gstreamer/gst/rtp/gstrtph264pay.c

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/* ex: set tabstop=2 shiftwidth=2 expandtab: */
/* GStreamer
*
* 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 <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtph264pay.h"
#define IDR_TYPE_ID 5
#define SPS_TYPE_ID 7
#define PPS_TYPE_ID 8
#define USE_MEMCMP
GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug);
#define GST_CAT_DEFAULT (rtph264pay_debug)
/* references:
*
* RFC 3984
*/
static GstStaticPadTemplate gst_rtp_h264_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-h264")
);
static GstStaticPadTemplate gst_rtp_h264_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"video\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 90000, " "encoding-name = (string) \"H264\"")
);
#define GST_TYPE_H264_SCAN_MODE (gst_h264_scan_mode_get_type())
static GType
gst_h264_scan_mode_get_type (void)
{
static GType h264_scan_mode_type = 0;
static const GEnumValue h264_scan_modes[] = {
{GST_H264_SCAN_MODE_BYTESTREAM,
"Scan complete bytestream for NALUs (not implemented)",
"bytestream"},
{GST_H264_SCAN_MODE_MULTI_NAL, "Buffers contain multiple complete NALUs",
"multiple"},
{GST_H264_SCAN_MODE_SINGLE_NAL, "Buffers contain a single complete NALU",
"single"},
{0, NULL, NULL},
};
if (!h264_scan_mode_type) {
h264_scan_mode_type =
g_enum_register_static ("GstH264PayScanMode", h264_scan_modes);
}
return h264_scan_mode_type;
}
#define DEFAULT_PROFILE_LEVEL_ID NULL
#define DEFAULT_SPROP_PARAMETER_SETS NULL
#define DEFAULT_SCAN_MODE GST_H264_SCAN_MODE_MULTI_NAL
#define DEFAULT_BUFFER_LIST FALSE
#define DEFAULT_CONFIG_INTERVAL 0
enum
{
PROP_0,
PROP_PROFILE_LEVEL_ID,
PROP_SPROP_PARAMETER_SETS,
PROP_SCAN_MODE,
PROP_BUFFER_LIST,
PROP_CONFIG_INTERVAL,
PROP_LAST
};
#define IS_ACCESS_UNIT(x) (((x) > 0x00) && ((x) < 0x06))
static void gst_rtp_h264_pay_finalize (GObject * object);
static void gst_rtp_h264_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_h264_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * pad,
GstBuffer * buffer);
GST_BOILERPLATE (GstRtpH264Pay, gst_rtp_h264_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_h264_pay_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_h264_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_h264_pay_sink_template));
gst_element_class_set_details_simple (element_class, "RTP H264 payloader",
"Codec/Payloader/Network",
"Payload-encode H264 video into RTP packets (RFC 3984)",
"Laurent Glayal <spglegle@yahoo.fr>");
}
static void
gst_rtp_h264_pay_class_init (GstRtpH264PayClass * klass)
{
GObjectClass *gobject_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
gobject_class->set_property = gst_rtp_h264_pay_set_property;
gobject_class->get_property = gst_rtp_h264_pay_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_PROFILE_LEVEL_ID, g_param_spec_string ("profile-level-id",
"profile-level-id",
"The base64 profile-level-id to set in the sink caps (deprecated)",
DEFAULT_PROFILE_LEVEL_ID,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_SPROP_PARAMETER_SETS, g_param_spec_string ("sprop-parameter-sets",
"sprop-parameter-sets",
"The base64 sprop-parameter-sets to set in out caps (set to NULL to "
"extract from stream)",
DEFAULT_SPROP_PARAMETER_SETS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SCAN_MODE,
g_param_spec_enum ("scan-mode", "Scan Mode",
"How to scan the input buffers for NAL units. Performance can be "
"increased when certain assumptions are made about the input buffers",
GST_TYPE_H264_SCAN_MODE, DEFAULT_SCAN_MODE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_LIST,
g_param_spec_boolean ("buffer-list", "Buffer List",
"Use Buffer Lists",
DEFAULT_BUFFER_LIST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_CONFIG_INTERVAL,
g_param_spec_uint ("config-interval",
"SPS PPS Send Interval",
"Send SPS and PPS Insertion Interval in seconds (sprop parameter sets "
"will be multiplexed in the data stream when detected.) (0 = disabled)",
0, 3600, DEFAULT_CONFIG_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
gobject_class->finalize = gst_rtp_h264_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_h264_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_h264_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtph264pay_debug, "rtph264pay", 0,
"H264 RTP Payloader");
}
static void
gst_rtp_h264_pay_init (GstRtpH264Pay * rtph264pay, GstRtpH264PayClass * klass)
{
2009-08-03 16:59:32 +00:00
rtph264pay->queue = g_array_new (FALSE, FALSE, sizeof (guint));
rtph264pay->profile = 0;
rtph264pay->sps = NULL;
rtph264pay->pps = NULL;
rtph264pay->last_spspps = -1;
rtph264pay->scan_mode = GST_H264_SCAN_MODE_MULTI_NAL;
rtph264pay->buffer_list = DEFAULT_BUFFER_LIST;
rtph264pay->spspps_interval = DEFAULT_CONFIG_INTERVAL;
}
static void
gst_rtp_h264_pay_clear_sps_pps (GstRtpH264Pay * rtph264pay)
{
g_list_foreach (rtph264pay->sps, (GFunc) gst_mini_object_unref, NULL);
g_list_free (rtph264pay->sps);
rtph264pay->sps = NULL;
g_list_foreach (rtph264pay->pps, (GFunc) gst_mini_object_unref, NULL);
