- gst-plugins-base -> Modify Makefile and patch, compile ok.
- gst-plugins-good -> Modify Makefile and patch, compile ok.
- gst-plugins-ugly -> To be check if required.
- gst-plugins-bad -> Modify Makefile and patch, compile ok.
- gst-libav -> To be check if required.
Proprietary plugins
- package folder -> Modify the Makefile.
- gst-plugin foler -> Modify the configure.ac
- gst-plugin/src folder -> Modify the Makefile.am
- plugin source
* GST_BOILERPLATE_FULL is gone, replace by G_DEFINE_TYPE_WITH_CODE
* add the parent_class definition.
#if 0
GST_BOILERPLATE_FULL (GstA5sSrc, gst_a5s_src, GstBaseSrc, GST_TYPE_BASE_SRC, gst_a5s_src_init_interfaces);
#else
#define gst_a5s_src_parent_iclass parent_class
G_DEFINE_TYPE_WITH_CODE (GstA5sSrc, gst_a5s_src, GST_TYPE_BASE_SRC, gst_a5s_src_init_interfaces);
#endif
* init funtion parameters is change.
#if 0
static void gst_a5s_src_init (GstA5sSrc * a5ssrc, GstA5sSrcClass * a5ssrc_class)
#else
static void gst_a5s_src_init (GstA5sSrc * a5ssrc)
#endif
* remove the gone already functions.
* audio: Need lots of works to do.
* buffer size: replace the GST_BUFFER_SIZE() to gst_buffer_get_size().
* buffer data: replace the GST_BUFFER_DATA() by using the gst_buffer_map to access the data.
Audio plugins:
- GstRingBufferSpec -> GstAudioRingBufferSpec
- The GstAudioRingBufferSpec has big changes. Lots parameters are re-factored in the GstAudioInfo.
- The GstAudioInfo structure is as follow. The format part is in GstAudioFormatInfo.
/**
* GstAudioInfo:
* @finfo: the format info of the audio
* @flags: additional audio flags
* @layout: audio layout
* @rate: the audio sample rate
* @channels: the number of channels
* @bpf: the number of bytes for one frame, this is the size of one
* sample * @channels
* @position: the positions for each channel
*
* Information describing audio properties. This information can be filled
* in from GstCaps with gst_audio_info_from_caps().
*
* Use the provided macros to access the info in this structure.
*/
struct _GstAudioInfo {
const GstAudioFormatInfo *finfo;
GstAudioFlags flags;
GstAudioLayout layout;
gint rate;
gint channels;
gint bpf;
GstAudioChannelPosition position[64];
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
- GstAudioFormatInfo
/**
* GstAudioFormatInfo:
* @format: #GstAudioFormat
* @name: string representation of the format
* @description: user readable description of the format
* @flags: #GstAudioFormatFlags
* @endianness: the endianness
* @width: amount of bits used for one sample
* @depth: amount of valid bits in @width
* @silence: @width/8 bytes with 1 silent sample
* @unpack_format: the format of the unpacked samples
* @unpack_func: function to unpack samples
* @pack_func: function to pack samples
*
* Information for an audio format.
*/
struct _GstAudioFormatInfo {
GstAudioFormat format;
const gchar *name;
const gchar *description;
GstAudioFormatFlags flags;
gint endianness;
gint width;
gint depth;
guint8 silence[8];
GstAudioFormat unpack_format;
GstAudioFormatUnpack unpack_func;
GstAudioFormatPack pack_func;
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
- Change the audio/x-raw-int to audio/x-raw for pad template.
- To do list:
* Check the timestamps mechanism in new alsa plugin example.
* Check the audio GstAudioRingBufferSpec parameters.
Applications
- gst-rtsp-server
* execute the autogen.sh and install the required tool to generate necessary files.
* add the AC_PROG_CXX in configure.ac because our example is cpp based.
* create patch to make gst-libs/gst/rtsp in gst-plugins-base to be the latest master version.
* after above patch, the gst-rtsp-server's gst/rtsp-server library could be successfully built.
* gst_debug_add_log_function() function parameters change from 2 to 3.
gst_debug_add_log_function(gst_log_function_syslog, NULL, NULL);
* buffer size: replace the GST_BUFFER_SIZE() to gst_buffer_get_size().
* buffer data: replace the GST_BUFFER_DATA() by using the gst_buffer_map to access the data.
GstBuffer *buffer = gst_app_sink_pull_preroll(sink);
GstMapInfo mapinfo = {0, };
gst_buffer_map(buffer, &mapinfo, GST_MAP_READ);
/* access the buffer by mapinfo.data and mapinfo.size */
gst_buffer_unmap(buffer, &mapinfo);
* g_thread_create remove, use g_thread_new -> check further.
* gst_app_sink_set_callbacks() -> callbacks structure is change.
* ways change to handle sink callback functions.
#if 0
GstBuffer *buffer = gst_app_sink_pull_preroll(sink);
#else
GstSample *sample = gst_app_sink_pull_preroll(sink);
GstBuffer *buffer = gst_sampe_get_buffer(sample);
#endif
#if 0
GstBuffer *buffer = gst_app_sink_pull_buffer(sink);
#else
GstSample *sample = gst_app_sink_pull_sample(sink);
GstBuffer *buffer = gst_sampe_get_buffer(sample);
#endif
* The GstRTSPMediaMapping rename to be GstRTSPMountPoints, we can compare by the examples.
* gst_rtsp_server_get_media_mapping() become gst_rtsp_server_get_mount_points()
* gst_rtsp_media_mapping_add_factory() become gst_rtsp_mount_points_add_factory()
* In configure.ac, remove the WARNING_CFLAGS because some warnings would cause errors.
#GST_OPTION_CFLAGS="\$(WARNING_CFLAGS) \$(ERROR_CFLAGS) \$(DEBUG_CFLAGS) \$(PROFILE_CFLAGS) \$(GCOV_CFLAGS) \$(OPT_CFLAGS) \$(DEPRECATED_CFLAGS)"
GST_OPTION_CFLAGS="\$(ERROR_CFLAGS) \$(DEBUG_CFLAGS) \$(PROFILE_CFLAGS) \$(GCOV_CFLAGS) \$(OPT_CFLAGS) \$(DEPRECATED_CFLAGS)"
