我编写了一个类来代表我的 RTSP 服务器。实施：
gstRtsp::gstRtsp()
{
initializeGst();
m_server = gst_rtsp_server_new ();
gst_rtsp_server_set_service (m_server, "6666"); //set port number
m_mounts = gst_rtsp_server_get_mount_points (m_server);
}
gstRtsp::~gstRtsp(){
if (m_running) {
gst_rtsp_mount_points_remove_factory(m_mounts, "/stream");
g_source_remove(m_serverId);
}
g_object_unref(m_server);
g_object_unref (m_mounts);
}
void gstRtsp::initializeGst(int debugLevel)
{
if (!gst_is_initialized())
{
setenv("GST_DEBUG", ("*:" + std::to_string(debugLevel)).c_str(), 1);
gst_init(nullptr, nullptr);
}
}
void gstRtsp::rtspSendData(const uint8_t *data)
{
mtx.lock();
for (RtspClient &client: rtspClients) {
if (!GST_IS_ELEMENT(client.appsrc)) {
std::vector<RtspClient>::iterator i = std::find_if(rtspClients.begin(), rtspClients.end(), client);
rtspClients.erase(i);
} else {
if (client.data_needed) {
GstBuffer *buffer;
GstMemory *memory;
memory = gst_memory_new_wrapped( (GstMemoryFlags)0, (gpointer) data, m_width * m_height * 3, 0, m_width * m_height * 3, NULL, NULL);
buffer = gst_buffer_new();
gst_buffer_append_memory(buffer, memory);
if (client.timepoint == 0) {
client.timepoint = currentTime.toMSecsSinceEpoch();
} else {
client.timestamp = currentTime.toMSecsSinceEpoch()* 1000000;
}
GST_BUFFER_PTS (buffer) = client.timestamp;
GST_BUFFER_DTS (buffer) = client.timestamp;
GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (1, GST_SECOND, (int)m_frameRate);
gst_app_src_push_buffer(GST_APP_SRC(client.appsrc), buffer);
}
}
}
mtx.unlock();
}
/* called when we need to give data to appsrc */
static void rtspDataNeeded(GstElement * appsrc, guint unused, gstRtsp *instance)
{
instance->mtx.lock();
std::vector<gstRtsp::RtspClient>::iterator i = std::find_if(instance->rtspClients.begin(), instance->rtspClients.end(),
[appsrc] (const gstRtsp::RtspClient &client) { return client.appsrc == appsrc; } );
if (i != instance->rtspClients.end()) {
(*i).data_needed = true;
}
instance->mtx.unlock();
}
/* called when a new media pipeline is constructed. We can query the pipeline and configure our appsrc */
static void rtspMediaConfigure(GstRTSPMediaFactory * , GstRTSPMedia * media, gstRtsp *instance)
{
GstElement *element, *appsrc;
/* get the element used for providing the streams of the media */
element = gst_rtsp_media_get_element (media);
/* get our appsrc, we named it 'mysrc' with the name property */
appsrc = gst_bin_get_by_name_recurse_up (GST_BIN (element), "mysrc");
/* this instructs appsrc that we will be dealing with timed buffer */
gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
/* configure the caps of the video */
g_object_set (G_OBJECT (appsrc), "caps",
gst_caps_new_simple ("video/x-raw",
"format", G_TYPE_STRING, "RGB",
"width", G_TYPE_INT, instance->width(),
"height", G_TYPE_INT, instance->height(),
"interlace-mode", G_TYPE_STRING, "progressive",
"framerate", GST_TYPE_FRACTION, (int)(instance->frameRate()), 1,
"pixel-aspect-ratio", GST_TYPE_FRACTION, 1, 1,
NULL), NULL);
instance->mtx.lock();
instance->rtspClients.push_back({appsrc});
instance->mtx.unlock();
/* install the callback that will be called when a buffer is needed */
g_signal_connect (appsrc, "need-data", (GCallback) rtspDataNeeded, instance);
gst_object_unref (appsrc);
gst_object_unref (element);
}
void gstRtsp::startRtsp(int width, int height, int frameRate )
{
m_frameRate = frameRate;
m_width = width;
m_height = height;
/* make a media factory for a test stream. The default media factory can use
* gst-launch syntax to create pipelines.