g_list_free (rtph264pay->pps);
rtph264pay->pps = NULL;
}
static void
gst_rtp_h264_pay_finalize (GObject * object)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
2009-08-03 16:59:32 +00:00
g_array_free (rtph264pay->queue, TRUE);
gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
g_free (rtph264pay->sprop_parameter_sets);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/* take the currently configured SPS and PPS lists and set them on the caps as
* sprop-parameter-sets */
static gboolean
gst_rtp_h264_pay_set_sps_pps (GstBaseRTPPayload * basepayload)
{
GstRtpH264Pay *payloader = GST_RTP_H264_PAY (basepayload);
gchar *profile;
gchar *set;
GList *walk;
GString *sprops;
guint count;
gboolean res;
sprops = g_string_new ("");
count = 0;
/* build the sprop-parameter-sets */
for (walk = payloader->sps; walk; walk = g_list_next (walk)) {
GstBuffer *sps_buf = GST_BUFFER_CAST (walk->data);
set =
g_base64_encode (GST_BUFFER_DATA (sps_buf), GST_BUFFER_SIZE (sps_buf));
g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
g_free (set);
count++;
}
for (walk = payloader->pps; walk; walk = g_list_next (walk)) {
GstBuffer *pps_buf = GST_BUFFER_CAST (walk->data);
set =
g_base64_encode (GST_BUFFER_DATA (pps_buf), GST_BUFFER_SIZE (pps_buf));
g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
g_free (set);
count++;
}
/* profile is 24 bit. Force it to respect the limit */
profile = g_strdup_printf ("%06x", payloader->profile & 0xffffff);
/* combine into output caps */
res = gst_basertppayload_set_outcaps (basepayload,
"sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
g_string_free (sprops, TRUE);
g_free (profile);
return res;
}
static gboolean
gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpH264Pay *rtph264pay;
GstStructure *str;
const GValue *value;
guint8 *data;
guint size;
rtph264pay = GST_RTP_H264_PAY (basepayload);
str = gst_caps_get_structure (caps, 0);
/* we can only set the output caps when we found the sprops and profile
* NALs */
gst_basertppayload_set_options (basepayload, "video", TRUE, "H264", 90000);
/* packetized AVC video has a codec_data */
if ((value = gst_structure_get_value (str, "codec_data"))) {
GstBuffer *buffer;
guint num_sps, num_pps;
gint i, nal_size;
GST_DEBUG_OBJECT (rtph264pay, "have packetized h264");
rtph264pay->packetized = TRUE;
buffer = gst_value_get_buffer (value);
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
/* parse the avcC data */
if (size < 7)
goto avcc_too_small;
/* parse the version, this must be 1 */
if (data[0] != 1)
goto wrong_version;
/* AVCProfileIndication */
/* profile_compat */
/* AVCLevelIndication */
rtph264pay->profile = (data[1] << 16) | (data[2] << 8) | data[3];
GST_DEBUG_OBJECT (rtph264pay, "profile %06x", rtph264pay->profile);
/* 6 bits reserved | 2 bits lengthSizeMinusOne */
/* this is the number of bytes in front of the NAL units to mark their
* length */
rtph264pay->nal_length_size = (data[4] & 0x03) + 1;
GST_DEBUG_OBJECT (rtph264pay, "nal length %u", rtph264pay->nal_length_size);
/* 3 bits reserved | 5 bits numOfSequenceParameterSets */
num_sps = data[5] & 0x1f;
GST_DEBUG_OBJECT (rtph264pay, "num SPS %u", num_sps);
data += 6;
size -= 6;
/* create the sprop-parameter-sets */
for (i = 0; i < num_sps; i++) {
GstBuffer *sps_buf;
if (size < 2)
goto avcc_error;
nal_size = (data[0] << 8) | data[1];
data += 2;
size -= 2;
gst/rtp/gstrtpL16depay.c: Check if clock-rate and channels are valid. Original commit message from CVS: * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps), (gst_rtp_L16_depay_process): Check if clock-rate and channels are valid. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_setcaps), (gst_rtp_ac3_depay_process): Don't ignore the return value of set_caps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process): * gst/rtp/gstrtpamrdepay.h: Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set output caps on the buffers, the base class does that for us. The subclass will make sure we are negotiated. * gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_setcaps), (gst_rtp_dv_depay_process), (gst_rtp_dv_depay_reset): * gst/rtp/gstrtpdvdepay.h: Clean up caps negotiation. The subclass will make sure we are negotiated. * gst/rtp/gstrtpg726depay.c: (gst_rtp_g726_depay_setcaps), (gst_rtp_g726_depay_process): Clean up caps negotiation. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_init), (gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process): * gst/rtp/gstrtpg729depay.h: The subclass will make sure we are negotiated. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_setcaps), (gst_rtp_gsm_depay_process): Clean up caps negotiation. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_setcaps): Clean up caps negotiation. Don't ignore the return value of set_outcaps. * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_setcaps), (gst_rtp_h263_depay_process): Clean up caps negotiation. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_setcaps), (gst_rtp_h263_pay_flush), (gst_rtp_h263_pay_handle_buffer): * gst/rtp/gstrtph263pay.h: Don't ignore the return value of set_outcaps. Do some more timestamps. * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps), (gst_rtp_h263p_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_setcaps), (gst_rtp_h263p_pay_flush), (gst_rtp_h263p_pay_handle_buffer): * gst/rtp/gstrtph263ppay.h: Don't ignore the return value of set_outcaps. Do some more timestamps. * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. Fix possible caps leak. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_setcaps): Add some more debug info. * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps), (gst_rtp_ilbc_depay_process): Clean up caps negotiation. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_sink_setcaps): Clean up caps negotiation. * gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_setcaps), (gst_rtp_mp1s_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set caps on buffers, subclass does that for us. * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps), (gst_rtp_mp2t_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set caps on buffers, subclass does that for us. * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_new_caps), (gst_rtp_mp4a_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set caps on buffers, subclass does that for us. * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set caps on buffers, subclass does that for us. * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_new_caps), (gst_rtp_mp4v_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_setcaps), (gst_rtp_mpv_depay_process): Clean up caps negotiation. Actually set output caps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtpmpvpay.c: (gst_rtp_mpv_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps), (gst_rtp_pcma_depay_process): Clean up caps negotiation. Set output buffer duration because we can. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps), (gst_rtp_pcmu_depay_process): Clean up caps negotiation. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init), (gst_rtp_speex_depay_setcaps), (gst_rtp_speex_depay_process): Clean up caps negotiation. Set output caps on the pad and header buffers. Set duration on output buffers because we can. * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_parse_ident): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_setcaps), (gst_rtp_sv3v_depay_process): Clean up caps negotiation. No need to validate the buffer, the base class does that for us. No need to set caps out output buffers, subclass does that. * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps), (gst_rtp_theora_depay_process): Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_class_init), (gst_rtp_theora_pay_flush_packet), (encode_base64), (gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id), (gst_rtp_theora_pay_handle_buffer): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps), (gst_rtp_vorbis_depay_process): Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_setcaps): Clean up caps negotiation, don't ignore setcaps return. * gst/rtp/gstrtpvrawpay.c: (gst_rtp_vraw_pay_setcaps): Don't ignore the return value of set_outcaps.
2008-10-27 11:03:53 +00:00
GST_LOG_OBJECT (rtph264pay, "SPS %d size %d", i, nal_size);
if (size < nal_size)
goto avcc_error;
/* make a buffer out of it and add to SPS list */
sps_buf = gst_buffer_new_and_alloc (nal_size);
memcpy (GST_BUFFER_DATA (sps_buf), data, nal_size);
rtph264pay->sps = g_list_append (rtph264pay->sps, sps_buf);
data += nal_size;
size -= nal_size;
}
if (size < 1)
goto avcc_error;
/* 8 bits numOfPictureParameterSets */
num_pps = data[0];
data += 1;
size -= 1;
GST_DEBUG_OBJECT (rtph264pay, "num PPS %u", num_pps);
for (i = 0; i < num_pps; i++) {
GstBuffer *pps_buf;
if (size < 2)
goto avcc_error;
nal_size = (data[0] << 8) | data[1];
data += 2;
size -= 2;
gst/rtp/gstrtpL16depay.c: Check if clock-rate and channels are valid. Original commit message from CVS: * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps), (gst_rtp_L16_depay_process): Check if clock-rate and channels are valid. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_setcaps), (gst_rtp_ac3_depay_process): Don't ignore the return value of set_caps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process): * gst/rtp/gstrtpamrdepay.h: Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set output caps on the buffers, the base class does that for us. The subclass will make sure we are negotiated. * gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_setcaps), (gst_rtp_dv_depay_process), (gst_rtp_dv_depay_reset): * gst/rtp/gstrtpdvdepay.h: Clean up caps negotiation. The subclass will make sure we are negotiated. * gst/rtp/gstrtpg726depay.c: (gst_rtp_g726_depay_setcaps), (gst_rtp_g726_depay_process): Clean up caps negotiation. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_init), (gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process): * gst/rtp/gstrtpg729depay.h: The subclass will make sure we are negotiated. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_setcaps), (gst_rtp_gsm_depay_process): Clean up caps negotiation. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_setcaps): Clean up caps negotiation. Don't ignore the return value of set_outcaps. * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_setcaps), (gst_rtp_h263_depay_process): Clean up caps negotiation. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_setcaps), (gst_rtp_h263_pay_flush), (gst_rtp_h263_pay_handle_buffer): * gst/rtp/gstrtph263pay.h: Don't ignore the return value of set_outcaps. Do some more timestamps. * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps), (gst_rtp_h263p_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_setcaps), (gst_rtp_h263p_pay_flush), (gst_rtp_h263p_pay_handle_buffer): * gst/rtp/gstrtph263ppay.h: Don't ignore the return value of set_outcaps. Do some more timestamps. * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. Fix possible caps leak. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_setcaps): Add some more debug info. * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps), (gst_rtp_ilbc_depay_process): Clean up caps negotiation. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_sink_setcaps): Clean up caps negotiation. * gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_setcaps), (gst_rtp_mp1s_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set caps on buffers, subclass does that for us. * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps), (gst_rtp_mp2t_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set caps on buffers, subclass does that for us. * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_new_caps), (gst_rtp_mp4a_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set caps on buffers, subclass does that for us. * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set caps on buffers, subclass does that for us. * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_new_caps), (gst_rtp_mp4v_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_setcaps), (gst_rtp_mpv_depay_process): Clean up caps negotiation. Actually set output caps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtpmpvpay.c: (gst_rtp_mpv_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps), (gst_rtp_pcma_depay_process): Clean up caps negotiation. Set output buffer duration because we can. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps), (gst_rtp_pcmu_depay_process): Clean up caps negotiation. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init), (gst_rtp_speex_depay_setcaps), (gst_rtp_speex_depay_process): Clean up caps negotiation. Set output caps on the pad and header buffers. Set duration on output buffers because we can. * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_parse_ident): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_setcaps), (gst_rtp_sv3v_depay_process): Clean up caps negotiation. No need to validate the buffer, the base class does that for us. No need to set caps out output buffers, subclass does that. * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps), (gst_rtp_theora_depay_process): Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_class_init), (gst_rtp_theora_pay_flush_packet), (encode_base64), (gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id), (gst_rtp_theora_pay_handle_buffer): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps), (gst_rtp_vorbis_depay_process): Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_setcaps): Clean up caps negotiation, don't ignore setcaps return. * gst/rtp/gstrtpvrawpay.c: (gst_rtp_vraw_pay_setcaps): Don't ignore the return value of set_outcaps.
2008-10-27 11:03:53 +00:00
GST_LOG_OBJECT (rtph264pay, "PPS %d size %d", i, nal_size);
if (size < nal_size)
goto avcc_error;
/* make a buffer out of it and add to PPS list */
pps_buf = gst_buffer_new_and_alloc (nal_size);
memcpy (GST_BUFFER_DATA (pps_buf), data, nal_size);
rtph264pay->pps = g_list_append (rtph264pay->pps, pps_buf);
data += nal_size;
size -= nal_size;
}
/* and update the caps with the collected data */
if (!gst_rtp_h264_pay_set_sps_pps (basepayload))
return FALSE;
} else {
GST_DEBUG_OBJECT (rtph264pay, "have bytestream h264");
rtph264pay->packetized = FALSE;
}
return TRUE;
avcc_too_small:
{
GST_ERROR_OBJECT (rtph264pay, "avcC size %u < 7", size);
return FALSE;
}
wrong_version:
{
GST_ERROR_OBJECT (rtph264pay, "wrong avcC version");
return FALSE;
}
avcc_error:
{
GST_ERROR_OBJECT (rtph264pay, "avcC too small ");
return FALSE;
}
}
static guint
next_start_code (guint8 * data, guint size)
{
/* Boyer-Moore string matching algorithm, in a degenerative
* sense because our search 'alphabet' is binary - 0 & 1 only.
* This allow us to simplify the general BM algorithm to a very
* simple form. */
/* assume 1 is in the 4th byte */
guint offset = 3;
while (offset < size) {
if (1 == data[offset]) {
unsigned int shift = offset;
if (0 == data[--shift]) {
if (0 == data[--shift]) {
if (0 == data[--shift]) {
return shift;
}
}
}
/* The jump is always 4 because of the 1 previously matched.
