OpenHarmony AVPlayer扩展支持rtsp协议:编译gstreamer插件库(二)
上篇传送门:OpenHarmony AVPlayer扩展支持rtsp协议:编译gstreamer插件库(一) 五、插件库 1、gstrtsprtsp插件库,库名必须为libgstrtsp.z.so,不可修改third_party/gstreamer/gstplugins_good/BUILD.gn 增加 ohos_
上篇传送门:OpenHarmony AVPlayer扩展支持rtsp协议:编译gstreamer插件库(一)
五、插件库
1、gstrtsp
rtsp插件库,库名必须为libgstrtsp.z.so,不可修改
third_party/gstreamer/gstplugins_good/BUILD.gn 增加
ohos_source_set("gstrtsp_source") {
sources = [
"gst/rtsp/xxx.c",
...
]
configs = [ ":gst_plugins_config" ]
}
ohos_shared_library("gstrtsp") {
deps = [
":gstrtsp_source",
"//third_party/glib:glib",
"//third_party/glib:gobject",
"//third_party/glib:gio",
"//third_party/gstreamer/gstreamer:gstbase",
"//third_party/gstreamer/gstreamer:gstreamer",
"//third_party/gstreamer/gstplugins_base:gstrtsp-1.0",
"//third_party/gstreamer/gstplugins_base:gstrtp-1.0",
"//third_party/gstreamer/gstplugins_base:gstsdp",
]
relative_install_dir = "media/plugins"
part_name = "gstreamer"
subsystem_name = "thirdparty"
}
编译生成libgstrtsp.z.so,安装到/system/lib/media/plugins 目录
2、gstrtpmanager
rtpmanager插件库,库名必须为libgstrtpmanager.z.so,不可修改
third_party/gstreamer/gstplugins_good/BUILD.gn 增加
ohos_source_set("gstrtpmanager_source") {
sources = [
"gst/rtpmanager/xxx.c",
...
]
configs = [ ":gst_plugins_config" ]
}
ohos_shared_library("gstrtpmanager") {
deps = [
":gstrtpmanager_source",
"//third_party/glib:glib",
"//third_party/glib:gobject",
"//third_party/glib:gio",
"//third_party/gstreamer/gstreamer:gstbase",
"//third_party/gstreamer/gstreamer:gstreamer",
"//third_party/gstreamer/gstplugins_base:gstaudio",
"//third_party/gstreamer/gstplugins_base:gstrtp-1.0",
]
relative_install_dir = "media/plugins"
part_name = "gstreamer"
subsystem_name = "thirdparty"
}
编译生成libgstrtpmanager.z.so,安装到/system/lib/media/plugins 目录
3、gstrtp
rtp插件库,库名必须为libgstrtp.z.so,不可修改
third_party/gstreamer/gstplugins_good/BUILD.gn 增加
ohos_source_set("gstrtp_source") {
sources = [
"gst/rtp/xxx.c",
...
]
configs = [ ":gst_plugins_config" ]
}
ohos_shared_library("gstrtp") {
deps = [
":gstrtp_source",
"//third_party/glib:glib",
"//third_party/glib:gobject",
"//third_party/gstreamer/gstreamer:gstbase",
"//third_party/gstreamer/gstreamer:gstreamer",
"//third_party/gstreamer/gstplugins_base:gstaudio",
"//third_party/gstreamer/gstplugins_base:gstvideo",
"//third_party/gstreamer/gstplugins_base:gsttag",
"//third_party/gstreamer/gstplugins_base:gstrtp-1.0",
"//third_party/gstreamer/gstplugins_base:gstpbutils",
]
relative_install_dir = "media/plugins"
part_name = "gstreamer"
subsystem_name = "thirdparty"
}
编译生成libgstrtp.z.so,安装到/system/lib/media/plugins 目录
4、gstudp
如果rtp底层传输协议使用的是udp,需要编译该库
udp插件库,库名必须为libgstudp.z.so,不可修改
third_party/gstreamer/gstplugins_good/BUILD.gn 增加
ohos_source_set("gstudp_source") {
sources = [
"gst/udp/xxx.c",
...
