| /* GStreamer |
| * Copyright (C) <2006> Wim Taymans <wim.taymans@gmail.com> |
| * |
| * This library is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Library General Public |
| * License as published by the Free Software Foundation; either |
| * version 2 of the License, or (at your option) any later version. |
| * |
| * This library is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Library General Public License for more details. |
| * |
| * You should have received a copy of the GNU Library General Public |
| * License along with this library; if not, write to the |
| * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, |
| * Boston, MA 02110-1301, USA. |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| # include "config.h" |
| #endif |
| |
| #include <string.h> |
| |
| #include <gst/base/gstbitreader.h> |
| #include <gst/rtp/gstrtpbuffer.h> |
| |
| #include "gstrtpmp4gpay.h" |
| #include "gstrtputils.h" |
| |
| GST_DEBUG_CATEGORY_STATIC (rtpmp4gpay_debug); |
| #define GST_CAT_DEFAULT (rtpmp4gpay_debug) |
| |
| static GstStaticPadTemplate gst_rtp_mp4g_pay_sink_template = |
| GST_STATIC_PAD_TEMPLATE ("sink", |
| GST_PAD_SINK, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("video/mpeg," |
| "mpegversion=(int) 4," |
| "systemstream=(boolean)false;" |
| "audio/mpeg," "mpegversion=(int) 4, " "stream-format=(string) raw") |
| ); |
| |
| static GstStaticPadTemplate gst_rtp_mp4g_pay_src_template = |
| GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("application/x-rtp, " |
| "media = (string) { \"video\", \"audio\", \"application\" }, " |
| "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " |
| "clock-rate = (int) [1, MAX ], " |
| "encoding-name = (string) \"MPEG4-GENERIC\", " |
| /* required string params */ |
| "streamtype = (string) { \"4\", \"5\" }, " /* 4 = video, 5 = audio */ |
| /* "profile-level-id = (string) [1,MAX], " */ |
| /* "config = (string) [1,MAX]" */ |
| "mode = (string) { \"generic\", \"CELP-cbr\", \"CELP-vbr\", \"AAC-lbr\", \"AAC-hbr\" } " |
| /* Optional general parameters */ |
| /* "objecttype = (string) [1,MAX], " */ |
| /* "constantsize = (string) [1,MAX], " *//* constant size of each AU */ |
| /* "constantduration = (string) [1,MAX], " *//* constant duration of each AU */ |
| /* "maxdisplacement = (string) [1,MAX], " */ |
| /* "de-interleavebuffersize = (string) [1,MAX], " */ |
| /* Optional configuration parameters */ |
| /* "sizelength = (string) [1, 16], " *//* max 16 bits, should be enough... */ |
| /* "indexlength = (string) [1, 8], " */ |
| /* "indexdeltalength = (string) [1, 8], " */ |
| /* "ctsdeltalength = (string) [1, 64], " */ |
| /* "dtsdeltalength = (string) [1, 64], " */ |
| /* "randomaccessindication = (string) {0, 1}, " */ |
| /* "streamstateindication = (string) [0, 64], " */ |
| /* "auxiliarydatasizelength = (string) [0, 64]" */ ) |
| ); |
| |
| |
| static void gst_rtp_mp4g_pay_finalize (GObject * object); |
| |
| static GstStateChangeReturn gst_rtp_mp4g_pay_change_state (GstElement * element, |
| GstStateChange transition); |
| |
| static gboolean gst_rtp_mp4g_pay_setcaps (GstRTPBasePayload * payload, |
| GstCaps * caps); |
| static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstRTPBasePayload * |
| payload, GstBuffer * buffer); |