* All the 0's must be after this '1' matched at offset */
offset += 4;
} else if (0 == data[offset]) {
/* maybe next byte is 1? */
offset++;
} else {
/* can jump 4 bytes forward */
offset += 4;
}
/* at each iteration, we rescan in a backward manner until
* we match 0.0.0.1 in reverse order. Since our search string
* has only 2 'alpabets' (i.e. 0 & 1), we know that any
* mismatch will force us to shift a fixed number of steps */
}
GST_DEBUG ("Cannot find next NAL start code. returning %u", size);
return size;
}
static gboolean
gst_rtp_h264_pay_decode_nal (GstRtpH264Pay * payloader,
guint8 * data, guint size, GstClockTime timestamp)
{
guint8 *sps = NULL, *pps = NULL;
guint sps_len = 0, pps_len = 0;
guint8 header, type;
guint len;
gboolean updated;
/* default is no update */
updated = FALSE;
GST_DEBUG ("NAL payload len=%u", size);
len = size;
header = data[0];
type = header & 0x1f;
/* keep sps & pps separately so that we can update either one
* independently. We also record the timestamp of the last SPS/PPS so
* that we can insert them at regular intervals and when needed. */
if (SPS_TYPE_ID == type) {
/* encode the entire SPS NAL in base64 */
GST_DEBUG ("Found SPS %x %x %x Len=%u", (header >> 7),
(header >> 5) & 3, type, len);
sps = data;
sps_len = len;
/* remember when we last saw SPS */
if (timestamp != -1)
payloader->last_spspps = timestamp;
} else if (PPS_TYPE_ID == type) {
/* encoder the entire PPS NAL in base64 */
GST_DEBUG ("Found PPS %x %x %x Len = %u",
(header >> 7), (header >> 5) & 3, type, len);
pps = data;
pps_len = len;
/* remember when we last saw PPS */
if (timestamp != -1)
payloader->last_spspps = timestamp;
} else {
GST_DEBUG ("NAL: %x %x %x Len = %u", (header >> 7),
(header >> 5) & 3, type, len);
}
/* If we encountered an SPS and/or a PPS, check if it's the
* same as the one we have. If not, update our version and
* set updated to TRUE
*/
if (sps_len > 0) {
GstBuffer *sps_buf;
if (payloader->sps != NULL) {
sps_buf = GST_BUFFER_CAST (payloader->sps->data);
if ((GST_BUFFER_SIZE (sps_buf) != sps_len)
|| memcmp (GST_BUFFER_DATA (sps_buf), sps, sps_len)) {
/* something changed, update */
payloader->profile = (sps[1] << 16) + (sps[2] << 8) + sps[3];
GST_DEBUG ("Profile level IDC = %06x", payloader->profile);
updated = TRUE;
}
} else {
/* no previous SPS, update */
updated = TRUE;
}
if (updated) {
sps_buf = gst_buffer_new_and_alloc (sps_len);
memcpy (GST_BUFFER_DATA (sps_buf), sps, sps_len);
if (payloader->sps) {
/* replace old buffer */
gst_buffer_unref (payloader->sps->data);
payloader->sps->data = sps_buf;
} else {
/* add new buffer */
payloader->sps = g_list_prepend (payloader->sps, sps_buf);
}
}
}
if (pps_len > 0) {
GstBuffer *pps_buf;
if (payloader->pps != NULL) {
pps_buf = GST_BUFFER_CAST (payloader->pps->data);
if ((GST_BUFFER_SIZE (pps_buf) != pps_len)
|| memcmp (GST_BUFFER_DATA (pps_buf), pps, pps_len)) {
/* something changed, update */
updated = TRUE;
}
} else {
/* no previous SPS, update */
updated = TRUE;
}
if (updated) {
pps_buf = gst_buffer_new_and_alloc (pps_len);
memcpy (GST_BUFFER_DATA (pps_buf), pps, pps_len);
if (payloader->pps) {
/* replace old buffer */
gst_buffer_unref (payloader->pps->data);
payloader->pps->data = pps_buf;
} else {
/* add new buffer */
payloader->pps = g_list_prepend (payloader->pps, pps_buf);
}
}
}
return updated;
}
static GstFlowReturn
gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload, guint8 * data,
guint size, GstClockTime timestamp, GstBuffer * buffer_orig);
static GstFlowReturn
gst_rtp_h264_pay_send_sps_pps (GstBaseRTPPayload * basepayload,
GstRtpH264Pay * rtph264pay, GstClockTime timestamp)
{
2009-12-23 18:39:05 +00:00
GstFlowReturn ret = GST_FLOW_OK;
GList *walk;
for (walk = rtph264pay->sps; walk; walk = g_list_next (walk)) {
GstBuffer *sps_buf = GST_BUFFER_CAST (walk->data);
GST_DEBUG_OBJECT (rtph264pay, "inserting SPS in the stream");
/* resend SPS */
ret = gst_rtp_h264_pay_payload_nal (basepayload,
GST_BUFFER_DATA (sps_buf), GST_BUFFER_SIZE (sps_buf), timestamp,
sps_buf);
/* Not critical here; but throw a warning */
if (ret != GST_FLOW_OK)
GST_WARNING ("Problem pushing SPS");
}
for (walk = rtph264pay->pps; walk; walk = g_list_next (walk)) {
GstBuffer *pps_buf = GST_BUFFER_CAST (walk->data);
GST_DEBUG_OBJECT (rtph264pay, "inserting PPS in the stream");
/* resend PPS */
ret = gst_rtp_h264_pay_payload_nal (basepayload,
