diff options
Diffstat (limited to 'gst/rtp/gstrtpqcelpdepay.c')
| -rwxr-xr-x | gst/rtp/gstrtpqcelpdepay.c | 435 |
1 files changed, 435 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpqcelpdepay.c b/gst/rtp/gstrtpqcelpdepay.c new file mode 100755 index 0000000..fc88f4a --- /dev/null +++ b/gst/rtp/gstrtpqcelpdepay.c @@ -0,0 +1,435 @@ +/* GStreamer + * Copyright (C) <2010> 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 <gst/rtp/gstrtpbuffer.h> + +#include <stdlib.h> +#include <string.h> +#include "gstrtpqcelpdepay.h" + +GST_DEBUG_CATEGORY_STATIC (rtpqcelpdepay_debug); +#define GST_CAT_DEFAULT (rtpqcelpdepay_debug) + +/* references: + * + * RFC 2658 - RTP Payload Format for PureVoice(tm) Audio + */ +#define FRAME_DURATION (20 * GST_MSECOND) + +/* RtpQCELPDepay signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + PROP_0 +}; + +static GstStaticPadTemplate gst_rtp_qcelp_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) 8000, " + "encoding-name = (string) \"QCELP\"; " + "application/x-rtp, " + "media = (string) \"audio\", " + "payload = (int) " GST_RTP_PAYLOAD_QCELP_STRING ", " + "clock-rate = (int) 8000") + ); + +static GstStaticPadTemplate gst_rtp_qcelp_depay_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/qcelp, " "channels = (int) 1," "rate = (int) 8000") + ); + +static void gst_rtp_qcelp_depay_finalize (GObject * object); + +static gboolean gst_rtp_qcelp_depay_setcaps (GstRTPBaseDepayload * depayload, + GstCaps * caps); +static GstBuffer *gst_rtp_qcelp_depay_process (GstRTPBaseDepayload * depayload, + GstBuffer * buf); + +#define gst_rtp_qcelp_depay_parent_class parent_class +G_DEFINE_TYPE (GstRtpQCELPDepay, gst_rtp_qcelp_depay, + GST_TYPE_RTP_BASE_DEPAYLOAD); + +static void +gst_rtp_qcelp_depay_class_init (GstRtpQCELPDepayClass * 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_qcelp_depay_finalize; + + gstrtpbasedepayload_class->process = gst_rtp_qcelp_depay_process; + gstrtpbasedepayload_class->set_caps = gst_rtp_qcelp_depay_setcaps; + + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_rtp_qcelp_depay_src_template)); + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_rtp_qcelp_depay_sink_template)); + + gst_element_class_set_static_metadata (gstelement_class, + "RTP QCELP depayloader", "Codec/Depayloader/Network/RTP", + "Extracts QCELP (PureVoice) audio from RTP packets (RFC 2658)", + "Wim Taymans <wim.taymans@gmail.com>"); + + GST_DEBUG_CATEGORY_INIT (rtpqcelpdepay_debug, "rtpqcelpdepay", 0, + "QCELP RTP Depayloader"); +} + +static void +gst_rtp_qcelp_depay_init (GstRtpQCELPDepay * rtpqcelpdepay) +{ +} + +static void +gst_rtp_qcelp_depay_finalize (GObject * object) +{ + GstRtpQCELPDepay *depay; + + depay = GST_RTP_QCELP_DEPAY (object); + + if (depay->packets != NULL) { + g_ptr_array_foreach (depay->packets, (GFunc) gst_buffer_unref, NULL); + g_ptr_array_free (depay->packets, TRUE); + depay->packets = NULL; + } + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + + +static gboolean +gst_rtp_qcelp_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) +{ + GstCaps *srccaps; + gboolean res; + + srccaps = gst_caps_new_simple ("audio/qcelp", + "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL); + res = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps); + gst_caps_unref (srccaps); + + return res; +} + +static const gint frame_size[16] = { + 1, 4, 8, 17, 35, -8, 0, 0, + 0, 0, 0, 0, 0, 0, 1, 0 +}; + +/* get the