diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am index 5117a7b1eb..b6aefe6c44 100644 --- a/gst/rtp/Makefile.am +++ b/gst/rtp/Makefile.am @@ -16,6 +16,7 @@ libgstrtp_la_SOURCES = \ gstrtpilbcpay.c \ gstrtpmpadepay.c \ gstrtpmpapay.c \ + gstrtpmparobustdepay.c \ gstrtpmpvdepay.c \ gstrtpmpvpay.c \ gstrtppcmadepay.c \ @@ -114,6 +115,7 @@ noinst_HEADERS = \ gstrtpgsmdepay.h \ gstrtpgsmpay.h \ gstrtpmpadepay.h \ + gstrtpmparobustdepay.h \ gstrtpmpapay.h \ gstrtpmpvdepay.h \ gstrtpmpvpay.h \ diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c index b35e854adf..852906efa6 100644 --- a/gst/rtp/gstrtp.c +++ b/gst/rtp/gstrtp.c @@ -47,6 +47,7 @@ #include "gstrtpamrdepay.h" #include "gstrtpmpapay.h" #include "gstrtpmpadepay.h" +#include "gstrtpmparobustdepay.h" #include "gstrtpmpvdepay.h" #include "gstrtpmpvpay.h" #include "gstrtph263pdepay.h" @@ -166,6 +167,9 @@ plugin_init (GstPlugin * plugin) if (!gst_rtp_mpa_pay_plugin_init (plugin)) return FALSE; + if (!gst_rtp_mpa_robust_depay_plugin_init (plugin)) + return FALSE; + if (!gst_rtp_mpv_depay_plugin_init (plugin)) return FALSE; diff --git a/gst/rtp/gstrtpmparobustdepay.c b/gst/rtp/gstrtpmparobustdepay.c new file mode 100644 index 0000000000..e50b5cfb25 --- /dev/null +++ b/gst/rtp/gstrtpmparobustdepay.c @@ -0,0 +1,782 @@ +/* GStreamer + * Copyright (C) <2010> Mark Nauwelaerts + * Copyright (C) <2010> Nokia Corporation + * + * 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 + +#include +#include "gstrtpmparobustdepay.h" + +GST_DEBUG_CATEGORY_STATIC (rtpmparobustdepay_debug); +#define GST_CAT_DEFAULT (rtpmparobustdepay_debug) + +static GstStaticPadTemplate gst_rtp_mpa_robust_depay_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1") + ); + +static GstStaticPadTemplate gst_rtp_mpa_robust_depay_sink_template = + 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) 90000, " + "encoding-name = (string) \"MPA-ROBUST\" " "; " + /* draft versions appear still in use out there */ + "application/x-rtp, " + "media = (string) \"audio\", " + "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " + "clock-rate = (int) [1, MAX], " + "encoding-name = (string) { \"X-MP3-DRAFT-00\", \"X-MP3-DRAFT-01\", " + " \"X-MP3-DRAFT-02\", \"X-MP3-DRAFT-03\", \"X-MP3-DRAFT-04\", " + " \"X-MP3-DRAFT-05\", \"X-MP3-DRAFT-06\" }") + ); + +typedef struct _GstADUFrame +{ + guint32 header; + gint size; + gint side_info; + gint data_size; + gint layer; + gint backpointer; + + GstBuffer *buffer; +} GstADUFrame; + +GST_BOILERPLATE (GstRtpMPARobustDepay, gst_rtp_mpa_robust_depay, + GstBaseRTPDepayload, GST_TYPE_BASE_RTP_DEPAYLOAD); + +static GstStateChangeReturn gst_rtp_mpa_robust_change_state (GstElement * + element, GstStateChange transition); + +static gboolean gst_rtp_mpa_robust_depay_setcaps (GstBaseRTPDepayload * + depayload, GstCaps * caps); +static GstBuffer *gst_rtp_mpa_robust_depay_process (GstBaseRTPDepayload * + depayload, GstBuffer * buf); + +static void +gst_rtp_mpa_robust_depay_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_mpa_robust_depay_src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_mpa_robust_depay_sink_template)); + + gst_element_class_set_details_simple (element_class, + "RTP MPEG audio depayloader", "Codec/Depayloader/Network", + "Extracts MPEG audio from RTP packets (RFC 5219)", + "Mark Nauwelaerts "); +} + +static void +gst_rtp_mpa_robust_depay_finalize (GObject * object) +{ + GstRtpMPARobustDepay *rtpmpadepay; + + rtpmpadepay = (GstRtpMPARobustDepay *) object; + + g_object_unref (rtpmpadepay->adapter); + g_queue_free (rtpmpadepay->adu_frames); + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + + +static void +gst_rtp_mpa_robust_depay_class_init (GstRtpMPARobustDepayClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseRTPDepayloadClass *gstbasertpdepayload_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; + + gobject_class->finalize = gst_rtp_mpa_robust_depay_finalize; + + gstelement_class->change_state = + GST_DEBUG_FUNCPTR (gst_rtp_mpa_robust_change_state); + + gstbasertpdepayload_class->set_caps = gst_rtp_mpa_robust_depay_setcaps; + gstbasertpdepayload_class->process = gst_rtp_mpa_robust_depay_process; + + GST_DEBUG_CATEGORY_INIT (rtpmparobustdepay_debug, "rtpmparobustdepay", 0, + "Robust MPEG Audio RTP Depayloader"); +} + +static void +gst_rtp_mpa_robust_depay_init (GstRtpMPARobustDepay * rtpmpadepay, + GstRtpMPARobustDepayClass * klass) +{ + rtpmpadepay->adapter = gst_adapter_new (); + rtpmpadepay->adu_frames = g_queue_new (); +} + +static gboolean +gst_rtp_mpa_robust_depay_setcaps (GstBaseRTPDepayload * depayload, + GstCaps * caps) +{ + GstRtpMPARobustDepay *rtpmpadepay; + GstStructure *structure; + GstCaps *outcaps; + gint clock_rate, draft; + gboolean res; + const gchar *encoding; + + rtpmpadepay = GST_RTP_MPA_ROBUST_DEPAY (depayload); + + structure = gst_caps_get_structure (caps, 0); + + if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) + clock_rate = 90000; + depayload->clock_rate = clock_rate; + + rtpmpadepay->has_descriptor = TRUE; + if ((encoding = gst_structure_get_string (structure, "encoding-name"))) { + if (sscanf (encoding, "X-MP3-DRAFT-%d", &draft) && (draft == 0)) + rtpmpadepay->has_descriptor = FALSE; + } + + outcaps = + gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 1, NULL); + res = gst_pad_set_caps (depayload->srcpad, outcaps); + gst_caps_unref (outcaps); + + return res; +} + +/* thanks again go to mp3parse ... */ + +static const guint mp3types_bitrates[2][3][16] = { + { + {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,}, + {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,}, + {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,} + }, + { + {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,}, + {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}, + {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,} + }, +}; + +static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000}, +{22050, 24000, 16000}, +{11025, 12000, 8000} +}; + +static inline guint +mp3_type_frame_length_from_header (GstElement * mp3parse, guint32 header, + guint * put_version, guint * put_layer, guint * put_channels, + guint * put_bitrate, guint * put_samplerate, guint * put_mode, + guint * put_crc) +{ + guint length; + gulong mode, samplerate, bitrate, layer, channels, padding, crc; + gulong version; + gint lsf, mpg25; + + if (header & (1 << 20)) { + lsf = (header & (1 << 19)) ? 