GST_BUFFER_DATA (pps_buf), GST_BUFFER_SIZE (pps_buf), timestamp,
pps_buf);
/* Not critical here; but throw a warning */
if (ret != GST_FLOW_OK)
GST_WARNING ("Problem pushing PPS");
}
if (timestamp != -1)
rtph264pay->last_spspps = timestamp;
return ret;
}
static GstFlowReturn
gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload, guint8 * data,
guint size, GstClockTime timestamp, GstBuffer * buffer_orig)
{
GstRtpH264Pay *rtph264pay;
GstFlowReturn ret;
guint8 nalType;
guint packet_len, payload_len, mtu;
GstBuffer *outbuf;
guint8 *payload;
GstBufferList *list = NULL;
GstBufferListIterator *it = NULL;
gboolean send_spspps;
rtph264pay = GST_RTP_H264_PAY (basepayload);
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtph264pay);
nalType = data[0] & 0x1f;
GST_DEBUG_OBJECT (rtph264pay, "Processing Buffer with NAL TYPE=%d", nalType);
send_spspps = FALSE;
/* check if we need to emit an SPS/PPS now */
if (nalType == IDR_TYPE_ID && rtph264pay->spspps_interval > 0) {
if (rtph264pay->last_spspps != -1) {
guint64 diff;
GST_LOG_OBJECT (rtph264pay,
"now %" GST_TIME_FORMAT ", last SPS/PPS %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp), GST_TIME_ARGS (rtph264pay->last_spspps));
/* calculate diff between last SPS/PPS in milliseconds */
if (timestamp > rtph264pay->last_spspps)
diff = timestamp - rtph264pay->last_spspps;
else
diff = 0;
GST_DEBUG_OBJECT (rtph264pay,
"interval since last SPS/PPS %" GST_TIME_FORMAT,
GST_TIME_ARGS (diff));
/* bigger than interval, queue SPS/PPS */
if (GST_TIME_AS_SECONDS (diff) >= rtph264pay->spspps_interval) {
GST_DEBUG_OBJECT (rtph264pay, "time to send SPS/PPS");
send_spspps = TRUE;
}
} else {
/* no know previous SPS/PPS time, send now */
GST_DEBUG_OBJECT (rtph264pay, "no previous SPS/PPS time, send now");
send_spspps = TRUE;
}
}
if (send_spspps) {
/* we need to send SPS/PPS now first. FIXME, don't use the timestamp for
* checking when we need to send SPS/PPS but convert to running_time first. */
ret = gst_rtp_h264_pay_send_sps_pps (basepayload, rtph264pay, timestamp);
if (ret != GST_FLOW_OK)
return ret;
}
packet_len = gst_rtp_buffer_calc_packet_len (size, 0, 0);
if (packet_len < mtu) {
GST_DEBUG_OBJECT (basepayload,
"NAL Unit fit in one packet datasize=%d mtu=%d", size, mtu);
/* will fit in one packet */
if (rtph264pay->buffer_list) {
/* use buffer lists
* first create buffer without payload containing only the RTP header
* and then another buffer containing the payload. both buffers will
* be then added to the list */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
} else {
/* use the old-fashioned way with a single buffer and memcpy */
outbuf = gst_rtp_buffer_new_allocate (size, 0, 0);
}
/* only set the marker bit on packets containing access units */
if (IS_ACCESS_UNIT (nalType)) {
gst_rtp_buffer_set_marker (outbuf, 1);
}
/* timestamp the outbuffer */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
if (rtph264pay->buffer_list) {
GstBuffer *paybuf;
/* create another buffer with the payload. */
if (buffer_orig)
paybuf = gst_buffer_create_sub (buffer_orig, data -
GST_BUFFER_DATA (buffer_orig), size);
else {
paybuf = gst_buffer_new_and_alloc (size);
memcpy (GST_BUFFER_DATA (paybuf), data, size);
}
list = gst_buffer_list_new ();
it = gst_buffer_list_iterate (list);
/* add both buffers to the buffer list */
gst_buffer_list_iterator_add_group (it);
gst_buffer_list_iterator_add (it, outbuf);
gst_buffer_list_iterator_add (it, paybuf);
gst_buffer_list_iterator_free (it);
/* push the list to the next element in the pipe */
ret = gst_basertppayload_push_list (basepayload, list);
} else {
payload = gst_rtp_buffer_get_payload (outbuf);
GST_DEBUG_OBJECT (basepayload, "Copying %d bytes to outbuf", size);
memcpy (payload, data, size);
ret = gst_basertppayload_push (basepayload, outbuf);
}
} else {
/* fragmentation Units FU-A */
guint8 nalHeader;
guint limitedSize;
int ii = 0, start = 1, end = 0, pos = 0;
GST_DEBUG_OBJECT (basepayload,
"NAL Unit DOES NOT fit in one packet datasize=%d mtu=%d", size, mtu);
nalHeader = *data;
pos++;
size--;
ret = GST_FLOW_OK;
GST_DEBUG_OBJECT (basepayload, "Using FU-A fragmentation for data size=%d",
size);
/* We keep 2 bytes for FU indicator and FU Header */
payload_len = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
if (rtph264pay->buffer_list) {
list = gst_buffer_list_new ();
it = gst_buffer_list_iterate (list);
