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rtp: add mpa-robust depayloader

Browse source 
Fixes #589997.
This commit is contained in:
Mark Nauwelaerts 2010-06-10 16:14:43 +02:00
parent de5cb168ee
commit 815e06ba55
4 changed files with 866 additions and 0 deletions
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2
gst/rtp/Makefile.am
View file

@ -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 \

4
gst/rtp/gstrtp.c
View file

@ -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;

782
gst/rtp/gstrtpmparobustdepay.c Normal file
View file

@ -0,0 +1,782 @@
/* 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 <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_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 <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)
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);
}

78
gst/rtp/gstrtpmparobustdepay.h Normal file
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/* 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.
*/
#ifndef __GST_RTP_MPA_ROBUST_DEPAY_H__
#define __GST_RTP_MPA_ROBUST_DEPAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertpdepayload.h>
#include <gst/base/gstadapter.h>
#include <gst/base/gstbytewriter.h>
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__ */