0 : 1; + mpg25 = 0; + } else { + lsf = 1; + mpg25 = 1; + } + + version = 1 + lsf + mpg25; + + layer = 4 - ((header >> 17) & 0x3); + + crc = (header >> 16) & 0x1; + + bitrate = (header >> 12) & 0xF; + bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000; + /* The caller has ensured we have a valid header, so bitrate can't be + zero here. */ + if (bitrate == 0) + return 0; + + samplerate = (header >> 10) & 0x3; + samplerate = mp3types_freqs[lsf + mpg25][samplerate]; + + padding = (header >> 9) & 0x1; + + mode = (header >> 6) & 0x3; + channels = (mode == 3) ? 1 : 2; + + switch (layer) { + case 1: + length = 4 * ((bitrate * 12) / samplerate + padding); + break; + case 2: + length = (bitrate * 144) / samplerate + padding; + break; + default: + case 3: + length = (bitrate * 144) / (samplerate << lsf) + padding; + break; + } + + GST_LOG_OBJECT (mp3parse, "Calculated mp3 frame length of %u bytes", length); + GST_LOG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, version = %lu, " + "layer = %lu, channels = %lu, mode = %lu", samplerate, bitrate, version, + layer, channels, mode); + + if (put_version) + *put_version = version; + if (put_layer) + *put_layer = layer; + if (put_channels) + *put_channels = channels; + if (put_bitrate) + *put_bitrate = bitrate; + if (put_samplerate) + *put_samplerate = samplerate; + if (put_mode) + *put_mode = mode; + if (put_crc) + *put_crc = crc; + + return length; +} + +/* generate empty/silent/dummy frame that mimics @frame, + * except for rate, where maximum possible is selected */ +static GstADUFrame * +gst_rtp_mpa_robust_depay_generate_dummy_frame (GstRtpMPARobustDepay * + rtpmpadepay, GstADUFrame * frame) +{ + GstADUFrame *dummy; + + dummy = g_slice_dup (GstADUFrame, frame); + + /* go for maximum bitrate */ + dummy->header = frame->header | (0xf << 12); + dummy->size = + mp3_type_frame_length_from_header (GST_ELEMENT_CAST (rtpmpadepay), + dummy->header, NULL, NULL, NULL, NULL, NULL, NULL, NULL); + dummy->data_size = dummy->size - dummy->side_info; + dummy->backpointer = 0; + + dummy->buffer = gst_buffer_new_and_alloc (dummy->size); + memset (GST_BUFFER_DATA (dummy->buffer), 0, dummy->size); + GST_WRITE_UINT32_BE (GST_BUFFER_DATA (dummy->buffer), dummy->header); + GST_BUFFER_TIMESTAMP (dummy->buffer) = GST_BUFFER_TIMESTAMP (frame->buffer); + + return dummy; +} + +/* validates and parses @buf, and queues for further transformation if valid, + * otherwise discards @buf + * Takes ownership of @buf. */ +static gboolean +gst_rtp_mpa_robust_depay_queue_frame (GstRtpMPARobustDepay * rtpmpadepay, + GstBuffer * buf) +{ + GstADUFrame *frame = NULL; + guint version, layer, channels, size; + guint crc; + + g_return_val_if_fail (buf != NULL, FALSE); + + if (GST_BUFFER_SIZE (buf) < 6) { + goto corrupt_frame; + } + + frame = g_slice_new0 (GstADUFrame); + frame->header = GST_READ_UINT32_BE (GST_BUFFER_DATA (buf)); + + size = mp3_type_frame_length_from_header (GST_ELEMENT_CAST (rtpmpadepay), + frame->header, &version, &layer, &channels, NULL, NULL, NULL, &crc); + if (!size) + goto corrupt_frame; + + frame->size = size; + frame->layer = layer; + if (version == 1 && channels == 2) + frame->side_info = 32; + else if ((version == 1 && channels == 1) || (version >= 2 && channels == 2)) + frame->side_info = 17; + else if (version >= 2 && channels == 1) + frame->side_info = 9; + else { + g_assert_not_reached (); + goto corrupt_frame; + } + + /* backpointer */ + if (layer == 3) { + frame->backpointer = GST_READ_UINT16_BE (GST_BUFFER_DATA (buf) + 4); + frame->backpointer >>= 7; + GST_LOG_OBJECT (rtpmpadepay, "backpointer: %d", frame->backpointer); + } + + if (crc) + frame->side_info += 2; + + GST_LOG_OBJECT (rtpmpadepay, "side info: %d", frame->side_info); + frame->data_size = frame->size - 4 - frame->side_info; + + /* some size validation checks */ + if (4 + frame->side_info > GST_BUFFER_SIZE (buf)) + goto corrupt_frame; + + /* ADU data would then extend past MP3 frame, + * even using past byte reservoir */ + if (-frame->backpointer + GST_BUFFER_SIZE (buf) > frame->size) + goto corrupt_frame; + + /* ok, take buffer and queue */ + frame->buffer = buf; + g_queue_push_tail (rtpmpadepay->adu_frames, frame); + + return TRUE; + + /* ERRORS */ +corrupt_frame: + { + GST_DEBUG_OBJECT (rtpmpadepay, "frame is corrupt"); + gst_buffer_unref (buf); + if (frame) + g_slice_free (GstADUFrame, frame); + return FALSE; + } +} + +static inline void +gst_rtp_mpa_robust_depay_free_frame (GstADUFrame * frame) +{ + if (frame->buffer) + gst_buffer_unref (frame->buffer); + g_slice_free (GstADUFrame, frame); +} + +static inline void +gst_rtp_mpa_robust_depay_dequeue_frame (GstRtpMPARobustDepay * rtpmpadepay) +{ + GstADUFrame *head; + + GST_LOG_OBJECT (rtpmpadepay, "dequeueing ADU frame"); + + if (rtpmpadepay->adu_frames->head == rtpmpadepay->cur_adu_frame) + rtpmpadepay->cur_adu_frame = NULL; + + head = g_queue_pop_head (rtpmpadepay->adu_frames); + g_assert (head->buffer); + gst_rtp_mpa_robust_depay_free_frame (head); + + return; +} + +/* returns TRUE if at least one new ADU frame was enqueued for MP3 conversion. + * Takes ownership of @buf. */ +static gboolean +gst_rtp_mpa_robust_depay_deinterleave (GstRtpMPARobustDepay * rtpmpadepay, + GstBuffer * buf) +{ + gboolean ret = FALSE; + guint8 *data; + guint val, iindex, icc; + + data = GST_BUFFER_DATA (buf); + val = GST_READ_UINT16_BE (data) >> 5; + iindex = val >> 3; + icc = val & 0x7; + + GST_LOG_OBJECT (rtpmpadepay, "sync: 0x%x, index: %u, cycle count: %u", + val, iindex, icc); + + /* basic case; no interleaving ever seen */ + if (val == 0x7ff && rtpmpadepay->last_icc < 0) { + ret = gst_rtp_mpa_robust_depay_queue_frame (rtpmpadepay, buf); + } else { + if (G_UNLIKELY (rtpmpadepay->last_icc < 0)) { + rtpmpadepay->last_icc = icc; + rtpmpadepay->last_ii = iindex; + } + if (icc != rtpmpadepay->last_icc || iindex == rtpmpadepay->last_ii) { + gint i; + + for (i = 0; i < 256; ++i) { + if (rtpmpadepay->deinter[i] != NULL) { + ret |= gst_rtp_mpa_robust_depay_queue_frame (rtpmpadepay, + rtpmpadepay->deinter[i]); + rtpmpadepay->deinter[i] = NULL; + } + } + } + /* rewrite buffer sync header */ + val = GST_READ_UINT16_BE (buf); + val = (0x7ff << 5) | val; + GST_WRITE_UINT16_BE (buf, val); + /* store and keep track of last indices */ + rtpmpadepay->last_icc = icc; + rtpmpadepay->last_ii = iindex; + rtpmpadepay->deinter[iindex] = buf; + } + + return ret; +} + +/* Head ADU frame corresponds to mp3_frame (i.e. in header in side-info) that + * is currently being written + * cur_adu_frame refers to ADU frame whose data should be bytewritten next + * (possibly starting from offset rather than start 0) (and is typicall tail + * at time of last push round). + * If at start, position where it should start writing depends on (data) sizes + * of previous mp3 frames (corresponding to foregoing ADU frames) kept in size, + * and its backpointer */ +static GstFlowReturn +gst_rtp_mpa_robust_depay_push_mp3_frames (GstRtpMPARobustDepay * rtpmpadepay) +{ + GstBuffer *buf; + GstADUFrame *frame, *head; + gint av; + GstFlowReturn ret = GST_FLOW_OK; + + while (1) { + + if (G_UNLIKELY (!rtpmpadepay->cur_adu_frame)) { + rtpmpadepay->cur_adu_frame = rtpmpadepay->adu_frames->head; + rtpmpadepay->offset = 0; + rtpmpadepay->size = 0; + } + + if (G_UNLIKELY (!rtpmpadepay->cur_adu_frame)) + break; + + frame = (GstADUFrame *) rtpmpadepay->cur_adu_frame->data; + head = (GstADUFrame *) rtpmpadepay->adu_frames->head->data; + + /* special case: non-layer III are sent straight through */ + if (G_UNLIKELY (frame->layer != 3)) { + GST_DEBUG_OBJECT (rtpmpadepay, "layer %d frame, sending as-is", + frame->layer); + gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpmpadepay), + frame->buffer); + frame->buffer = NULL; + /* and remove it from any further consideration */ + g_slice_free (GstADUFrame, frame); + g_queue_delete_link (rtpmpadepay->adu_frames, rtpmpadepay->cur_adu_frame); + rtpmpadepay->cur_adu_frame = NULL; + continue; + } + + if (rtpmpadepay->offset == GST_BUFFER_SIZE (frame->buffer)) { + if (g_list_next (rtpmpadepay->cur_adu_frame)) { + GST_LOG_OBJECT (rtpmpadepay, + "moving to next ADU frame, size %d, side_info %d", + frame->size, frame->side_info); + rtpmpadepay->size += frame->data_size; + rtpmpadepay->cur_adu_frame = g_list_next (rtpmpadepay->cur_adu_frame); + frame = (GstADUFrame *) rtpmpadepay->cur_adu_frame->data; + rtpmpadepay->offset = 0; + /* layer I and II packets have no bitreservoir and must be sent as-is; + * so flush any pending frame */ + if (G_UNLIKELY (frame->layer != 3 && rtpmpadepay->mp3_frame)) + goto flush; + } else { + break; + } + } + + if (G_UNLIKELY (!rtpmpadepay->mp3_frame)) { + GST_LOG_OBJECT (rtpmpadepay, + "setting up new MP3 frame of size %d, side_info %d", + head->size, head->side_info); + rtpmpadepay->mp3_frame = gst_byte_writer_new_with_size (head->size, TRUE); + /* 0-fill possible gaps */ + gst_byte_writer_fill (rtpmpadepay->mp3_frame, 0, head->size); + gst_byte_writer_set_pos (rtpmpadepay->mp3_frame, 0); + /* bytewriter corresponds to head frame, + * i.e. the header and the side info must match */ + gst_byte_writer_put_data (rtpmpadepay->mp3_frame, + GST_BUFFER_DATA (head->buffer), 4 + head->side_info); + } + + buf = frame->buffer; + av = gst_byte_writer_get_remaining (rtpmpadepay->mp3_frame); + GST_LOG_OBJECT (rtpmpadepay, "current mp3 frame remaining: %d", av); + GST_LOG_OBJECT (rtpmpadepay, "accumulated ADU frame data_size: %d", + rtpmpadepay->size); + + if (rtpmpadepay->offset) { + /* no need to position, simply append */ + g_assert (GST_BUFFER_SIZE (buf) > rtpmpadepay->offset); + av = MIN (av, GST_BUFFER_SIZE (buf) - rtpmpadepay->offset); + GST_LOG_OBJECT (rtpmpadepay, + "appending %d bytes from ADU frame at offset %d", av, + rtpmpadepay->offset); + gst_byte_writer_put_data (rtpmpadepay->mp3_frame, + GST_BUFFER_DATA (buf) + rtpmpadepay->offset, av); + rtpmpadepay->offset += av; + } else { + gint pos, tpos; + + /* position writing according to ADU frame backpointer */ + pos = gst_byte_writer_get_pos (rtpmpadepay->mp3_frame); + tpos = rtpmpadepay->size - frame->backpointer + 4 + head->side_info; + GST_LOG_OBJECT (rtpmpadepay, "current MP3 frame at position %d, " + "starting new ADU frame data at offset %d", pos, tpos); + if (tpos < pos) { + GstADUFrame *dummy; + + /* try to insert as few frames as possible, + * so go for a reasonably large dummy frame size */ + GST_LOG_OBJECT (rtpmpadepay, + "overlapping previous data; inserting dummy frame"); + dummy = + gst_rtp_mpa_robust_depay_generate_dummy_frame (rtpmpadepay, frame); + g_queue_insert_before (rtpmpadepay->adu_frames, + rtpmpadepay->cur_adu_frame, dummy); + /* offset is known to be zero, so we can shift current one */ + rtpmpadepay->cur_adu_frame = rtpmpadepay->cur_adu_frame->prev; + /* ... and continue adding that empty one immediately, + * and then see if that provided enough extra space */ + continue; + } else if (tpos >= pos + av) { + /* ADU frame no longer needs current MP3 frame; move to its end */ + GST_LOG_OBJECT (rtpmpadepay, "passed current MP3 frame"); + gst_byte_writer_set_pos (rtpmpadepay->mp3_frame, pos + av); + } else { + /* position and append */ + GST_LOG_OBJECT (rtpmpadepay, "adding to current MP3 frame"); + gst_byte_writer_set_pos (rtpmpadepay->mp3_frame, tpos); + av = MIN (av, GST_BUFFER_SIZE (buf) - 4 - frame->side_info); + gst_byte_writer_put_data (rtpmpadepay->mp3_frame, + GST_BUFFER_DATA (buf) + 4 + frame->side_info, av); + rtpmpadepay->offset += av + 4 + frame->side_info; + } + } + + /* if mp3 frame filled, send on its way */ + if (gst_byte_writer_get_remaining (rtpmpadepay->mp3_frame) == 0) { + flush: + buf = gst_byte_writer_free_and_get_buffer (rtpmpadepay->mp3_frame); + rtpmpadepay->mp3_frame = NULL; + GST_BUFFER_TIMESTAMP (buf) = GST_BUFFER_TIMESTAMP (head->buffer); + /* no longer need head ADU frame header and side info */ + /* NOTE maybe head == current, then size and offset go off a bit, + * but current gets reset to NULL, and then also offset and size */ + rtpmpadepay->size -= head->data_size; + gst_rtp_mpa_robust_depay_dequeue_frame (rtpmpadepay); + /* send */ + ret = gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpmpadepay), + buf); + } + } + + return ret; +} + +/* process ADU frame @buf through: + * - deinterleaving + * - converting to MP3 frames + * Takes ownership of @buf. + */ +static GstFlowReturn +gst_rtp_mpa_robust_depay_submit_adu (GstRtpMPARobustDepay * rtpmpadepay, + GstBuffer * buf) +{ + if (gst_rtp_mpa_robust_depay_deinterleave (rtpmpadepay, buf)) + return gst_rtp_mpa_robust_depay_push_mp3_frames (rtpmpadepay); + + return GST_FLOW_OK; +} + +static GstBuffer * +gst_rtp_mpa_robust_depay_process (GstBaseRTPDepayload * depayload, + GstBuffer * buf) +{ + GstRtpMPARobustDepay *rtpmpadepay; + gint payload_len, offset; + guint8 *payload; + gboolean cont, dtype; + guint av, size; + GstClockTime timestamp; + + rtpmpadepay = GST_RTP_MPA_ROBUST_DEPAY (depayload); + + payload_len = gst_rtp_buffer_get_payload_len (buf); + timestamp = GST_BUFFER_TIMESTAMP (buf); + + if (payload_len <= 1) + goto short_read; + + payload = gst_rtp_buffer_get_payload (buf); + offset = 0; + GST_LOG_OBJECT (rtpmpadepay, "payload_len: %d", payload_len); + + /* strip off descriptor + * + * 0 1 + * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * |C|T| ADU size | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * + * C: if 1, data is continuation + * T: if 1, size is 14 bits, otherwise 6 bits + * ADU size: size of following packet (not including descriptor) + */ + while (payload_len) { + if (G_LIKELY (rtpmpadepay->has_descriptor)) { + cont = !!(payload[offset] & 0x80); + dtype = !!(payload[offset] & 0x40); + if (dtype) { + size = (payload[offset] & 0x3f) << 8 | payload[offset + 1]; + payload_len--; + offset++; + } else if (payload_len >= 2) { + size = (payload[offset] & 0x3f); + payload_len -= 2; + offset += 2; + } else { + goto short_read; + } + } else { + cont = FALSE; + dtype = -1; + size = payload_len; + } + + GST_LOG_OBJECT (rtpmpadepay, "offset %d has cont: %d, dtype: %d, size: %d", + offset, cont, dtype, size); + + buf = gst_rtp_buffer_get_payload_subbuffer (buf, offset, + MIN (size, payload_len)); + + if (cont) { + av = gst_adapter_available (rtpmpadepay->adapter); + if (G_UNLIKELY (!av)) { + GST_DEBUG_OBJECT (rtpmpadepay, + "discarding continuation fragment without prior fragment"); + gst_buffer_unref (buf); + } else { + av += GST_BUFFER_SIZE (buf); + gst_adapter_push (rtpmpadepay->adapter, buf); + if (av == size) { + timestamp = gst_adapter_prev_timestamp (rtpmpadepay->adapter, NULL); + buf = gst_adapter_take_buffer (rtpmpadepay->adapter, size); + GST_BUFFER_TIMESTAMP (buf) = timestamp; + gst_rtp_mpa_robust_depay_submit_adu (rtpmpadepay, buf); + } else if (av > size) { + GST_DEBUG_OBJECT (rtpmpadepay, + "assembled ADU size %d larger than expected %d; discarding", + av, size); + gst_adapter_clear (rtpmpadepay->adapter); + } + } + size = payload_len; + } else { + /* not continuation, first fragment or whole ADU */ + if (payload_len == size) { + /* whole ADU */ + GST_BUFFER_TIMESTAMP (buf) = timestamp; + gst_rtp_mpa_robust_depay_submit_adu (rtpmpadepay, buf); + } else if (payload_len < size) { + /* first fragment */ + gst_adapter_push (rtpmpadepay->adapter, buf); + size = payload_len; + } + } + + offset += size; + payload_len -= size; + + /* timestamp applies to first payload, no idea for subsequent ones */ + timestamp = GST_CLOCK_TIME_NONE; + } + + return NULL; + + /* ERRORS */ +short_read: + { + GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE, + (NULL), ("Packet contains invalid