X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=gst%2Frtp%2Fgstrtpamrdepay.c;h=c08c406882aabd4e685c05eb4f48c4ec180c19bb;hb=22560c473d8b1c2ed345afdbae87a9dbf5690dd0;hp=c9df817045a9400a07b14193a1c66737def3a390;hpb=27f2c9b2555be2fda77179d16cd0c19f0ee37cfa;p=platform%2Fupstream%2Fgst-plugins-good.git diff --git a/gst/rtp/gstrtpamrdepay.c b/gst/rtp/gstrtpamrdepay.c index c9df817..c08c406 100644 --- a/gst/rtp/gstrtpamrdepay.c +++ b/gst/rtp/gstrtpamrdepay.c @@ -1,5 +1,5 @@ /* GStreamer - * Copyright (C) <2005> Wim Taymans + * Copyright (C) <2005> Wim Taymans * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public @@ -9,7 +9,12 @@ * 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 + * 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 @@ -18,23 +23,20 @@ #include +#include #include #include "gstrtpamrdepay.h" +GST_DEBUG_CATEGORY_STATIC (rtpamrdepay_debug); +#define GST_CAT_DEFAULT (rtpamrdepay_debug) + /* references: * - * RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File Storage Format - * for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio - * Codecs. + * RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File + * Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate + * Wideband (AMR-WB) Audio Codecs. */ -/* elementfactory information */ -static const GstElementDetails gst_rtp_amrdepay_details = -GST_ELEMENT_DETAILS ("RTP packet parser", - "Codec/Depayr/Network", - "Extracts AMR audio from RTP packets (RFC 3267)", - "Wim Taymans "); - /* RtpAMRDepay signals and args */ enum { @@ -44,27 +46,48 @@ enum enum { - ARG_0, - ARG_FREQUENCY + ARG_0 }; -/* input is an RTP packet +/* input is an RTP packet * * params see RFC 3267, section 8.1 */ static GstStaticPadTemplate gst_rtp_amr_depay_sink_template = -GST_STATIC_PAD_TEMPLATE ("sink", + GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " + "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 8000, " "encoding-name = (string) \"AMR\", " "encoding-params = (string) \"1\", " + /* NOTE that all values must be strings in orde to be able to do SDP <-> + * GstCaps mapping. */ + "octet-align = (string) \"1\", " + "crc = (string) { \"0\", \"1\" }, " + "robust-sorting = (string) \"0\", " "interleaving = (string) \"0\";" + /* following options are not needed for a decoder + * + "mode-set = (int) [ 0, 7 ], " + "mode-change-period = (int) [ 1, MAX ], " + "mode-change-neighbor = (boolean) { TRUE, FALSE }, " + "maxptime = (int) [ 20, MAX ], " + "ptime = (int) [ 20, MAX ]" + */ + "application/x-rtp, " + "media = (string) \"audio\", " + "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " + "clock-rate = (int) 16000, " + "encoding-name = (string) \"AMR-WB\", " + "encoding-params = (string) \"1\", " + /* NOTE that all values must be strings in orde to be able to do SDP <-> + * GstCaps mapping. */ "octet-align = (string) \"1\", " "crc = (string) { \"0\", \"1\" }, " "robust-sorting = (string) \"0\", " "interleaving = (string) \"0\"" - /* following options are not needed for a decoder + /* following options are not needed for a decoder * "mode-set = (int) [ 0, 7 ], " "mode-change-period = (int) [ 1, MAX ], " @@ -76,56 +99,23 @@ GST_STATIC_PAD_TEMPLATE ("sink", ); static GstStaticPadTemplate gst_rtp_amr_depay_src_template = -GST_STATIC_PAD_TEMPLATE ("src", + GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000") + GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000;" + "audio/AMR-WB, " "channels = (int) 1," "rate = (int) 16000") ); -static void gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass); -static void gst_rtp_amr_depay_base_init (GstRtpAMRDepayClass * klass); -static void gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay); - -static gboolean gst_rtp_amr_depay_sink_setcaps (GstPad * pad, GstCaps * caps); -static GstFlowReturn gst_rtp_amr_depay_chain (GstPad * pad, GstBuffer * buffer); - -static void gst_rtp_amr_depay_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_rtp_amr_depay_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -static GstStateChangeReturn gst_rtp_amr_depay_change_state (GstElement * - element, GstStateChange transition); +static gboolean gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload, + GstCaps * caps); +static GstBuffer *gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, + GstBuffer * buf); -static GstElementClass *parent_class = NULL; - -static GType -gst_rtp_amr_depay_get_type (void) -{ - static GType rtpamrdepay_type = 0; - - if (!rtpamrdepay_type) { - static const GTypeInfo rtpamrdepay_info = { - sizeof (GstRtpAMRDepayClass), - (GBaseInitFunc) gst_rtp_amr_depay_base_init, - NULL, - (GClassInitFunc) gst_rtp_amr_depay_class_init, - NULL, - NULL, - sizeof (GstRtpAMRDepay), - 0, - (GInstanceInitFunc) gst_rtp_amr_depay_init, - }; - - rtpamrdepay_type = - g_type_register_static (GST_TYPE_ELEMENT, "GstRtpAMRDepay", - &rtpamrdepay_info, 0); - } - return rtpamrdepay_type; -} +GST_BOILERPLATE (GstRtpAMRDepay, gst_rtp_amr_depay, GstBaseRTPDepayload, + GST_TYPE_BASE_RTP_DEPAYLOAD); static void -gst_rtp_amr_depay_base_init (GstRtpAMRDepayClass * klass) +gst_rtp_amr_depay_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); @@ -134,56 +124,67 @@ gst_rtp_amr_depay_base_init (GstRtpAMRDepayClass * klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_amr_depay_sink_template)); - gst_element_class_set_details (element_class, &gst_rtp_amrdepay_details); + gst_element_class_set_details_simple (element_class, "RTP AMR depayloader", + "Codec/Depayloader/Network", + "Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)", + "Wim Taymans "); } static void gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass) { - GObjectClass *gobject_class; - GstElementClass *gstelement_class; - - gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; + GstBaseRTPDepayloadClass *gstbasertpdepayload_class; - parent_class = g_type_class_peek_parent (klass); + gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; - gobject_class->set_property = gst_rtp_amr_depay_set_property; - gobject_class->get_property = gst_rtp_amr_depay_get_property; + gstbasertpdepayload_class->process = gst_rtp_amr_depay_process; + gstbasertpdepayload_class->set_caps = gst_rtp_amr_depay_setcaps; - gstelement_class->change_state = gst_rtp_amr_depay_change_state; + GST_DEBUG_CATEGORY_INIT (rtpamrdepay_debug, "rtpamrdepay", 0, + "AMR/AMR-WB RTP Depayloader"); } static void -gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay) +gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay, + GstRtpAMRDepayClass * klass) { - rtpamrdepay->srcpad = - gst_pad_new_from_static_template (&gst_rtp_amr_depay_src_template, "src"); - - gst_element_add_pad (GST_ELEMENT (rtpamrdepay), rtpamrdepay->srcpad); - - rtpamrdepay->sinkpad = - gst_pad_new_from_static_template (&gst_rtp_amr_depay_sink_template, - "sink"); - gst_pad_set_setcaps_function (rtpamrdepay->sinkpad, - gst_rtp_amr_depay_sink_setcaps); - gst_pad_set_chain_function (rtpamrdepay->sinkpad, gst_rtp_amr_depay_chain); - gst_element_add_pad (GST_ELEMENT (rtpamrdepay), rtpamrdepay->sinkpad); + GstBaseRTPDepayload *depayload; + + depayload = GST_BASE_RTP_DEPAYLOAD (rtpamrdepay); + + gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload)); } static gboolean -gst_rtp_amr_depay_sink_setcaps (GstPad * pad, GstCaps * caps) +gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) { GstStructure *structure; GstCaps *srccaps; GstRtpAMRDepay *rtpamrdepay; const gchar *params; - const gchar *str; + const gchar *str, *type; + gint clock_rate, need_clock_rate; + gboolean res; - rtpamrdepay = GST_RTP_AMR_DEPAY (GST_OBJECT_PARENT (pad)); + rtpamrdepay = GST_RTP_AMR_DEPAY (depayload); structure = gst_caps_get_structure (caps, 0); + /* figure out the mode first and set the clock rates */ + if ((str = gst_structure_get_string (structure, "encoding-name"))) { + if (strcmp (str, "AMR") == 0) { + rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_NB; + need_clock_rate = 8000; + type = "audio/AMR"; + } else if (strcmp (str, "AMR-WB") == 0) { + rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB; + need_clock_rate = 16000; + type = "audio/AMR-WB"; + } else + goto invalid_mode; + } else + goto invalid_mode; + if (!