* any launch line works as long as it contains elements named pay%d. Each
* element with pay%d names will be a stream */
GstRTSPMediaFactory *factory = gst_rtsp_media_factory_new ();
std::string initializationString = "( appsrc name=mysrc ! videoconvert ! avenc_h264_omx ! h264parse ! rtph264pay name=pay0 pt=96 )";
gst_rtsp_media_factory_set_launch (factory, initializationString.c_str() );
gst_rtsp_media_factory_set_shared (factory, TRUE);
gst_rtsp_media_factory_set_eos_shutdown(factory, TRUE);
/* notify when our media is ready, This is called whenever someone asks for
* the media and a new pipeline with our appsrc is created */
g_signal_connect (factory, "media-configure", (GCallback) rtspMediaConfigure, this);
rtspCounter = 0;
rtspCounterTimer = QDateTime::currentDateTime().toMSecsSinceEpoch();
m_running = true;
gst_rtsp_mount_points_add_factory (m_mounts, "/stream", factory);
m_serverId = gst_rtsp_server_attach (m_server, NULL);
}
int gstRtsp::width(){
return m_width;
}
int gstRtsp::height(){
return m_height;
}
int gstRtsp::frameRate(){
return m_frameRate;
}
为了测试它，我每隔 0.3 秒调用 rtspSendData() 填充 160x120x3 数组（分辨率 160x120，RGB）（只是一个睡眠循环）。但是，接收端的协商失败（管道“gst-launch-1.0 rtspsrc location=rtsp://IP:6666/stream Latency=100！rtph264depay！avdec_h264！autovideosink”失败，元素 /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0 出现错误：未处理的错误）。
在服务器端的调试输出（在 RPI0w 上）有一些警告
0:00:46.544840112 4180 0x7f4c00ef00 WARN videoencoder gstvideoencoder.c:755:gst_video_encoder_setcaps:<avenc_h264_omx0> rejected caps video/x-raw, width=(int)160, height=(int)120, framerate=(fraction)9/1, pixel-aspect-ratio=(fraction)1/1, format=(string)I420, interlace-mode=(string)progressive
0:00:46.549196023 4180 0x7f4c00ef00 WARN videoencoder gstvideoencoder.c:755:gst_video_encoder_setcaps:<avenc_h264_omx0> rejected caps video/x-raw, width=(int)160, height=(int)120, framerate=(fraction)9/1, pixel-aspect-ratio=(fraction)1/1, format=(string)I420, interlace-mode=(string)progressive
0:00:46.549330711 4180 0x7f4c00ef00 WARN GST_PADS gstpad.c:4303:gst_pad_peer_query:<videoconvert0:src> could not send sticky events
0:01:06.517746667 4180 0x55c95532a0 WARN rtspmedia rtsp-media.c:3576:wait_preroll: failed to preroll pipeline
0:01:06.517892501 4180 0x55c95532a0 WARN rtspmedia rtsp-media.c:3946:gst_rtsp_media_prepare: failed to preroll pipeline
0:01:06.517987866 4180 0x55c95532a0 INFO rtspmedia rtsp-media.c:4079:gst_rtsp_media_unprepare: unprepare media 0x7f540e01b0
还有两个错误
0:01:06.549108304 4180 0x55c95532a0 ERROR rtspclient rtsp-client.c:1087:find_media: client 0x55c9636140: can't prepare media
...