| static gboolean gst_rtp_mp4g_pay_sink_event (GstRTPBasePayload * payload, |
| GstEvent * event); |
| |
| #define gst_rtp_mp4g_pay_parent_class parent_class |
| G_DEFINE_TYPE (GstRtpMP4GPay, gst_rtp_mp4g_pay, GST_TYPE_RTP_BASE_PAYLOAD) |
| |
| static void gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * klass) |
| { |
| GObjectClass *gobject_class; |
| GstElementClass *gstelement_class; |
| GstRTPBasePayloadClass *gstrtpbasepayload_class; |
| |
| gobject_class = (GObjectClass *) klass; |
| gstelement_class = (GstElementClass *) klass; |
| gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; |
| |
| gobject_class->finalize = gst_rtp_mp4g_pay_finalize; |
| |
| gstelement_class->change_state = gst_rtp_mp4g_pay_change_state; |
| |
| gstrtpbasepayload_class->set_caps = gst_rtp_mp4g_pay_setcaps; |
| gstrtpbasepayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer; |
| gstrtpbasepayload_class->sink_event = gst_rtp_mp4g_pay_sink_event; |
| |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_rtp_mp4g_pay_src_template); |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_rtp_mp4g_pay_sink_template); |
| |
| gst_element_class_set_static_metadata (gstelement_class, |
| "RTP MPEG4 ES payloader", |
| "Codec/Payloader/Network/RTP", |
| "Payload MPEG4 elementary streams as RTP packets (RFC 3640)", |
| "Wim Taymans <wim.taymans@gmail.com>"); |
| |
| GST_DEBUG_CATEGORY_INIT (rtpmp4gpay_debug, "rtpmp4gpay", 0, |
| "MP4-generic RTP Payloader"); |
| } |
| |
| static void |
| gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay) |
| { |
| rtpmp4gpay->adapter = gst_adapter_new (); |
| } |
| |
| static void |
| gst_rtp_mp4g_pay_reset (GstRtpMP4GPay * rtpmp4gpay) |
| { |
| GST_DEBUG_OBJECT (rtpmp4gpay, "reset"); |
| |
| gst_adapter_clear (rtpmp4gpay->adapter); |
| } |
| |
| static void |
| gst_rtp_mp4g_pay_cleanup (GstRtpMP4GPay * rtpmp4gpay) |
| { |
| gst_rtp_mp4g_pay_reset (rtpmp4gpay); |
| |
| g_free (rtpmp4gpay->params); |
| rtpmp4gpay->params = NULL; |
| |
| if (rtpmp4gpay->config) |
| gst_buffer_unref (rtpmp4gpay->config); |
| rtpmp4gpay->config = NULL; |
| |
| g_free (rtpmp4gpay->profile); |
| rtpmp4gpay->profile = NULL; |
| |
| rtpmp4gpay->streamtype = NULL; |
| rtpmp4gpay->mode = NULL; |
| |
| rtpmp4gpay->frame_len = 0; |
| } |
| |
| static void |
| gst_rtp_mp4g_pay_finalize (GObject * object) |
| { |
| GstRtpMP4GPay *rtpmp4gpay; |
| |
| rtpmp4gpay = GST_RTP_MP4G_PAY (object); |
| |
| gst_rtp_mp4g_pay_cleanup (rtpmp4gpay); |
| |
| g_object_unref (rtpmp4gpay->adapter); |
| rtpmp4gpay->adapter = NULL; |
| |
| G_OBJECT_CLASS (parent_class)->finalize (object); |
| } |
| |
| static const unsigned int sampling_table[16] = { |
| 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, |
| 16000, 12000, 11025, 8000, 7350, 0, 0, 0 |
| }; |
| |
| static gboolean |
| gst_rtp_mp4g_pay_parse_audio_config (GstRtpMP4GPay * rtpmp4gpay, |
| GstBuffer * buffer) |
| { |
| GstMapInfo map; |
| guint8 objectType = 0; |
| guint8 samplingIdx = 0; |
| guint8 channelCfg = 0; |
| GstBitReader br; |
| |
| gst_buffer_map (buffer, &map, GST_MAP_READ); |
| |
| gst_bit_reader_init (&br, map.data, map.size); |
| |
| /* any object type is fine, we need to copy it to the profile-level-id field. */ |
