diff options
Diffstat (limited to 'wearable/gst/rtp/gstrtpmparobustdepay.c')
| -rw-r--r-- | wearable/gst/rtp/gstrtpmparobustdepay.c | 792 |
1 files changed, 0 insertions, 792 deletions
diff --git a/wearable/gst/rtp/gstrtpmparobustdepay.c b/wearable/gst/rtp/gstrtpmparobustdepay.c deleted file mode 100644 index 17645a4..0000000 --- a/wearable/gst/rtp/gstrtpmparobustdepay.c +++ /dev/null @@ -1,792 +0,0 @@ -/* GStreamer - * Copyright (C) <2010> Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk> - * 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 <gst/rtp/gstrtpbuffer.h> - -#include <stdio.h> -#include <string.h> -#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_static_pad_template (element_class, - &gst_rtp_mpa_robust_depay_src_template); - gst_element_class_add_static_pad_template (element_class, - &gst_rtp_mpa_robust_depay_sink_template); - - gst_element_class_set_details_simple (element_class, - "RTP MPEG audio depayloader", "Codec/Depayloader/Network/RTP", - "Extracts MPEG audio from RTP packets (RFC 5219)", - "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>"); -} - -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) { - GST_DEBUG_OBJECT (mp3parse, "invalid bitrate"); - 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; - - GST_LOG_OBJECT (mp3parse, "size = %u", length); - 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)) | (0xe << 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->side_info + 4); - memset (GST_BUFFER_DATA (dummy->buffer), 0, dummy->side_info + 4); - 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 + (gint) (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; - if (!rtpmpadepay->size) { - g_assert (rtpmpadepay->cur_adu_frame == - rtpmpadepay->adu_frames->head); - GST_LOG_OBJECT (rtpmpadepay, "... which is new head frame"); - gst_byte_writer_free (rtpmpadepay->mp3_frame); - rtpmpadepay->mp3_frame = NULL; - } - /* ... 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_replace (&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_SECONDARY, GST_TYPE_RTP_MPA_ROBUST_DEPAY); -} |