data")); + return NULL; + } +} + +static GstStateChangeReturn +gst_rtp_mpa_robust_change_state (GstElement * element, + GstStateChange transition) +{ + GstStateChangeReturn ret; + GstRtpMPARobustDepay *rtpmpadepay; + + rtpmpadepay = GST_RTP_MPA_ROBUST_DEPAY (element); + + switch (transition) { + case GST_STATE_CHANGE_READY_TO_PAUSED: + rtpmpadepay->last_ii = -1; + rtpmpadepay->last_icc = -1; + rtpmpadepay->size = 0; + rtpmpadepay->offset = 0; + default: + break; + } + + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + if (ret != GST_STATE_CHANGE_SUCCESS) + return ret; + + switch (transition) { + case GST_STATE_CHANGE_PAUSED_TO_READY: + { + gint i; + + gst_adapter_clear (rtpmpadepay->adapter); + for (i = 0; i < G_N_ELEMENTS (rtpmpadepay->deinter); i++) { + gst_buffer_unref (rtpmpadepay->deinter[i]); + rtpmpadepay->deinter[i] = NULL; + } + rtpmpadepay->cur_adu_frame = NULL; + g_queue_foreach (rtpmpadepay->adu_frames, + (GFunc) gst_rtp_mpa_robust_depay_free_frame, NULL); + g_queue_clear (rtpmpadepay->adu_frames); + break; + } + default: + break; + } + + return ret; +} + +gboolean +gst_rtp_mpa_robust_depay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpmparobustdepay", + GST_RANK_MARGINAL, GST_TYPE_RTP_MPA_ROBUST_DEPAY); +} diff --git a/gst/rtp/gstrtpmparobustdepay.h b/gst/rtp/gstrtpmparobustdepay.h new file mode 100644 index 0000000000..8ef5af10ba --- /dev/null +++ b/gst/rtp/gstrtpmparobustdepay.h @@ -0,0 +1,78 @@ +/* GStreamer + * Copyright (C) <2010> Mark Nauwelaerts + * Copyright (C) <2010> Nokia Corporation + * + * 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. + */ + +#ifndef __GST_RTP_MPA_ROBUST_DEPAY_H__ +#define __GST_RTP_MPA_ROBUST_DEPAY_H__ + +#include +#include +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_RTP_MPA_ROBUST_DEPAY \ + (gst_rtp_mpa_robust_depay_get_type()) +#define GST_RTP_MPA_ROBUST_DEPAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_MPA_ROBUST_DEPAY,GstRtpMPARobustDepay)) +#define GST_RTP_MPA_ROBUST_DEPAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_MPA_ROBUST_DEPAY,GstRtpMPARobustDepayClass)) +#define GST_IS_RTP_MPA_ROBUST_DEPAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_MPA_ROBUST_DEPAY)) +#define GST_IS_RTP_MPA_ROBUST_DEPAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_MPA_ROBUST_DEPAY)) + +typedef struct _GstRtpMPARobustDepay GstRtpMPARobustDepay; +typedef struct _GstRtpMPARobustDepayClass GstRtpMPARobustDepayClass; + +struct _GstRtpMPARobustDepay +{ + GstBaseRTPDepayload depayload; + + GstAdapter *adapter; + gboolean has_descriptor; + + /* last interleave index */ + gint last_ii; + /* last interleave cycle count */ + gint last_icc; + /* buffers pending deinterleaving */ + GstBuffer *deinter[256]; + + /* ADU buffers pending MP3 transformation */ + GQueue *adu_frames; + GList *cur_adu_frame; + gint offset; + gint size; + GstByteWriter *mp3_frame; +}; + +struct _GstRtpMPARobustDepayClass +{ + GstBaseRTPDepayloadClass parent_class; +}; + +GType gst_rtp_mpa_robust_depay_get_type (void); + +gboolean gst_rtp_mpa_robust_depay_plugin_init (GstPlugin * plugin); + +G_END_DECLS + +#endif /* __GST_RTP_MPA_ROBUST_DEPAY_H__ */