2008-10-27 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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 Wim Taymans <wim.taymans@collabora.co.uk>
+
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_handle_src_event):
Forward unknown events upstream.
gint clock_rate, payload;
gint channels;
GstCaps *srccaps;
+ gboolean res;
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
/* caps can overwrite defaults */
clock_rate =
gst_rtp_L16_depay_parse_int (structure, "clock-rate", clock_rate);
+ if (clock_rate == 0)
+ goto no_clockrate;
+
channels = gst_rtp_L16_depay_parse_int (structure, "channels", channels);
+ if (channels == 0)
+ goto no_channels;
depayload->clock_rate = clock_rate;
rtpL16depay->rate = clock_rate;
"depth", G_TYPE_INT, 16,
"rate", G_TYPE_INT, clock_rate, "channels", G_TYPE_INT, channels, NULL);
- gst_pad_set_caps (depayload->srcpad, srccaps);
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
- return TRUE;
+ return res;
+
+ /* ERRORS */
+no_clockrate:
+ {
+ GST_ERROR_OBJECT (depayload, "no clock-rate specified");
+ return FALSE;
+ }
+no_channels:
+ {
+ GST_ERROR_OBJECT (depayload, "no channels specified");
+ return FALSE;
+ }
}
static GstBuffer *
{
GstRtpL16Depay *rtpL16depay;
GstBuffer *outbuf;
+ gint payload_len;
+ gboolean marker;
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
- {
- gint payload_len;
-
- payload_len = gst_rtp_buffer_get_payload_len (buf);
+ payload_len = gst_rtp_buffer_get_payload_len (buf);
- if (payload_len <= 0)
- goto empty_packet;
+ if (payload_len <= 0)
+ goto empty_packet;
- GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
+ GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
- outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+ outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+ marker = gst_rtp_buffer_get_marker (buf);
- return outbuf;
+ if (marker) {
+ /* mark talk spurt with DISCONT */
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
- return NULL;
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
- ("Packet did not validate."), (NULL));
- return NULL;
- }
+ return outbuf;
+
+ /* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
GstRtpL16Pay *rtpL16pay;
GstStructure *structure;
gint channels, rate;
+ gboolean res;
rtpL16pay = GST_RTP_L16_PAY (basepayload);
goto no_channels;
gst_basertppayload_set_options (basepayload, "audio", TRUE, "L16", rate);
- gst_basertppayload_set_outcaps (basepayload,
+ res = gst_basertppayload_set_outcaps (basepayload,
"channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, rate, NULL);
rtpL16pay->rate = rate;
rtpL16pay->channels = channels;
- return TRUE;
+ return res;
/* ERRORS */
no_rate:
{
GstStructure *structure;
GstRtpAC3Depay *rtpac3depay;
- gint clock_rate = 90000; /* default */
+ gint clock_rate;
+ GstCaps *srccaps;
+ gboolean res;
rtpac3depay = GST_RTP_AC3_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
- return TRUE;
+ srccaps = gst_caps_new_simple ("audio/ac3", NULL);
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
+ gst_caps_unref (srccaps);
+
+ return res;
}
struct frmsize_s
rtpac3depay = GST_RTP_AC3_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
{
gint payload_len;
guint8 *payload;
return NULL;
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtpac3depay, STREAM, DECODE,
- ("Packet did not validate."), (NULL));
- return NULL;
- }
-#if 0
-bad_payload:
- {
- GST_ELEMENT_WARNING (rtpac3depay, STREAM, DECODE,
- ("Unexpected payload type."), (NULL));
- return NULL;
- }
-#endif
+ /* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpac3depay, STREAM, DECODE,
const gchar *params;
const gchar *str, *type;
gint clock_rate, need_clock_rate;
+ gboolean res;
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
if ((str = gst_structure_get_string (structure, "encoding-name"))) {
if (strcmp (str, "AMR") == 0) {
rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_NB;
- clock_rate = need_clock_rate = 8000;
+ need_clock_rate = 8000;
type = "audio/AMR";
} else if (strcmp (str, "AMR-WB") == 0) {
rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB;
- clock_rate = need_clock_rate = 16000;
+ need_clock_rate = 16000;
type = "audio/AMR-WB";
} else
goto invalid_mode;
rtpamrdepay->channels = atoi (params);
}
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = need_clock_rate;
depayload->clock_rate = clock_rate;
/* we require 1 channel, 8000 Hz, octet aligned, no CRC,
srccaps = gst_caps_new_simple (type,
"channels", G_TYPE_INT, rtpamrdepay->channels,
"rate", G_TYPE_INT, clock_rate, NULL);
-
- gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+ res = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
- rtpamrdepay->negotiated = TRUE;
-
- return TRUE;
+ return res;
/* ERRORS */
invalid_mode:
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
- if (!rtpamrdepay->negotiated)
- goto not_negotiated;
-
- if (!gst_rtp_buffer_validate (buf))
- goto invalid_packet;
-
/* setup frame size pointer */
if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB)
frame_size = nb_frame_size;
GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;
if (marker) {
- /* marker bit marks a discont buffer */
+ /* marker bit marks a discont buffer after a talkspurt. */
GST_DEBUG_OBJECT (depayload, "marker bit was set");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
marker = FALSE;
}
- gst_buffer_set_caps (outbuf,
- GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload)));
-
GST_DEBUG_OBJECT (depayload, "pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
}
return outbuf;
/* ERRORS */
-invalid_packet:
- {
- GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
- (NULL), ("AMR RTP packet did not validate"));
- goto bad_packet;
- }
-not_negotiated:
- {
- GST_ELEMENT_ERROR (rtpamrdepay, STREAM, NOT_IMPLEMENTED,
- (NULL), ("not negotiated"));
- return NULL;
- }
too_small:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
{
GstBaseRTPDepayload depayload;
- gboolean negotiated;
GstRtpAMRDepayMode mode;
gboolean octet_align;
GstStructure *structure;
GstRTPDVDepay *rtpdvdepay;
GstCaps *srccaps;
- gint clock_rate = 90000; /* default */
+ gint clock_rate;
gboolean systemstream, ret;
const gchar *encode, *media;
structure = gst_caps_get_structure (caps, 0);
- if (gst_structure_has_field (structure, "clock-rate"))
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; /* default */
+ depayload->clock_rate = clock_rate;
/* we really need the encode property to figure out the frame size, it's also
* required by the spec */
ret = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
- depayload->clock_rate = clock_rate;
- rtpdvdepay->negotiated = TRUE;
-
return ret;
/* ERRORS */
GstRTPDVDepay *dvdepay = GST_RTP_DV_DEPAY (base);
gboolean marker;
- if (!dvdepay->negotiated)
- goto not_negotiated;
-
marker = gst_rtp_buffer_get_marker (in);
/* Check if the received packet contains (the start of) a new frame, we do
dvdepay->prev_ts = rtp_ts;
}
return out;
-
- /* ERRORS */
-not_negotiated:
- {
- GST_ELEMENT_ERROR (dvdepay, STREAM, NOT_IMPLEMENTED,
- (NULL), ("not negotiated"));
- return NULL;
- }
}
static void
depay->prev_ts = -1;
depay->have_header = FALSE;
depay->frame_nr = 0;
- depay->negotiated = FALSE;
}
static GstStateChangeReturn
gint width, height;
gint rate_num, rate_denom;
- gboolean negotiated;
};
struct _GstRTPDVDepayClass
GstCaps *srccaps;
GstStructure *structure;
gboolean ret;
- gint clock_rate = 8000; /* default */
+ gint clock_rate;
const gchar *encoding_name;
gint bitrate;
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 8000; /* default */
depayload->clock_rate = clock_rate;
encoding_name = gst_structure_get_string (structure, "encoding-name");
static GstBuffer *
gst_rtp_g726_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
- GstCaps *srccaps;
GstBuffer *outbuf = NULL;
+ gboolean marker;
+
+ marker = gst_rtp_buffer_get_marker (buf);
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
- GST_BUFFER_SIZE (buf),
- gst_rtp_buffer_get_marker (buf),
+ GST_BUFFER_SIZE (buf), marker,
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