| if (!gst_bit_reader_get_bits_uint8 (&br, &objectType, 5)) |
| goto too_short; |
| if (objectType == 0) |
| goto invalid_object; |
| |
| if (!gst_bit_reader_get_bits_uint8 (&br, &samplingIdx, 4)) |
| goto too_short; |
| /* only fixed values for now */ |
| if (samplingIdx > 12 && samplingIdx != 15) |
| goto wrong_freq; |
| |
| if (!gst_bit_reader_get_bits_uint8 (&br, &channelCfg, 4)) |
| goto too_short; |
| if (channelCfg > 7) |
| goto wrong_channels; |
| |
| /* rtp rate depends on sampling rate of the audio */ |
| if (samplingIdx == 15) { |
| guint32 rate = 0; |
| |
| /* index of 15 means we get the rate in the next 24 bits */ |
| if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24)) |
| goto too_short; |
| |
| rtpmp4gpay->rate = rate; |
| } else { |
| /* else use the rate from the table */ |
| rtpmp4gpay->rate = sampling_table[samplingIdx]; |
| } |
| |
| rtpmp4gpay->frame_len = 1024; |
| |
| switch (objectType) { |
| case 1: |
| case 2: |
| case 3: |
| case 4: |
| case 6: |
| case 7: |
| { |
| guint8 frameLenFlag = 0; |
| |
| if (gst_bit_reader_get_bits_uint8 (&br, &frameLenFlag, 1)) |
| if (frameLenFlag) |
| rtpmp4gpay->frame_len = 960; |
| |
| break; |
| } |
| default: |
| break; |
| } |
| |
| /* extra rtp params contain the number of channels */ |
| g_free (rtpmp4gpay->params); |
| rtpmp4gpay->params = g_strdup_printf ("%d", channelCfg); |
| /* audio stream type */ |
| rtpmp4gpay->streamtype = "5"; |
| /* mode only high bitrate for now */ |
| rtpmp4gpay->mode = "AAC-hbr"; |
| /* profile */ |
| g_free (rtpmp4gpay->profile); |
| rtpmp4gpay->profile = g_strdup_printf ("%d", objectType); |
| |
| GST_DEBUG_OBJECT (rtpmp4gpay, |
| "objectType: %d, samplingIdx: %d (%d), channelCfg: %d, frame_len %d", |
| objectType, samplingIdx, rtpmp4gpay->rate, channelCfg, |
| rtpmp4gpay->frame_len); |
| |
| gst_buffer_unmap (buffer, &map); |
| return TRUE; |
| |
| /* ERROR */ |
| too_short: |
| { |
| GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT, |
| (NULL), ("config string too short")); |
| gst_buffer_unmap (buffer, &map); |
| return FALSE; |
| } |
| invalid_object: |
| { |
| GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT, |
| (NULL), ("invalid object type")); |
| gst_buffer_unmap (buffer, &map); |
| return FALSE; |
| } |
| wrong_freq: |
| { |
| GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED, |
| (NULL), ("unsupported frequency index %d", samplingIdx)); |
| gst_buffer_unmap (buffer, &map); |
| return FALSE; |
| } |
| wrong_channels: |
| { |
| GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED, |
| (NULL), ("unsupported number of channels %d, must < 8", channelCfg)); |
| gst_buffer_unmap (buffer, &map); |
| return FALSE; |
| } |
| } |
| |
| #define VOS_STARTCODE 0x000001B0 |
| |
| static gboolean |
| gst_rtp_mp4g_pay_parse_video_config (GstRtpMP4GPay * rtpmp4gpay, |
| GstBuffer * buffer) |
| { |
| GstMapInfo map; |
| guint32 code; |
| |
| gst_buffer_map (buffer, &map, GST_MAP_READ); |
| |
| if (map.size < 5) |
| goto too_short; |
| |
| code = GST_READ_UINT32_BE (map.data); |
| |
| g_free (rtpmp4gpay->profile); |
| if (code == VOS_STARTCODE) { |
| /* get profile */ |
| rtpmp4gpay->profile = g_strdup_printf ("%d", (gint) map.data[4]); |
| } else { |
| GST_ELEMENT_WARNING (rtpmp4gpay, STREAM, FORMAT, |
