION(1,0,0)
// block waiting for the sample
GstSample* gstSample = gst_app_sink_pull_sample(mAppSink);
if( !gstSample )
{
LogError(LOG_GSTREAMER "gstDecoder -- app_sink_pull_sample() returned NULL...\n");
return;
}
// retrieve sample caps
GstCaps* gstCaps = gst_sample_get_caps(gstSample);
if( !gstCaps )
{
LogError(LOG_GSTREAMER "gstDecoder -- gst_sample had NULL caps...\n");
release_return;
}
// retrieve the buffer from the sample
GstBuffer* gstBuffer = gst_sample_get_buffer(gstSample);
if( !gstBuffer )
{
LogError(LOG_GSTREAMER "gstDecoder -- app_sink_pull_sample() returned NULL...\n");
release_return;
}
#else
// block waiting for the buffer
GstBuffer* gstBuffer = gst_app_sink_pull_buffer(mAppSink);
if( !gstBuffer )
{
LogError(LOG_GSTREAMER "gstDecoder -- app_sink_pull_buffer() returned NULL...\n");
return;
}
// retrieve caps
GstCaps* gstCaps = gst_buffer_get_caps(gstBuffer);
if( !gstCaps )
{
LogError(LOG_GSTREAMER "gstDecoder -- gst_buffer had NULL caps...\n");
release_return;
}
#endif
const uint32_t width = mOptions.width;
const uint32_t height = mOptions.height;
// enqueue the buffer for color conversion
if( !mBufferManager->Enqueue(gstBuffer, gstCaps) )
{
LogError(LOG_GSTREAMER "gstDecoder -- failed to handle incoming buffer\n");
release_return;
}
if( (width > 0 && width != mOptions.width) || (height > 0 && height != mOptions.height) )
{
LogWarning(LOG_GSTREAMER "gstDecoder -- resolution changing from (%ux%u) to (%ux%u)\n", width, height, mOptions.width, mOptions.height);
// nvbufsurface: NvBufSurfaceCopy: buffer param mismatch
GstElement* vidconv = gst_bin_get_by_name(GST_BIN(mPipeline), "vidconv");
if( vidconv != NULL )
{
gst_element_set_state(vidconv, GST_STATE_NULL);
gst_element_set_state(vidconv, GST_STATE_PLAYING);
gst_object_unref(vidconv);
}
}
mOptions.frameCount++;
release_return;
}
#define RETURN_STATUS(code) { if( status != NULL ) { *status=(code); } return ((code) == videoSource::OK ? true : false); }
// Capture
bool gstDecoder::Capture( void** output, imageFormat format, uint64_t timeout, int* status, cudaStream_t stream )
{
// update the webrtc server if needed
if( mWebRTCServer != NULL && !mWebRTCServer->IsThreaded() )
mWebRTCServer->ProcessRequests();
// verify the output pointer exists
if( !output )
RETURN_STATUS(ERROR);
// confirm the stream is open
if( !mStreaming || mEOS )
{
if( !Open() )
RETURN_STATUS(mEOS ? EOS : ERROR);
}
// wait until a new frame is recieved
const int result = mBufferManager->Dequeue(output, format, timeout, stream);
if( result < 0 )
{
LogError(LOG_GSTREAMER "gstDecoder::Capture() -- an error occurred retrieving the next image buffer\n");
RETURN_STATUS(ERROR);
}
else if( result == 0 )
{
LogWarning(LOG_GSTREAMER "gstDecoder::Capture() -- a timeout occurred waiting for the next image buffer\n");
RETURN_STATUS(TIMEOUT);
}
mLastTimestamp = mBufferManager->GetLastTimestamp();
mRawFormat = mBufferManager->GetRawFormat();
RETURN_STATUS(OK);
}
#if 0
static void queryPipelineState( GstElement* pipeline )
{
GstState state = GST_STATE_VOID_PENDING;
GstState pending = GST_STATE_VOID_PENDING;
GstStateChangeReturn result = gst_element_get_state (pipeline,
&state, &pending, GST_CLOCK_TIME_NONE);
if( result == GST_STATE_CHANGE_FAILURE )
printf("GST_STATE_CHANGE_FAILURE\n");
printf("state - %s\n", gst_element_state_get_name(state));
printf("pending - %s\n", gst_element_state_get_name(pending));
}
#endif
// Open
bool gstDecoder::Open()
{
if( mEOS )
{
if( isLooping() )
{
// seek stream back to the beginning
GstEvent *seek_event = NULL;
const bool seek = gst_element_seek(mPipeline, 1.0, GST_FORMAT_TIME,
(GstSeekFlags)(GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT),
GST_SEEK_TYPE_SET, 0LL,
GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE );
if( !seek )
{
LogError(LOG_GSTREAMER "gstDecoder -- failed to seek stream to beginning (loop %zu of %i)\n", mLoopCount+1, mOptions.loop);
return false;
}
LogWarning(LOG_GSTREAMER "gstDecoder -- seeking stream to beginning (loop %zu of %i)\n", mLoopCount+1, mOptions.loop);
mLoopCount++;
mEOS = false;
}
else
{
LogWarning(LOG_GSTREAMER "gstDecoder -- end of stream (EOS) has been reached, stream has been closed\n");
return false;
}
}
if( mStreaming || (mWebRTCServer != NULL && !mWebRTCConnected) ) // with WebRTC, don't start the pipeline until peer connected
return true;
// transition pipline to STATE_PLAYING
LogInfo(LOG_GSTREAMER "opening gstDecoder for streaming, transitioning pipeline to GST_STATE_PLAYING\n");
const GstStateChangeReturn result = gst_element_set_state(mPipeline, GST_STATE_PLAYING);
if( result == GST_STATE_CHANGE_ASYNC )
{
#if 0
GstMessage* asyncMsg = gst_bus_timed_pop_filtered(mBus, 5 * GST_SECOND,
(GstMessageType)(GST_MESSAGE_ASYNC_DONE|GST_MESSAGE_ERROR));
if( asyncMsg != NULL )
{
gst_message_print(mBus, asyncMsg, this);
gst_message_unref(asyncMsg);
}
else
printf(LOG_GSTREAMER "gstDecoder -- NULL message after transitioning pipeline to PLAYING...\n");
#endif