- srccaps = GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
- if (!srccaps) {
- /* Set the default caps */
- srccaps = gst_caps_new_simple ("audio/x-adpcm",
- "channels", G_TYPE_INT, 1,
- "rate", G_TYPE_INT, SAMPLE_RATE,
- "bitrate", G_TYPE_INT, DEFAULT_BIT_RATE,
- "layout", G_TYPE_STRING, LAYOUT_G726, NULL);
- gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
- gst_caps_unref (srccaps);
- }
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+ if (marker) {
+ /* mark start of talkspurt with discont */
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ }
+
return outbuf;
}
depayload = GST_BASE_RTP_DEPAYLOAD (rtpg729depay);
- depayload->clock_rate = 8000;
gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
}
if (clock_rate != 8000)
goto wrong_clock_rate;
+ depayload->clock_rate = clock_rate;
+
srccaps = gst_caps_new_simple ("audio/G729",
"channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, clock_rate, NULL);
ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
- rtpg729depay->negotiated = ret;
-
return ret;
/* ERRORS */
GstRtpG729Depay *rtpg729depay;
GstBuffer *outbuf = NULL;
gint payload_len;
+ gboolean marker;
rtpg729depay = GST_RTP_G729_DEPAY (depayload);
- if (!rtpg729depay->negotiated)
- goto not_negotiated;
-
- if (!gst_rtp_buffer_validate (buf)) {
- GST_ELEMENT_WARNING (rtpg729depay, STREAM, DECODE,
- (NULL), ("G729 RTP packet did not validate"));
- goto bad_packet;
- }
-
payload_len = gst_rtp_buffer_get_payload_len (buf);
/* At least 2 bytes (CNG from G729 Annex B) */
}
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+ marker = gst_rtp_buffer_get_marker (buf);
+
+ if (marker) {
+ /* marker bit starts talkspurt */
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ }
GST_DEBUG ("gst_rtp_g729_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return outbuf;
/* ERRORS */
-not_negotiated:
- {
- GST_ELEMENT_ERROR (rtpg729depay, STREAM, NOT_IMPLEMENTED,
- (NULL), ("not negotiated"));
- return NULL;
- }
bad_packet:
{
/* no fatal error */
{
GstBaseRTPDepayload depayload;
- gboolean negotiated;
-
gint channels;
};
GstCaps *srccaps;
gboolean ret;
GstStructure *structure;
- gint clock_rate = 8000; /* default */
+ gint clock_rate;
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 8000; /* default */
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("audio/x-gsm",
gst_rtp_gsm_depay_process (GstBaseRTPDepayload * _depayload, GstBuffer * buf)
{
GstBuffer *outbuf = NULL;
+ gboolean marker;
+
+ marker = gst_rtp_buffer_get_marker (buf);
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
- GST_BUFFER_SIZE (buf),
- gst_rtp_buffer_get_marker (buf),
+ GST_BUFFER_SIZE (buf), marker,
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+ if (marker) {
+ /* mark start of talkspurt with DISCONT */
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ }
+
return outbuf;
}
{
const char *stname;
GstStructure *structure;
+ gboolean res;
structure = gst_caps_get_structure (caps, 0);
stname = gst_structure_get_name (structure);
- if (0 == strcmp ("audio/x-gsm", stname)) {
- gst_basertppayload_set_options (payload, "audio", FALSE, "GSM", 8000);
- } else {
- return FALSE;
- }
+ if (strcmp ("audio/x-gsm", stname))
+ goto invalid_type;
+
+ gst_basertppayload_set_options (payload, "audio", FALSE, "GSM", 8000);
+ res = gst_basertppayload_set_outcaps (payload, NULL);
- gst_basertppayload_set_outcaps (payload, NULL);
+ return res;
- return TRUE;
+ /* ERRORS */
+invalid_type:
+ {
+ GST_WARNING_OBJECT (payload, "invalid media type received");
+ return FALSE;
+ }
}
static GstFlowReturn
{
GstCaps *srccaps;
GstStructure *structure = gst_caps_get_structure (caps, 0);
- gint clock_rate = 90000; /* default */
+ gint clock_rate;
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; /* default */
filter->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("video/x-h263",
static GstBuffer *
gst_rtp_h263_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
-
GstRtpH263Depay *rtph263depay;
GstBuffer *outbuf;
gint payload_len;