frame length, 0 is invalid, negative values are invalid but can be + * recovered from. */ +static gint +get_frame_len (GstRtpQCELPDepay * depay, guint8 frame_type) +{ + if (frame_type >= G_N_ELEMENTS (frame_size)) + return 0; + + return frame_size[frame_type]; +} + +static guint +count_packets (GstRtpQCELPDepay * depay, guint8 * data, guint size) +{ + guint count = 0; + + while (size > 0) { + gint frame_len; + + frame_len = get_frame_len (depay, data[0]); + + /* 0 is invalid and we throw away the remainder of the frames */ + if (frame_len == 0) + break; + + if (frame_len < 0) + frame_len = -frame_len; + + if (frame_len > size) + break; + + size -= frame_len; + data += frame_len; + count++; + } + return count; +} + +static void +flush_packets (GstRtpQCELPDepay * depay) +{ + guint i, size; + + GST_DEBUG_OBJECT (depay, "flushing packets"); + + size = depay->packets->len; + + for (i = 0; i < size; i++) { + GstBuffer *outbuf; + + outbuf = g_ptr_array_index (depay->packets, i); + g_ptr_array_index (depay->packets, i) = NULL; + + gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (depay), outbuf); + } + + /* and reset interleaving state */ + depay->interleaved = FALSE; + depay->bundling = 0; +} + +static void +add_packet (GstRtpQCELPDepay * depay, guint LLL, guint NNN, guint index, + GstBuffer * outbuf) +{ + guint idx; + GstBuffer *old; + + /* figure out the position in the array, note that index is never 0 because we + * push those packets immediately. */ + idx = NNN + ((LLL + 1) * (index - 1)); + + GST_DEBUG_OBJECT (depay, "adding packet at index %u", idx); + /* free old buffer (should not happen) */ + old = g_ptr_array_index (depay->packets, idx); + if (old) + gst_buffer_unref (old); + + /* store new buffer */ + g_ptr_array_index (depay->packets, idx) = outbuf; +} + +static GstBuffer * +create_erasure_buffer (GstRtpQCELPDepay * depay) +{ + GstBuffer *outbuf; + GstMapInfo map; + + outbuf = gst_buffer_new_and_alloc (1); + gst_buffer_map (outbuf, &map, GST_MAP_WRITE); + map.data[0] = 14; + gst_buffer_unmap (outbuf, &map); + + return outbuf; +} + +static GstBuffer * +gst_rtp_qcelp_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf) +{ + GstRtpQCELPDepay *depay; + GstBuffer *outbuf; + GstClockTime timestamp; + guint payload_len, offset, index; + guint8 *payload; + guint LLL, NNN; + GstRTPBuffer rtp = { NULL }; + + depay = GST_RTP_QCELP_DEPAY (depayload); + + gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp); + + payload_len = gst_rtp_buffer_get_payload_len (&rtp); + + if (payload_len < 2) + goto too_small; + + timestamp = GST_BUFFER_TIMESTAMP (buf); + + payload = gst_rtp_buffer_get_payload (&rtp); + + /* 0 1 2 3 4 5 6 7 + * +-+-+-+-+-+-+-+-+ + * |RR | LLL | NNN | + * +-+-+-+-+-+-+-+-+ + */ + /* RR = payload[0] >> 6; */ + LLL = (payload[0] & 0x38) >> 3; + NNN = (payload[0] & 0x07); + + payload_len--; + payload++; + + GST_DEBUG_OBJECT (depay, "LLL %u, NNN %u", LLL, NNN); + + if (LLL > 5) + goto invalid_lll; + + if (NNN > LLL) + goto invalid_nnn; + + if (LLL != 0) { + /* we are interleaved */ + if (!depay->interleaved) { + guint size; + + GST_DEBUG_OBJECT (depay, "starting interleaving group"); + /* bundling is not allowed to change in one interleave group */ + depay->bundling = count_packets (depay, payload, payload_len); + GST_DEBUG_OBJECT (depay, "got bundling of %u", depay->bundling); + /* we have one bundle where NNN goes from 0 to L, we don't store the