(str = gst_structure_get_string (structure, "octet-align"))) rtpamrdepay->octet_align = FALSE; else @@ -225,14 +226,15 @@ gst_rtp_amr_depay_sink_setcaps (GstPad * pad, GstCaps * caps) rtpamrdepay->channels = atoi (params); } - if (!gst_structure_get_int (structure, "clock-rate", &rtpamrdepay->rate)) - rtpamrdepay->rate = 8000; + if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) + clock_rate = need_clock_rate; + depayload->clock_rate = clock_rate; /* we require 1 channel, 8000 Hz, octet aligned, no CRC, * no robust sorting, no interleaving for now */ if (rtpamrdepay->channels != 1) return FALSE; - if (rtpamrdepay->rate != 8000) + if (clock_rate != need_clock_rate) return FALSE; if (rtpamrdepay->octet_align != TRUE) return FALSE; @@ -241,47 +243,53 @@ gst_rtp_amr_depay_sink_setcaps (GstPad * pad, GstCaps * caps) if (rtpamrdepay->interleaving != FALSE) return FALSE; - srccaps = gst_caps_new_simple ("audio/AMR", + srccaps = gst_caps_new_simple (type, "channels", G_TYPE_INT, rtpamrdepay->channels, - "rate", G_TYPE_INT, rtpamrdepay->rate, NULL); - gst_pad_set_caps (rtpamrdepay->srcpad, srccaps); + "rate", G_TYPE_INT, clock_rate, NULL); + res = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps); gst_caps_unref (srccaps); - rtpamrdepay->negotiated = TRUE; + return res; - return TRUE; + /* ERRORS */ +invalid_mode: + { + GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name"); + return FALSE; + } } /* -1 is invalid */ -static gint frame_size[16] = { +static const gint nb_frame_size[16] = { 12, 13, 15, 17, 19, 20, 26, 31, 5, -1, -1, -1, -1, -1, -1, 0 }; -static GstFlowReturn -gst_rtp_amr_depay_chain (GstPad * pad, GstBuffer * buf) +static const gint wb_frame_size[16] = { + 17, 23, 32, 36, 40, 46, 50, 58, + 60, 5, -1, -1, -1, -1, -1, 0 +}; + +static GstBuffer * +gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) { GstRtpAMRDepay *rtpamrdepay; - GstBuffer *outbuf; - GstFlowReturn ret; + const gint *frame_size; + GstBuffer *outbuf = NULL; + gint payload_len; - rtpamrdepay = GST_RTP_AMR_DEPAY (GST_OBJECT_PARENT (pad)); + rtpamrdepay = GST_RTP_AMR_DEPAY (depayload); - if (!rtpamrdepay->negotiated) - goto not_negotiated; - - if (!gst_rtp_buffer_validate (buf)) { - GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, - (NULL), ("AMR RTP packet did not validate")); - goto bad_packet; - } + /* setup frame size pointer */ + if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB) + frame_size = nb_frame_size; + else + frame_size = wb_frame_size; - /* when we get here, 1 channel, 8000 Hz, octet aligned, no CRC, + /* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC, * no robust sorting, no interleaving data is to be depayloaded */ { - gint payload_len; guint8 *payload, *p, *dp; - guint32 timestamp; guint8 CMR; gint i, num_packets, num_nonempty_packets; gint amr_len; @@ -290,18 +298,15 @@ gst_rtp_amr_depay_chain (GstPad * pad, GstBuffer * buf) payload_len = gst_rtp_buffer_get_payload_len (buf); /* need at least 2 bytes for the header */ - if (payload_len < 2) { - GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, - (NULL), ("AMR RTP payload too small (%d)", payload_len)); - goto bad_packet; - } + if (payload_len < 2) + goto too_small; payload = gst_rtp_buffer_get_payload (buf); /* depay CMR. The CMR is used by the sender to request * a new encoding mode. * - * 0 1 2 3 4 5 6 7 + * 0 1 2 3 4 5 6 7 * +-+-+-+-+-+-+-+-+ * | CMR |R|R|R|R| * +-+-+-+-+-+-+-+-+ @@ -321,15 +326,12 @@ gst_rtp_amr_depay_chain (GstPad * pad, GstBuffer * buf) payload_len -= 1; payload += 1; - if (ILP > ILL) { - GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, - (NULL), ("AMR RTP wrong interleaving")); - goto bad_packet; - } + if (ILP > ILL) + goto wrong_interleaving; } - /* - * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 + /* + * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 * +-+-+-+-+-+-+-+-+.. * |F| FT |Q|P|P| more FT.. * +-+-+-+-+-+-+-+-+.. @@ -348,11 +350,8 @@ gst_rtp_amr_depay_chain (GstPad * pad, GstBuffer * buf) fr_size = frame_size[FT]; GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size); - if (fr_size == -1) { - GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, - (NULL), ("AMR RTP frame size == -1")); - goto bad_packet; - } + if (fr_size == -1) + goto