| (NULL), ("profile not found in config string, assuming \'1\'")); |
| rtpmp4gpay->profile = g_strdup ("1"); |
| } |
| |
| /* fixed rate */ |
| rtpmp4gpay->rate = 90000; |
| /* video stream type */ |
| rtpmp4gpay->streamtype = "4"; |
| /* no params for video */ |
| rtpmp4gpay->params = NULL; |
| /* mode */ |
| rtpmp4gpay->mode = "generic"; |
| |
| GST_LOG_OBJECT (rtpmp4gpay, "profile %s", rtpmp4gpay->profile); |
| |
| gst_buffer_unmap (buffer, &map); |
| |
| return TRUE; |
| |
| /* ERROR */ |
| too_short: |
| { |
| GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT, |
| (NULL), ("config string too short")); |
| gst_buffer_unmap (buffer, &map); |
| return FALSE; |
| } |
| } |
| |
| static gboolean |
| gst_rtp_mp4g_pay_new_caps (GstRtpMP4GPay * rtpmp4gpay) |
| { |
| gchar *config; |
| GValue v = { 0 }; |
| gboolean res; |
| |
| #define MP4GCAPS \ |
| "streamtype", G_TYPE_STRING, rtpmp4gpay->streamtype, \ |
| "profile-level-id", G_TYPE_STRING, rtpmp4gpay->profile, \ |
| "mode", G_TYPE_STRING, rtpmp4gpay->mode, \ |
| "config", G_TYPE_STRING, config, \ |
| "sizelength", G_TYPE_STRING, "13", \ |
| "indexlength", G_TYPE_STRING, "3", \ |
| "indexdeltalength", G_TYPE_STRING, "3", \ |
| NULL |
| |
| g_value_init (&v, GST_TYPE_BUFFER); |
| gst_value_set_buffer (&v, rtpmp4gpay->config); |
| config = gst_value_serialize (&v); |
| |
| /* hmm, silly */ |
| if (rtpmp4gpay->params) { |
| res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4gpay), |
| "encoding-params", G_TYPE_STRING, rtpmp4gpay->params, MP4GCAPS); |
| } else { |
| res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4gpay), |
| MP4GCAPS); |
| } |
| |
| g_value_unset (&v); |
| g_free (config); |
| |
| #undef MP4GCAPS |
| return res; |
| } |
| |
| static gboolean |
| gst_rtp_mp4g_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps) |
| { |
| GstRtpMP4GPay *rtpmp4gpay; |
| GstStructure *structure; |
| const GValue *codec_data; |
| const gchar *media_type = NULL; |
| gboolean res; |
| const gchar *name; |
| |
| rtpmp4gpay = GST_RTP_MP4G_PAY (payload); |
| |
| structure = gst_caps_get_structure (caps, 0); |
| |
| codec_data = gst_structure_get_value (structure, "codec_data"); |
| if (codec_data) { |
| GST_LOG_OBJECT (rtpmp4gpay, "got codec_data"); |
| if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) { |
| GstBuffer *buffer; |
| |
| buffer = gst_value_get_buffer (codec_data); |
| GST_LOG_OBJECT (rtpmp4gpay, "configuring codec_data"); |
| |
| name = gst_structure_get_name (structure); |
| |
| /* parse buffer */ |
| if (!strcmp (name, "audio/mpeg")) { |
| res = gst_rtp_mp4g_pay_parse_audio_config (rtpmp4gpay, buffer); |
| media_type = "audio"; |
| } else if (!strcmp (name, "video/mpeg")) { |
| res = gst_rtp_mp4g_pay_parse_video_config (rtpmp4gpay, buffer); |
| media_type = "video"; |
| } else { |
| res = FALSE; |
| } |
| if (!res) |
| goto config_failed; |
| |
| /* now we can configure the buffer */ |
| if (rtpmp4gpay->config) |
| gst_buffer_unref (rtpmp4gpay->config); |
| |
| rtpmp4gpay->config = gst_buffer_copy (buffer); |
| } |
| } else { |
| name = gst_structure_get_name (structure); |
| |
| if (!strcmp (name, "video/mpeg")) { |
| rtpmp4gpay->profile = g_strdup ("1"); |
| |
| /* fixed rate */ |
| rtpmp4gpay->rate = 90000; |