index + * 0 frames, so L+1 packets. Each packet has 'bundling - 1' packets */ + size = (depay->bundling - 1) * (LLL + 1); + /* create the array to hold the packets */ + if (depay->packets == NULL) + depay->packets = g_ptr_array_sized_new (size); + GST_DEBUG_OBJECT (depay, "created packet array of size %u", size); + g_ptr_array_set_size (depay->packets, size); + /* we were previously not interleaved, figure out how much space we + * need to deinterleave */ + depay->interleaved = TRUE; + } + } else { + /* we are not interleaved */ + if (depay->interleaved) { + GST_DEBUG_OBJECT (depay, "stopping interleaving"); + /* flush packets if we were previously interleaved */ + flush_packets (depay); + } + depay->bundling = 0; + } + + index = 0; + offset = 1; + + while (payload_len > 0) { + gint frame_len; + gboolean do_erasure; + + frame_len = get_frame_len (depay, payload[0]); + GST_DEBUG_OBJECT (depay, "got frame len %d", frame_len); + + if (frame_len == 0) + goto invalid_frame; + + if (frame_len < 0) { + /* need to add an erasure frame but we can recover */ + frame_len = -frame_len; + do_erasure = TRUE; + } else { + do_erasure = FALSE; + } + + if (frame_len > payload_len) + goto invalid_frame; + + if (do_erasure) { + /* create erasure frame */ + outbuf = create_erasure_buffer (depay); + } else { + /* each frame goes into its buffer */ + outbuf = gst_rtp_buffer_get_payload_subbuffer (&rtp, offset, frame_len); + } + + GST_BUFFER_TIMESTAMP (outbuf) = timestamp; + GST_BUFFER_DURATION (outbuf) = FRAME_DURATION; + + if (!depay->interleaved || index == 0) { + /* not interleaved or first frame in packet, just push */ + gst_rtp_base_depayload_push (depayload, outbuf); + + if (timestamp != -1) + timestamp += FRAME_DURATION; + } else { + /* put in interleave buffer */ + add_packet (depay, LLL, NNN, index, outbuf); + + if (timestamp != -1) + timestamp += (FRAME_DURATION * (LLL + 1)); + } + + payload_len -= frame_len; + payload += frame_len; + offset += frame_len; + index++; + + /* discard excess packets */ + if (depay->bundling > 0 && depay->bundling <= index) + break; + } + while (index < depay->bundling) { + GST_DEBUG_OBJECT (depay, "filling with erasure buffer"); + /* fill remainder with erasure packets */ + outbuf = create_erasure_buffer (depay); + add_packet (depay, LLL, NNN, index, outbuf); + index++; + } + if (depay->interleaved && LLL == NNN) { + GST_DEBUG_OBJECT (depay, "interleave group ended, flushing"); + /* we have the complete interleave group, flush */ + flush_packets (depay); + } + + gst_rtp_buffer_unmap (&rtp); + return NULL; + + /* ERRORS */ +too_small: + { + GST_ELEMENT_WARNING (depay, STREAM, DECODE, + (NULL), ("QCELP RTP payload too small (%d)", payload_len)); + gst_rtp_buffer_unmap (&rtp); + return NULL; + } +invalid_lll: + { + GST_ELEMENT_WARNING (depay, STREAM, DECODE, + (NULL), ("QCELP RTP invalid LLL received (%d)", LLL)); + gst_rtp_buffer_unmap (&rtp); + return NULL; + } +invalid_nnn: + { + GST_ELEMENT_WARNING (depay, STREAM, DECODE, + (NULL), ("QCELP RTP invalid NNN received (%d)", NNN)); + gst_rtp_buffer_unmap (&rtp); + return NULL; + } +invalid_frame: + { + GST_ELEMENT_WARNING (depay, STREAM, DECODE, + (NULL), ("QCELP RTP invalid frame received")); + gst_rtp_buffer_unmap (&rtp); + return NULL; + } +} + +gboolean +gst_rtp_qcelp_depay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpqcelpdepay", + GST_RANK_SECONDARY, GST_TYPE_RTP_QCELP_DEPAY); +} |