wrong_framesize; if (fr_size > 0) { amr_len += fr_size; @@ -366,25 +365,15 @@ gst_rtp_amr_depay_chain (GstPad * pad, GstBuffer * buf) if (rtpamrdepay->crc) { /* data len + CRC len + header bytes should be smaller than payload_len */ - if (num_packets + num_nonempty_packets + amr_len > payload_len) { - GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, - (NULL), ("AMR RTP wrong length 1")); - goto bad_packet; - } + if (num_packets + num_nonempty_packets + amr_len > payload_len) + goto wrong_length_1; } else { /* data len + header bytes should be smaller than payload_len */ - if (num_packets + amr_len > payload_len) { - GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, - (NULL), ("AMR RTP wrong length 2")); - goto bad_packet; - } + if (num_packets + amr_len > payload_len) + goto wrong_length_2; } - timestamp = gst_rtp_buffer_get_timestamp (buf); - outbuf = gst_buffer_new_and_alloc (payload_len); - GST_BUFFER_TIMESTAMP (outbuf) = - gst_util_uint64_scale_int (timestamp, GST_SECOND, rtpamrdepay->rate); /* point to destination */ p = GST_BUFFER_DATA (outbuf); @@ -410,94 +399,61 @@ gst_rtp_amr_depay_chain (GstPad * pad, GstBuffer * buf) dp += fr_size; } } - gst_buffer_set_caps (outbuf, GST_PAD_CAPS (rtpamrdepay->srcpad)); + /* we can set the duration because each packet is 20 milliseconds */ + GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND; - GST_DEBUG ("gst_rtp_amr_depay_chain: pushing buffer of size %d", - GST_BUFFER_SIZE (outbuf)); - ret = gst_pad_push (rtpamrdepay->srcpad, outbuf); + if (gst_rtp_buffer_get_marker (buf)) { + /* marker bit marks a discont buffer after a talkspurt. */ + GST_DEBUG_OBJECT (depayload, "marker bit was set"); + GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); + } - gst_buffer_unref (buf); + GST_DEBUG_OBJECT (depayload, "pushing buffer of size %d", + GST_BUFFER_SIZE (outbuf)); } - - return ret; + return outbuf; /* ERRORS */ -not_negotiated: +too_small: { - GST_ELEMENT_ERROR (rtpamrdepay, STREAM, NOT_IMPLEMENTED, - (NULL), ("not negotiated")); - gst_buffer_unref (buf); - return GST_FLOW_NOT_NEGOTIATED; + GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, + (NULL), ("AMR RTP payload too small (%d)", payload_len)); + goto bad_packet; } -bad_packet: +wrong_interleaving: { - gst_buffer_unref (buf); - /* no fatal error */ - return GST_FLOW_OK; + GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, + (NULL), ("AMR RTP wrong interleaving")); + goto bad_packet; } -} - -static void -gst_rtp_amr_depay_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstRtpAMRDepay *rtpamrdepay; - - rtpamrdepay = GST_RTP_AMR_DEPAY (object); - - switch (prop_id) { - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; +wrong_framesize: + { + GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, + (NULL), ("AMR RTP frame size == -1")); + goto bad_packet; } -} - -static void -gst_rtp_amr_depay_get_property (GObject * object, guint prop_id, GValue * value, - GParamSpec * pspec) -{ - GstRtpAMRDepay *rtpamrdepay; - - rtpamrdepay = GST_RTP_AMR_DEPAY (object); - - switch (prop_id) { - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; +wrong_length_1: + { + GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, + (NULL), ("AMR RTP wrong length 1")); + goto bad_packet; } -} - -static GstStateChangeReturn -gst_rtp_amr_depay_change_state (GstElement * element, GstStateChange transition) -{ - GstRtpAMRDepay *rtpamrdepay; - GstStateChangeReturn ret; - - rtpamrdepay = GST_RTP_AMR_DEPAY (element); - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - break; - case GST_STATE_CHANGE_READY_TO_PAUSED: - break; - default: - break; +wrong_length_2: + { + GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, + (NULL), ("AMR RTP wrong length 2")); + goto bad_packet; } - - ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - switch (transition) { - case GST_STATE_CHANGE_READY_TO_NULL: - break; - default: - break; +bad_packet: + { + /* no fatal error */ + return NULL; } - return ret; } gboolean gst_rtp_amr_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpamrdepay", - GST_RANK_NONE, GST_TYPE_RTP_AMR_DEPAY); + GST_RANK_MARGINAL, GST_TYPE_RTP_AMR_DEPAY); }