}
while (end == 0) {
limitedSize = size < payload_len ? size : payload_len;
GST_DEBUG_OBJECT (basepayload,
"Inside FU-A fragmentation limitedSize=%d iteration=%d", limitedSize,
ii);
if (rtph264pay->buffer_list) {
/* use buffer lists
* first create buffer without payload containing only the RTP header
* and then another buffer containing the payload. both buffers will
* be then added to the list */
outbuf = gst_rtp_buffer_new_allocate (2, 0, 0);
} else {
/* use the old-fashioned way with a single buffer and memcpy
* first create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (limitedSize + 2, 0, 0);
}
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
payload = gst_rtp_buffer_get_payload (outbuf);
if (limitedSize == size) {
GST_DEBUG_OBJECT (basepayload, "end size=%d iteration=%d", size, ii);
end = 1;
}
if (IS_ACCESS_UNIT (nalType)) {
gst_rtp_buffer_set_marker (outbuf, end);
}
/* FU indicator */
payload[0] = (nalHeader & 0x60) | 28;
/* FU Header */
payload[1] = (start << 7) | (end << 6) | (nalHeader & 0x1f);
if (rtph264pay->buffer_list) {
GstBuffer *paybuf;
/* create another buffer to hold the payload */
if (buffer_orig)
paybuf = gst_buffer_create_sub (buffer_orig, data -
GST_BUFFER_DATA (buffer_orig) + pos, limitedSize);
else {
paybuf = gst_buffer_new_and_alloc (limitedSize);
memcpy (GST_BUFFER_DATA (paybuf), data + pos, limitedSize);
}
/* create a new group to hold the header and the payload */
gst_buffer_list_iterator_add_group (it);
/* add both buffers to the buffer list */
gst_buffer_list_iterator_add (it, outbuf);
gst_buffer_list_iterator_add (it, paybuf);
} else {
memcpy (&payload[2], data + pos, limitedSize);
GST_DEBUG_OBJECT (basepayload,
"recorded %d payload bytes into packet iteration=%d",
limitedSize + 2, ii);
ret = gst_basertppayload_push (basepayload, outbuf);
if (ret != GST_FLOW_OK)
break;
}
size -= limitedSize;
pos += limitedSize;
ii++;
start = 0;
}
if (rtph264pay->buffer_list) {
/* free iterator and push the whole buffer list at once */
gst_buffer_list_iterator_free (it);
ret = gst_basertppayload_push_list (basepayload, list);
}
}
return ret;
}
static GstFlowReturn
gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpH264Pay *rtph264pay;
GstFlowReturn ret;
2009-08-03 16:59:32 +00:00
guint size, nal_len, i;
guint8 *data, *nal_data;
GstClockTime timestamp;
2009-08-03 16:59:32 +00:00
GArray *nal_queue;
rtph264pay = GST_RTP_H264_PAY (basepayload);
/* the input buffer contains one or more NAL units */
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
ret = GST_FLOW_OK;
GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
/* now loop over all NAL units and put them in a packet
* FIXME, we should really try to pack multiple NAL units into one RTP packet
* if we can, especially for the config packets that wont't cause decoder
* latency. */
if (rtph264pay->packetized) {
guint nal_length_size;
nal_length_size = rtph264pay->nal_length_size;
while (size > nal_length_size) {
gint i;
nal_len = 0;
for (i = 0; i < nal_length_size; i++) {
nal_len = ((nal_len << 8) + data[i]);
}
/* skip the length bytes, make sure we don't run past the buffer size */
data += nal_length_size;
size -= nal_length_size;
if (size >= nal_len) {
GST_DEBUG_OBJECT (basepayload, "got NAL of size %u", nal_len);
} else {
nal_len = size;
GST_DEBUG_OBJECT (basepayload, "got incomplete NAL of size %u",
nal_len);
}
ret =
gst_rtp_h264_pay_payload_nal (basepayload, data, nal_len, timestamp,
buffer);
if (ret != GST_FLOW_OK)
break;
data += nal_len;
size -= nal_len;
}
} else {
guint next;
gboolean update = FALSE;
/* get offset of first start code */
next = next_start_code (data, size);
/* skip to start code, if no start code is found, next will be size and we
* will not collect data. */
data += next;
size -= next;
nal_data = data;
nal_queue = rtph264pay->queue;
2009-08-03 16:59:32 +00:00
/* array must be empty when we get here */
g_assert (nal_queue->len == 0);
GST_DEBUG_OBJECT (basepayload, "found first start at %u, bytes left %u",
next, size);
/* first pass to locate NALs and parse SPS/PPS */
while (size > 4) {
/* skip start code */
data += 4;
size -= 4;
if (rtph264pay->scan_mode == GST_H264_SCAN_MODE_SINGLE_NAL) {
/* we are told that there is only a single NAL in this packet so that we
* can avoid scanning for the next NAL. */
next = size;
} else {
/* use next_start_code() to scan buffer.