]
configs = [ ":gst_plugins_config" ]
}
ohos_shared_library("gstudp") {
deps = [
":gstudp_source",
"//third_party/glib:glib",
"//third_party/glib:gobject",
"//third_party/glib:gio",
"//third_party/gstreamer/gstreamer:gstbase",
"//third_party/gstreamer/gstreamer:gstreamer",
]
relative_install_dir = "media/plugins"
part_name = "gstreamer"
subsystem_name = "thirdparty"
}
third_party/gstreamer/gstplugins_good/gst/udp/gstudpsrc.c 修改,_GNU_SOURCE重定义错误
+ #ifdef _GNU_SOURCE
+ #undef _GNU_SOURCE
+ #endif
#define _GNU_SOURCE
/* Needed for OSX/iOS to define the IPv6 variants */
#define __APPLE_USE_RFC_3542
#include <sys/types.h>
#ifdef HAVE_SYS_SOCKET_H
#include <sys/socket.h>
#endif
编译生成libgstudp.z.so,安装到/system/lib/media/plugins 目录
5、gstplugins_good_packages
third_party/gstreamer/gstplugins_good/BUILD.gn 修改
group("gstplugins_good_packages") {
deps = [
":gstaudiofx",
":gstaudioparsers",
":gstisomp4",
":gstmatroska",
":gstmultifile",
":gstwavparse",
+ ":gstrtsp",
+ ":gstrtpmanager",
+ ":gstrtp",
+ ":gstudp",
]
}
六、AVPlayer framework
1、在AVPlayer的NAPI代码中限制只能设置http前缀的网络url,需要解除限制
/foundation/multimedia/player_framework/frameworks/js/avplayer/avplayer_napi.cpp 修改
void AVPlayerNapi::SetSource(std::string url)
{
MEDIA_LOGI("input url is %{public}s!", url.c_str());
bool isFd = (url.find("fd://") != std::string::npos) ? true : false;
- bool isNetwork = (url.find("http") != std::string::npos) ? true : false;
+ bool isNetwork = ((url.find("http") != std::string::npos) || (url.find("rtsp") != std::string::npos)) ? true : false;
if (isNetwork) {
auto task = std::make_shared<TaskHandler<void>>([this, url]() {
MEDIA_LOGI("SetNetworkSource Task");
std::unique_lock<std::mutex> lock(taskMutex_);
auto state = GetCurrentState();
if (state != AVPlayerState::STATE_IDLE) {
OnErrorCb(MSERR_EXT_API9_OPERATE_NOT_PERMIT, "current state is not idle, unsupport set url");
return;
}
if (player_ != nullptr) {
if (player_->SetSource(url) != MSERR_OK) {
OnErrorCb(MSERR_EXT_API9_INVALID_PARAMETER, "failed to SetSourceNetWork");
}
stopWait_ = false;
LISTENER(stateChangeCond_.wait(lock, [this]() { return stopWait_.load(); }), "SetSourceNetWork", false)
}
});
(void)taskQue_->EnqueueTask(task);
} else if (isFd) {
...
}
2、支持网络缓冲
/foundation/multimedia/services/engine/gstreamer/common/playbin_adapter/playbin_ctrler_base.cpp 修改
int32_t PlayBinCtrlerBase::SetSource(const std::string &url)
{
std::unique_lock<std::mutex> lock(mutex_);
uri_ = url;
- if (url.find("http") == 0 || url.find("https") == 0 || EnableBufferingBySysParam()) {
+ if (url.find("http") == 0 || url.find("https") == 0 || url.find("rtsp") == 0 || EnableBufferingBySysParam()) {
isNetWorkPlay_ = true;
}
MEDIA_LOGI("Set source: %{public}s", url.c_str());
return MSERR_OK;
}
七、rtp over tcp
RTSP支持以单播或组播模式通过TCP或UDP传输rtp数据包。默认情况下,rtspsrc将按以下顺序协商连接:UDP单播/UDP组播/TCP,不能更改顺序,但可以使用“protocols”属性控制允许的协议。
注:1>、安装上面的库后tcp和udp传输协议都能支持;2>、以下代码仅用于测试rtp over tcp的能力,除非有明确需求只能通过tcp传输,否则不要同步
1、protocols 属性支持的参数如下:
third_party/gstreamer/gstplugins_base/gst-libs/gst/rtsp/gstrtsptransport.h
typedef enum {
GST_RTSP_LOWER_TRANS_UNKNOWN = 0,
GST_RTSP_LOWER_TRANS_UDP = (1 << 0),
GST_RTSP_LOWER_TRANS_UDP_MCAST = (1 << 1),
GST_RTSP_LOWER_TRANS_TCP = (1 << 2),
GST_RTSP_LOWER_TRANS_HTTP = (1 << 4),
GST_RTSP_LOWER_TRANS_TLS = (1 << 5)
} GstRTSPLowerTrans;
2、/foundation/multimedia/services/engine/gstreamer/common/playbin_adapter/playbin_ctrler_base.cpp 添加
void PlayBinCtrlerBase::OnElementSetup(GstElement &elem)
{
MEDIA_LOGD("element setup: %{public}s", ELEM_NAME(&elem));
...
if (OnVideoDecoderSetup(elem) || strncmp(ELEM_NAME(&elem), "multiqueue", strlen("multiqueue")) == 0 ||
strncmp(ELEM_NAME(&elem), "qtdemux", strlen("qtdemux")) == 0) {
MEDIA_LOGI("add msgfilter element: %{public}s", ELEM_NAME(&elem));
msgProcessor_->AddMsgFilter(ELEM_NAME(&elem));
}
+ if (strncmp(ELEM_NAME(&elem), "source", strlen("source") == 0) {
+ GstElementFactory *elementFac = gst_element_get_factory(&elem);
+ const gchar *name = gst_plugin_feature_get_name (GST_PLUGIN_FEATURE (elementFac));
+ if ((name != nullptr) && (strstr(name, "rtspsrc") != nullptr)) {
+ g_object_set(&elem, "protocols", (1 << 2), nullptr); // (1<<2) 代表GST_RTSP_LOWER_TRANS_TCP,设置rtspsrc只允许用tcp协议与服务器协商
+ }
+ }
...
}
八、dot图
待补充
gstudpsrc->gstrtpsession->gstrtpstorage->gstrtpssrcdemux->gstrtpjitterbuffer->gstrtpptdemux->gsttypefindelement->gstrtpmp2tdepay->tsdemux...
九、注意事项
1、上述代码中BUILD.gn的sources 需要编译的C文件基本都没写,代码有点多,后续有时间补充;最简单的方法:gio库添加gio目录下所有的除window相关的源文件,gstreamer库添加相应目录下所有的源文件,编译过即可;其实部分源文件并不会被rtsp依赖,可以适当裁剪部分元素以减少动态库大小。
2、插件库的库名必须和文章中定义的相同,并设置relative_install_dir=“media/plugins”将库安装到/system/lib/media/plugins目录。
十、结语
下一篇将介绍gstreamer rtsp插件及板端播放rtsp节目异常的一些调试方法。
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