| /* video stream type */ |
| rtpmp4gpay->streamtype = "4"; |
| /* no params for video */ |
| rtpmp4gpay->params = NULL; |
| /* mode */ |
| rtpmp4gpay->mode = "generic"; |
| |
| media_type = "video"; |
| } |
| } |
| if (media_type == NULL) |
| goto config_failed; |
| |
| gst_rtp_base_payload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC", |
| rtpmp4gpay->rate); |
| |
| res = gst_rtp_mp4g_pay_new_caps (rtpmp4gpay); |
| |
| return res; |
| |
| /* ERRORS */ |
| config_failed: |
| { |
| GST_DEBUG_OBJECT (rtpmp4gpay, "failed to parse config"); |
| return FALSE; |
| } |
| } |
| |
| static GstFlowReturn |
| gst_rtp_mp4g_pay_flush (GstRtpMP4GPay * rtpmp4gpay) |
| { |
| guint avail, total; |
| GstBuffer *outbuf; |
| GstFlowReturn ret; |
| guint mtu; |
| |
| /* the data available in the adapter is either smaller |
| * than the MTU or bigger. In the case it is smaller, the complete |
| * adapter contents can be put in one packet. In the case the |
| * adapter has more than one MTU, we need to fragment the MPEG data |
| * over multiple packets. */ |
| total = avail = gst_adapter_available (rtpmp4gpay->adapter); |
| |
| ret = GST_FLOW_OK; |
| mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp4gpay); |
| |
| while (avail > 0) { |
| guint towrite; |
| guint8 *payload; |
| guint payload_len; |
| guint packet_len; |
| GstRTPBuffer rtp = { NULL }; |
| GstBuffer *paybuf; |
| |
| /* this will be the total lenght of the packet */ |
| packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0); |
| |
| /* fill one MTU or all available bytes, we need 4 spare bytes for |
| * the AU header. */ |
| towrite = MIN (packet_len, mtu - 4); |
| |
| /* this is the payload length */ |
| payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0); |
| |
| GST_DEBUG_OBJECT (rtpmp4gpay, |
| "avail %d, towrite %d, packet_len %d, payload_len %d", avail, towrite, |
| packet_len, payload_len); |
| |
| /* create buffer to hold the payload, also make room for the 4 header bytes. */ |
| outbuf = gst_rtp_buffer_new_allocate (4, 0, 0); |
| |
| gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); |
| |
| /* copy payload */ |
| payload = gst_rtp_buffer_get_payload (&rtp); |
| |
| /* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+ |
| * |AU-headers-length|AU-header|AU-header| |AU-header|padding| |
| * | | (1) | (2) | | (n) | bits | |
| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+ |
| */ |
| /* AU-headers-length, we only have 1 AU-header */ |
| payload[0] = 0x00; |
| payload[1] = 0x10; /* we use 16 bits for the header */ |
| |
| /* +---------------------------------------+ |
| * | AU-size | |
| * +---------------------------------------+ |
| * | AU-Index / AU-Index-delta | |
| * +---------------------------------------+ |
| * | CTS-flag | |
| * +---------------------------------------+ |
| * | CTS-delta | |
| * +---------------------------------------+ |
| * | DTS-flag | |
| * +---------------------------------------+ |
| * | DTS-delta | |
| * +---------------------------------------+ |
| * | RAP-flag | |
| * +---------------------------------------+ |
| * | Stream-state | |
| * +---------------------------------------+ |
| */ |
| /* The AU-header, no CTS, DTS, RAP, Stream-state |
| * |