0:01:06.558417003 4180 0x55c95532a0 ERROR rtspclient rtsp-client.c:3346:handle_describe_request: client 0x55c9636140: no media
我错过了什么？据我了解，两端的管道都开始正常，所以需要进行一些协商？
好的，所以 RPI 上的 h264 编码显然存在一些问题（https://github.com/aiortc/aiortc/issues/899；使用 v4l 编码器时，即使使用
GST_DEBUG=2 gst-launch-1.0 videotestsrc ! v4l2h264enc ! fakesink
gstRtsp::gstRtsp()
{
initializeGst();
m_server = gst_rtsp_server_new ();
gst_rtsp_server_set_service (m_server, "8554"); //set port number
m_mounts = gst_rtsp_server_get_mount_points (m_server);
}
gstRtsp::~gstRtsp(){
if (m_running) {
gst_rtsp_mount_points_remove_factory(m_mounts, "/stream");
g_source_remove(m_serverId);
}
g_object_unref(m_server);
g_object_unref (m_mounts);
}
void gstRtsp::initializeGst(int debugLevel)
{
if (!gst_is_initialized())
{
setenv("GST_DEBUG", ("*:" + std::to_string(debugLevel)).c_str(), 1);
gst_init(nullptr, nullptr);
}
}
void gstRtsp::rtspSendData(const uint8_t *data)
{
for (client_t &client: m_clients) {
if (!GST_IS_ELEMENT(client.appsrc)) {
std::vector<client_t>::iterator i = std::find_if(m_clients.begin(), m_clients.end(), client);
m_clients.erase(i);
} else {
if (client.data_needed) {
GstBuffer *buffer;
GstMemory *memory;
memory = gst_memory_new_wrapped( (GstMemoryFlags)0, (gpointer) data, m_width * m_height * 3, 0, m_width * m_height * 3, NULL, NULL);
buffer = gst_buffer_new();
gst_buffer_append_memory(buffer, memory);
if (client.timepoint == 0) {
client.timepoint = QDateTime::currentDateTime().toMSecsSinceEpoch();
} else {
client.timestamp = (QDateTime::currentDateTime().toMSecsSinceEpoch() - client.timepoint) * 1000000;
}
GST_BUFFER_PTS (buffer) = client.timestamp;
GST_BUFFER_DTS (buffer) = client.timestamp;
GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (1, GST_SECOND, (int)m_frameRate);
gst_app_src_push_buffer(GST_APP_SRC(client.appsrc), buffer);
}
}
}
}
/* called when we need to give data to appsrc */
static void rtspDataNeeded(GstElement * appsrc, guint/* unused*/, gstRtsp *instance)
{
std::vector<gstRtsp::client_t>::iterator i = std::find_if(instance->m_clients.begin(), instance->m_clients.end(),
[appsrc] (const gstRtsp::client_t &client) { return client.appsrc == appsrc; } );
if (i != instance->m_clients.end()) {
(*i).data_needed = true;
}
}
/* called when a new media pipeline is constructed. We can query the pipeline and configure our appsrc */
static void rtspMediaConfigure(GstRTSPMediaFactory * , GstRTSPMedia * media, gstRtsp *instance)
{
GstElement *element, *appsrc;
/* get the element used for providing the streams of the media */
element = gst_rtsp_media_get_element (media);
/* get our appsrc, we named it 'mysrc' with the name property */
appsrc = gst_bin_get_by_name_recurse_up (GST_BIN (element), "mysrc");
/* this instructs appsrc that we will be dealing with timed buffer */
gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
/* configure the caps of the video */
g_object_set (G_OBJECT (appsrc), "caps",
gst_caps_new_simple ("video/x-raw",
"format", G_TYPE_STRING, "RGB",
"width", G_TYPE_INT, instance->width(),
"height", G_TYPE_INT, instance->height(),
"interlace-mode", G_TYPE_STRING, "progressive",
"framerate", GST_TYPE_FRACTION, (int)(instance->frameRate()), 1,
"pixel-aspect-ratio", GST_TYPE_FRACTION, 1, 1,
NULL), NULL);
instance->m_clients.push_back({appsrc});
/* install the callback that will be called when a buffer is needed */
g_signal_connect (appsrc, "need-data", (GCallback) rtspDataNeeded, instance);
gst_object_unref (appsrc);
gst_object_unref (element);
}
void gstRtsp::startRtsp(int width, int height, int frameRate )
{
m_frameRate = frameRate;
m_width = width;
m_height = height;
/* make a media factory for a test stream. The default media factory can use
* gst-launch syntax to create pipelines.
* any launch line works as long as it contains elements named pay%d. Each
* element with pay%d names will be a stream */
GstRTSPMediaFactory *factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_launch (factory, "( appsrc name=mysrc ! videoconvert ! v4l2jpegenc ! jpegparse ! rtpjpegpay name=pay0 pt=96 )" );
gst_rtsp_media_factory_set_shared (factory, TRUE);
gst_rtsp_media_factory_set_eos_shutdown(factory, TRUE);
/* notify when our media is ready, This is called whenever someone asks for
* the media and a new pipeline with our appsrc is created */
g_signal_connect (factory, "media-configure", (GCallback) rtspMediaConfigure, this);
m_running = true;
gst_rtsp_mount_points_add_factory (m_mounts, "/stream", factory);
m_serverId = gst_rtsp_server_attach (m_server, NULL);
}
int gstRtsp::width(){
return m_width;
}
int gstRtsp::height(){
return m_height;
}
int gstRtsp::frameRate(){
return m_frameRate;
}