* next_start_code() returns the offset in data,
* starting from zero to the first byte of 0.0.0.1
* If no start code is found, it returns the value of the
* 'size' parameter.
* data is unchanged by the call to next_start_code()
*/
next = next_start_code (data, size);
}
/* nal length is distance to next start code */
nal_len = next;
GST_DEBUG_OBJECT (basepayload, "found next start at %u of size %u", next,
nal_len);
if (rtph264pay->sprop_parameter_sets != NULL) {
/* explicitly set profile and sprop, use those */
if (rtph264pay->update_caps) {
if (!gst_basertppayload_set_outcaps (basepayload,
"sprop-parameter-sets", G_TYPE_STRING,
rtph264pay->sprop_parameter_sets, NULL))
goto caps_rejected;
rtph264pay->update_caps = FALSE;
GST_DEBUG ("outcaps udpate: sprop-parameter-sets=%s",
rtph264pay->sprop_parameter_sets);
}
} else {
/* We know our stream is a valid H264 NAL packet,
* go parse it for SPS/PPS to enrich the caps */
/* order: make sure to check nal */
update =
gst_rtp_h264_pay_decode_nal (rtph264pay, data, nal_len, timestamp)
|| update;
}
/* move to next NAL packet */
data += nal_len;
size -= nal_len;
2009-08-03 16:59:32 +00:00
g_array_append_val (nal_queue, nal_len);
}
/* if has new SPS & PPS, update the output caps */
if (G_UNLIKELY (update))
if (!gst_rtp_h264_pay_set_sps_pps (basepayload))
goto caps_rejected;
/* second pass to payload and push */
data = nal_data;
2009-08-03 16:59:32 +00:00
for (i = 0; i < nal_queue->len; i++) {
nal_len = g_array_index (nal_queue, guint, i);
/* skip start code */
data += 4;
/* put the data in one or more RTP packets */
ret =
gst_rtp_h264_pay_payload_nal (basepayload, data, nal_len, timestamp,
buffer);
if (ret != GST_FLOW_OK) {
break;
}
/* move to next NAL packet */
data += nal_len;
size -= nal_len;
}
2009-08-03 16:59:32 +00:00
g_array_set_size (nal_queue, 0);
}
gst_buffer_unref (buffer);
return ret;
caps_rejected:
GST_WARNING_OBJECT (basepayload, "Could not set outcaps");
g_array_set_size (nal_queue, 0);
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
static void
gst_rtp_h264_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
switch (prop_id) {
case PROP_PROFILE_LEVEL_ID:
break;
case PROP_SPROP_PARAMETER_SETS:
g_free (rtph264pay->sprop_parameter_sets);
rtph264pay->sprop_parameter_sets = g_value_dup_string (value);
rtph264pay->update_caps = TRUE;
break;
case PROP_SCAN_MODE:
rtph264pay->scan_mode = g_value_get_enum (value);
break;
case PROP_BUFFER_LIST:
rtph264pay->buffer_list = g_value_get_boolean (value);
break;
case PROP_CONFIG_INTERVAL:
rtph264pay->spspps_interval = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_h264_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
switch (prop_id) {
case PROP_PROFILE_LEVEL_ID:
break;
case PROP_SPROP_PARAMETER_SETS:
g_value_set_string (value, rtph264pay->sprop_parameter_sets);
break;
case PROP_SCAN_MODE:
g_value_set_enum (value, rtph264pay->scan_mode);
break;
case PROP_BUFFER_LIST:
g_value_set_boolean (value, rtph264pay->buffer_list);
break;
case PROP_CONFIG_INTERVAL:
g_value_set_uint (value, rtph264pay->spspps_interval);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_rtp_h264_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtph264pay",
GST_RANK_NONE, GST_TYPE_RTP_H264_PAY);
}