diff options
Diffstat (limited to 'gst/rtp/gstrtpmp4gpay.c')
| -rw-r--r-- | gst/rtp/gstrtpmp4gpay.c | 631 |
1 files changed, 631 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpmp4gpay.c b/gst/rtp/gstrtpmp4gpay.c new file mode 100644 index 0000000..4100ab7 --- /dev/null +++ b/gst/rtp/gstrtpmp4gpay.c @@ -0,0 +1,631 @@ +/* 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, 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" + +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 (GstBaseRTPPayload * payload, + GstCaps * caps); +static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * + payload, GstBuffer * buffer); +static gboolean gst_rtp_mp4g_pay_handle_event (GstPad * pad, GstEvent * event); + +GST_BOILERPLATE (GstRtpMP4GPay, gst_rtp_mp4g_pay, GstBaseRTPPayload, + GST_TYPE_BASE_RTP_PAYLOAD) + + static void gst_rtp_mp4g_pay_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_add_static_pad_template (element_class, + &gst_rtp_mp4g_pay_src_template); + gst_element_class_add_static_pad_template (element_class, + &gst_rtp_mp4g_pay_sink_template); + + gst_element_class_set_details_simple (element_class, "RTP MPEG4 ES payloader", + "Codec/Payloader/Network/RTP", + "Payload MPEG4 elementary streams as RTP packets (RFC 3640)", + "Wim Taymans <wim.taymans@gmail.com>"); +} + +static void +gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseRTPPayloadClass *gstbasertppayload_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; + + gobject_class->finalize = gst_rtp_mp4g_pay_finalize; + + gstelement_class->change_state = gst_rtp_mp4g_pay_change_state; + + gstbasertppayload_class->set_caps = gst_rtp_mp4g_pay_setcaps; + gstbasertppayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer; + gstbasertppayload_class->handle_event = gst_rtp_mp4g_pay_handle_event; + + GST_DEBUG_CATEGORY_INIT (rtpmp4gpay_debug, "rtpmp4gpay", 0, + "MP4-generic RTP Payloader"); +} + +static void +gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay, GstRtpMP4GPayClass * klass) +{ + rtpmp4gpay->adapter = gst_adapter_new (); +} + +static void +gst_rtp_mp4g_pay_reset (GstRtpMP4GPay * rtpmp4gpay) +{ + GST_DEBUG_OBJECT (rtpmp4gpay, "reset"); + + gst_adapter_clear (rtpmp4gpay->adapter); + rtpmp4gpay->offset = 0; +} + +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) +{ + guint8 *data; + guint size; + guint8 objectType = 0; + guint8 samplingIdx = 0; + guint8 channelCfg = 0; + GstBitReader br; + + data = GST_BUFFER_DATA (buffer); + size = GST_BUFFER_SIZE (buffer); + + gst_bit_reader_init (&br, data, 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); + + return TRUE; + + /* ERROR */ +too_short: + { + GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT, + (NULL), ("config string too short")); + return FALSE; + } +invalid_object: + { + GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT, + (NULL), ("invalid object type")); + return FALSE; + } +wrong_freq: + { + GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED, + (NULL), ("unsupported frequency index %d", samplingIdx)); + return FALSE; + } +wrong_channels: + { + GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED, + (NULL), ("unsupported number of channels %d, must < 8", channelCfg)); + return FALSE; + } +} + +#define VOS_STARTCODE 0x000001B0 + +static gboolean +gst_rtp_mp4g_pay_parse_video_config (GstRtpMP4GPay * rtpmp4gpay, + GstBuffer * buffer) +{ + guint8 *data; + guint size; + guint32 code; + + data = GST_BUFFER_DATA (buffer); + size = GST_BUFFER_SIZE (buffer); + + if (size < 5) + goto too_short; + + code = GST_READ_UINT32_BE (data); + + g_free (rtpmp4gpay->profile); + if (code == VOS_STARTCODE) { + /* get profile */ + rtpmp4gpay->profile = g_strdup_printf ("%d", (gint) 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); + + return TRUE; + + /* ERROR */ +too_short: + { + GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT, + (NULL), ("config string too short")); + 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_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), + "encoding-params", G_TYPE_STRING, rtpmp4gpay->params, MP4GCAPS); + } else { + res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), + MP4GCAPS); + } + + g_value_unset (&v); + g_free (config); + +#undef MP4GCAPS + return res; +} + +static gboolean +gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) +{ + GstRtpMP4GPay *rtpmp4gpay; + GstStructure *structure; + const GValue *codec_data; + const gchar *media_type = NULL; + gboolean res; + + 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; + const gchar *name; + + 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); + } + } + if (media_type == NULL) + goto config_failed; + + gst_basertppayload_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_BASE_RTP_PAYLOAD_MTU (rtpmp4gpay); + + while (avail > 0) { + guint towrite; + guint8 *payload; + guint payload_len; + guint packet_len; + + /* 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 (payload_len + 4, 0, 0); + + /* copy payload */ + payload = gst_rtp_buffer_get_payload (outbuf); + + /* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+ + * |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 */ + + /* copy stuff from adapter to payload */ + gst_adapter_copy (rtpmp4gpay->adapter, &payload[4], 0, payload_len); + gst_adapter_flush (rtpmp4gpay->adapter, payload_len); + + /* marker only if the packet is complete */ + gst_rtp_buffer_set_marker (outbuf, avail <= payload_len); + + GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_timestamp; + GST_BUFFER_DURATION (outbuf) = rtpmp4gpay->first_duration; + + if (rtpmp4gpay->frame_len) { + GST_BUFFER_OFFSET (outbuf) = rtpmp4gpay->offset; + rtpmp4gpay->offset += rtpmp4gpay->frame_len; + } + + ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf); + + avail -= payload_len; + } + + return ret; +} + +/* we expect buffers as exactly one complete AU + */ +static GstFlowReturn +gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * basepayload, + GstBuffer * buffer) +{ + GstRtpMP4GPay *rtpmp4gpay; + + rtpmp4gpay = GST_RTP_MP4G_PAY (basepayload); + + rtpmp4gpay->first_timestamp = GST_BUFFER_TIMESTAMP (buffer); + rtpmp4gpay->first_duration = GST_BUFFER_DURATION (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_handle_event (GstPad * pad, GstEvent * event) +{ + GstRtpMP4GPay *rtpmp4gpay; + + rtpmp4gpay = GST_RTP_MP4G_PAY (gst_pad_get_parent (pad)); + + GST_DEBUG ("Got event: %s", GST_EVENT_TYPE_NAME (event)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_NEWSEGMENT: + 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; + } + + g_object_unref (rtpmp4gpay); + + /* let parent handle event too */ + return FALSE; +} + +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); +} |