| * AU-size is always the total size of the AU, not the fragmented size |
| */ |
| payload[2] = (total & 0x1fe0) >> 5; |
| payload[3] = (total & 0x1f) << 3; /* we use 13 bits for the size, 3 bits index */ |
| |
| /* marker only if the packet is complete */ |
| gst_rtp_buffer_set_marker (&rtp, avail <= payload_len); |
| |
| gst_rtp_buffer_unmap (&rtp); |
| |
| paybuf = gst_adapter_take_buffer_fast (rtpmp4gpay->adapter, payload_len); |
| gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmp4gpay), outbuf, paybuf, 0); |
| outbuf = gst_buffer_append (outbuf, paybuf); |
| |
| GST_BUFFER_PTS (outbuf) = rtpmp4gpay->first_timestamp; |
| GST_BUFFER_DURATION (outbuf) = rtpmp4gpay->first_duration; |
| |
| GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE; |
| |
| if (rtpmp4gpay->discont) { |
| GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); |
| /* Only the first outputted buffer has the DISCONT flag */ |
| rtpmp4gpay->discont = FALSE; |
| } |
| |
| ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp4gpay), outbuf); |
| |
| avail -= payload_len; |
| } |
| |
| return ret; |
| } |
| |
| /* we expect buffers as exactly one complete AU |
| */ |
| static GstFlowReturn |
| gst_rtp_mp4g_pay_handle_buffer (GstRTPBasePayload * basepayload, |
| GstBuffer * buffer) |
| { |
| GstRtpMP4GPay *rtpmp4gpay; |
| |
| rtpmp4gpay = GST_RTP_MP4G_PAY (basepayload); |
| |
| rtpmp4gpay->first_timestamp = GST_BUFFER_PTS (buffer); |
| rtpmp4gpay->first_duration = GST_BUFFER_DURATION (buffer); |
| rtpmp4gpay->discont = GST_BUFFER_IS_DISCONT (buffer); |
| |
| /* we always encode and flush a full AU */ |
| gst_adapter_push (rtpmp4gpay->adapter, buffer); |
| |
| return gst_rtp_mp4g_pay_flush (rtpmp4gpay); |
| } |
| |
| static gboolean |
| gst_rtp_mp4g_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event) |
| { |
| GstRtpMP4GPay *rtpmp4gpay; |
| |
| rtpmp4gpay = GST_RTP_MP4G_PAY (payload); |
| |
| GST_DEBUG ("Got event: %s", GST_EVENT_TYPE_NAME (event)); |
| |
| switch (GST_EVENT_TYPE (event)) { |
| case GST_EVENT_SEGMENT: |
| case GST_EVENT_EOS: |
| /* This flush call makes sure that the last buffer is always pushed |
| * to the base payloader */ |
| gst_rtp_mp4g_pay_flush (rtpmp4gpay); |
| break; |
| case GST_EVENT_FLUSH_STOP: |
| gst_rtp_mp4g_pay_reset (rtpmp4gpay); |
| break; |
| default: |
| break; |
| } |
| |
| /* let parent handle event too */ |
| return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event); |
| } |
| |
| static GstStateChangeReturn |
| gst_rtp_mp4g_pay_change_state (GstElement * element, GstStateChange transition) |
| { |
| GstStateChangeReturn ret; |
| GstRtpMP4GPay *rtpmp4gpay; |
| |
| rtpmp4gpay = GST_RTP_MP4G_PAY (element); |
| |
| switch (transition) { |
| case GST_STATE_CHANGE_READY_TO_PAUSED: |
| gst_rtp_mp4g_pay_cleanup (rtpmp4gpay); |
| break; |
| default: |
| break; |
| } |
| |
| ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); |
| |
| switch (transition) { |
| case GST_STATE_CHANGE_PAUSED_TO_READY: |
| gst_rtp_mp4g_pay_cleanup (rtpmp4gpay); |
| break; |
| default: |
| break; |
| } |
| |
| return ret; |
| } |
| |
| gboolean |
| gst_rtp_mp4g_pay_plugin_init (GstPlugin * plugin) |
| { |
| return gst_element_register (plugin, "rtpmp4gpay", |
| GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_PAY); |
| } |