/* GStreamer * Copyright (C) <2007> Nokia Corporation (contact <stefan.kost@nokia.com>) * <2007> 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 version 2 as published by the Free Software Foundation. * * 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 <gst/base/gstbitreader.h> #include <gst/rtp/gstrtpbuffer.h> #include <gst/audio/audio.h> #include <string.h> #include "gstrtpelements.h" #include "gstrtpmp4adepay.h" #include "gstrtputils.h" GST_DEBUG_CATEGORY_STATIC (rtpmp4adepay_debug); #define GST_CAT_DEFAULT (rtpmp4adepay_debug) static GstStaticPadTemplate gst_rtp_mp4a_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/mpeg," "mpegversion = (int) 4," "framed = (boolean) { false, true }, " "stream-format = (string) raw") ); static GstStaticPadTemplate gst_rtp_mp4a_depay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"MP4A-LATM\"" /* All optional parameters * * "profile-level-id=[1,MAX]" * "config=" */ ) ); #define gst_rtp_mp4a_depay_parent_class parent_class G_DEFINE_TYPE (GstRtpMP4ADepay, gst_rtp_mp4a_depay, GST_TYPE_RTP_BASE_DEPAYLOAD); GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpmp4adepay, "rtpmp4adepay", GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_DEPAY, rtp_element_init (plugin)); static void gst_rtp_mp4a_depay_finalize (GObject * object); static gboolean gst_rtp_mp4a_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp); static GstStateChangeReturn gst_rtp_mp4a_depay_change_state (GstElement * element, GstStateChange transition); static void gst_rtp_mp4a_depay_class_init (GstRtpMP4ADepayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstRTPBaseDepayloadClass *gstrtpbasedepayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass; gobject_class->finalize = gst_rtp_mp4a_depay_finalize; gstelement_class->change_state = gst_rtp_mp4a_depay_change_state; gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_mp4a_depay_process; gstrtpbasedepayload_class->set_caps = gst_rtp_mp4a_depay_setcaps; gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_mp4a_depay_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_mp4a_depay_sink_template); gst_element_class_set_static_metadata (gstelement_class, "RTP MPEG4 audio depayloader", "Codec/Depayloader/Network/RTP", "Extracts MPEG4 audio from RTP packets (RFC 3016)", "Nokia Corporation (contact <stefan.kost@nokia.com>), " "Wim Taymans <wim.taymans@gmail.com>"); GST_DEBUG_CATEGORY_INIT (rtpmp4adepay_debug, "rtpmp4adepay", 0, "MPEG4 audio RTP Depayloader"); } static void gst_rtp_mp4a_depay_init (GstRtpMP4ADepay * rtpmp4adepay) { rtpmp4adepay->adapter = gst_adapter_new (); rtpmp4adepay->framed = FALSE; } static void gst_rtp_mp4a_depay_finalize (GObject * object) { GstRtpMP4ADepay *rtpmp4adepay; rtpmp4adepay = GST_RTP_MP4A_DEPAY (object); g_object_unref (rtpmp4adepay->adapter); rtpmp4adepay->adapter = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350 }; static gboolean gst_rtp_mp4a_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) { GstStructure *structure; GstRtpMP4ADepay *rtpmp4adepay; GstCaps *srccaps; const gchar *str; gint clock_rate; gint object_type; gint channels = 2; /* default */ gboolean res; rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload); rtpmp4adepay->framed = FALSE; structure = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) clock_rate = 90000; /* default */ depayload->clock_rate = clock_rate; if (!gst_structure_get_int (structure, "object", &object_type)) object_type = 2; /* AAC LC default */ srccaps = gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 4, "framed", G_TYPE_BOOLEAN, FALSE, "channels", G_TYPE_INT, channels, "stream-format", G_TYPE_STRING, "raw", NULL); if ((str = gst_structure_get_string (structure, "config"))) { GValue v = { 0 }; g_value_init (&v, GST_TYPE_BUFFER); if (gst_value_deserialize (&v, str)) { GstBuffer *buffer; GstMapInfo map; guint8 *data; gsize size; gint i; guint32 rate = 0; guint8 obj_type = 0, sr_idx = 0, channels = 0; GstBitReader br; buffer = gst_value_get_buffer (&v); gst_buffer_ref (buffer); g_value_unset (&v); gst_buffer_map (buffer, &map, GST_MAP_READ); data = map.data; size = map.size; if (size < 2) { GST_WARNING_OBJECT (depayload, "config too short (%d < 2)", (gint) size); goto bad_config; } /* Parse StreamMuxConfig according to ISO/IEC 14496-3: * * audioMuxVersion == 0 (1 bit) * allStreamsSameTimeFraming == 1 (1 bit) * numSubFrames == rtpmp4adepay->numSubFrames (6 bits) * numProgram == 0 (4 bits) * numLayer == 0 (3 bits) * * We only require audioMuxVersion == 0; * * The remaining bit of the second byte and the rest of the bits are used * for audioSpecificConfig which we need to set in codec_info. */ if ((data[0] & 0x80) != 0x00) { GST_WARNING_OBJECT (depayload, "unknown audioMuxVersion 1"); goto bad_config; } rtpmp4adepay->numSubFrames = (data[0] & 0x3F); GST_LOG_OBJECT (rtpmp4adepay, "numSubFrames %d", rtpmp4adepay->numSubFrames); /* shift rest of string 15 bits down */ size -= 2; for (i = 0; i < size; i++) { data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1); } 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, &obj_type, 5)) goto bad_config; if (obj_type == 0) { GST_WARNING_OBJECT (depayload, "invalid object type 0"); goto bad_config; } if (!gst_bit_reader_get_bits_uint8 (&br, &sr_idx, 4)) goto bad_config; if (sr_idx >= G_N_ELEMENTS (aac_sample_rates) && sr_idx != 15) { GST_WARNING_OBJECT (depayload, "invalid sample rate index %d", sr_idx); goto bad_config; } GST_LOG_OBJECT (rtpmp4adepay, "sample rate index %u", sr_idx); if (!gst_bit_reader_get_bits_uint8 (&br, &channels, 4)) goto bad_config; if (channels > 7) { GST_WARNING_OBJECT (depayload, "invalid channels %u", (guint) channels); goto bad_config; } /* rtp rate depends on sampling rate of the audio */ if (sr_idx == 15) { /* index of 15 means we get the rate in the next 24 bits */ if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24)) goto bad_config; } else if (sr_idx >= G_N_ELEMENTS (aac_sample_rates)) { goto bad_config; } else { /* else use the rate from the table */ rate = aac_sample_rates[sr_idx]; } rtpmp4adepay->frame_len = 1024; switch (obj_type) { 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) rtpmp4adepay->frame_len = 960; break; } default: break; } /* ignore remaining bit, we're only interested in full bytes */ gst_buffer_resize (buffer, 0, size); gst_buffer_unmap (buffer, &map); data = NULL; gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, (gint) channels, "rate", G_TYPE_INT, (gint) rate, "codec_data", GST_TYPE_BUFFER, buffer, NULL); bad_config: if (data) gst_buffer_unmap (buffer, &map); gst_buffer_unref (buffer); } else { g_warning ("cannot convert config to buffer"); } } res = gst_pad_set_caps (depayload->srcpad, srccaps); gst_caps_unref (srccaps); return res; } static GstBuffer * gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp) { GstRtpMP4ADepay *rtpmp4adepay; GstBuffer *outbuf; GstMapInfo map; rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload); /* flush remaining data on discont */ if (GST_BUFFER_IS_DISCONT (rtp->buffer)) { gst_adapter_clear (rtpmp4adepay->adapter); } outbuf = gst_rtp_buffer_get_payload_buffer (rtp); if (!rtpmp4adepay->framed) { if (gst_rtp_buffer_get_marker (rtp)) { GstCaps *caps; rtpmp4adepay->framed = TRUE; gst_rtp_base_depayload_push (depayload, outbuf); caps = gst_pad_get_current_caps (depayload->srcpad); caps = gst_caps_make_writable (caps); gst_caps_set_simple (caps, "framed", G_TYPE_BOOLEAN, TRUE, NULL); gst_pad_set_caps (depayload->srcpad, caps); gst_caps_unref (caps); return NULL; } else { return outbuf; } } outbuf = gst_buffer_make_writable (outbuf); GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (rtp->buffer); gst_adapter_push (rtpmp4adepay->adapter, outbuf); /* RTP marker bit indicates the last packet of the AudioMuxElement => create * and push a buffer */ if (gst_rtp_buffer_get_marker (rtp)) { guint avail; guint i; guint8 *data; guint pos; GstClockTime timestamp; avail = gst_adapter_available (rtpmp4adepay->adapter); timestamp = gst_adapter_prev_pts (rtpmp4adepay->adapter, NULL); GST_LOG_OBJECT (rtpmp4adepay, "have marker and %u available", avail); outbuf = gst_adapter_take_buffer (rtpmp4adepay->adapter, avail); gst_buffer_map (outbuf, &map, GST_MAP_READ); data = map.data; /* position in data we are at */ pos = 0; /* looping through the number of sub-frames in the audio payload */ for (i = 0; i <= rtpmp4adepay->numSubFrames; i++) { /* determine payload length and set buffer data pointer accordingly */ guint skip; guint data_len; GstBuffer *tmp = NULL; /* each subframe starts with a variable length encoding */ data_len = 0; for (skip = 0; skip < avail; skip++) { data_len += data[skip]; if (data[skip] != 0xff) break; } skip++; /* this can not be possible, we have not enough data or the length * decoding failed because we ran out of data. */ if (skip + data_len > avail) goto wrong_size; GST_LOG_OBJECT (rtpmp4adepay, "subframe %u, header len %u, data len %u, left %u", i, skip, data_len, avail); /* take data out, skip the header */ pos += skip; tmp = gst_buffer_copy_region (outbuf, GST_BUFFER_COPY_ALL, pos, data_len); /* skip data too */ skip += data_len; pos += data_len; /* update our pointers with what we consumed */ data += skip; avail -= skip; GST_BUFFER_PTS (tmp) = timestamp; gst_rtp_drop_non_audio_meta (depayload, tmp); gst_rtp_base_depayload_push (depayload, tmp); /* shift ts for next buffers */ if (rtpmp4adepay->frame_len && timestamp != -1 && depayload->clock_rate != 0) { timestamp += gst_util_uint64_scale_int (rtpmp4adepay->frame_len, GST_SECOND, depayload->clock_rate); } } /* just a check that lengths match */ if (avail) { GST_ELEMENT_WARNING (depayload, STREAM, DECODE, ("Packet invalid"), ("Not all payload consumed: " "possible wrongly encoded packet.")); } gst_buffer_unmap (outbuf, &map); gst_buffer_unref (outbuf); } return NULL; /* ERRORS */ wrong_size: { GST_ELEMENT_WARNING (rtpmp4adepay, STREAM, DECODE, ("Packet did not validate"), ("wrong packet size")); gst_buffer_unmap (outbuf, &map); gst_buffer_unref (outbuf); return NULL; } } static GstStateChangeReturn gst_rtp_mp4a_depay_change_state (GstElement * element, GstStateChange transition) { GstRtpMP4ADepay *rtpmp4adepay; GstStateChangeReturn ret; rtpmp4adepay = GST_RTP_MP4A_DEPAY (element); switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: gst_adapter_clear (rtpmp4adepay->adapter); rtpmp4adepay->frame_len = 0; rtpmp4adepay->numSubFrames = 0; rtpmp4adepay->framed = FALSE; break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { default: break; } return ret; }