guint8 *payload;
guint header_len;
-
guint SBIT, EBIT;
gboolean F, P, M;
gboolean I;
rtph263depay = GST_RTP_H263_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf)) {
GST_LOG_OBJECT (depayload, "Discont buffer, flushing adapter");
}
return NULL;
-
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtph263depay, STREAM, DECODE,
- ("Packet did not validate"), (NULL));
- return NULL;
- }
}
static GstStateChangeReturn
static gboolean
gst_rtp_h263_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
- payload->pt = GST_RTP_PAYLOAD_H263;
+ gboolean res;
+
gst_basertppayload_set_options (payload, "video", TRUE, "H263", 90000);
- gst_basertppayload_set_outcaps (payload, NULL);
+ res = gst_basertppayload_set_outcaps (payload, NULL);
- return TRUE;
+ return res;
}
static guint
memcpy (payload, data + curpos, payload_len);
- GST_BUFFER_TIMESTAMP (outbuf) = rtph263pay->first_ts;
+ GST_BUFFER_TIMESTAMP (outbuf) = rtph263pay->first_timestamp;
+ GST_BUFFER_DURATION (outbuf) = rtph263pay->first_duration;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtph263pay), outbuf);
rtph263pay = GST_RTP_H263_PAY (payload);
size = GST_BUFFER_SIZE (buffer);
- rtph263pay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+ rtph263pay->first_timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ rtph263pay->first_duration = GST_BUFFER_DURATION (buffer);
/* we always encode and flush a full picture */
gst_adapter_push (rtph263pay->adapter, buffer);
GstBaseRTPPayload payload;
GstAdapter *adapter;
- GstClockTime first_ts;
+ GstClockTime first_timestamp;
+ GstClockTime first_duration;
};
struct _GstRtpH263PayClass
{
GstCaps *srccaps = NULL;
GstStructure *structure = gst_caps_get_structure (caps, 0);
- gint clock_rate = 90000; /* default */
+ gint clock_rate;
const gchar *encoding_name = NULL;
+ gboolean res;
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; /* default */
filter->clock_rate = clock_rate;
encoding_name = gst_structure_get_string (structure, "encoding-name");
if (!srccaps)
goto no_caps;
- gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (filter), srccaps);
+ res = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (filter), srccaps);
gst_caps_unref (srccaps);
- return TRUE;
+ return res;
/* ERRORS */
no_encoding_name:
static GstBuffer *
gst_rtp_h263p_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
-
GstRtpH263PDepay *rtph263pdepay;
GstBuffer *outbuf;
rtph263pdepay = GST_RTP_H263P_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf)) {
GST_LOG_OBJECT (depayload, "DISCONT, flushing adapter");
header_len = 2;
if (payload_len < header_len)
- goto bad_packet;
+ goto too_small;
M = gst_rtp_buffer_get_marker (buf);
}
if ((!P && payload_len < header_len) || (P && payload_len < header_len - 2))
- goto bad_packet;
+ goto too_small;
if (P) {
/* FIXME, have to make the packet writable hear. Better to reset these
}
return NULL;
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtph263pdepay, STREAM, DECODE,
- ("Packet did not validate"), (NULL));
- return NULL;
- }
too_small:
{
GST_ELEMENT_WARNING (rtph263pdepay, STREAM, DECODE,
GST_DEBUG_CATEGORY_INIT (rtph263ppay_debug, "rtph263ppay",
0, "rtph263ppay (RFC 4629)");
-
}
static void
static gboolean
gst_rtp_h263p_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
+ gboolean res;
+
gst_basertppayload_set_options (payload, "video", TRUE, "H263-1998", 90000);
- gst_basertppayload_set_outcaps (payload, NULL);
+ res = gst_basertppayload_set_outcaps (payload, NULL);
- return TRUE;
+ return res;
}
static void
}
}
-
-
static GstFlowReturn
gst_rtp_h263p_pay_flush (GstRtpH263PPay * rtph263ppay)
{
payload[0] = (fragmented && !found_gob) ? 0x00 : 0x04;
payload[1] = 0;
- GST_BUFFER_TIMESTAMP (outbuf) = rtph263ppay->first_ts;
+ GST_BUFFER_TIMESTAMP (outbuf) = rtph263ppay->first_timestamp;
+ GST_BUFFER_DURATION (outbuf) = rtph263ppay->first_duration;
+
gst_adapter_flush (rtph263ppay->adapter, towrite);
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtph263ppay), outbuf);
rtph263ppay = GST_RTP_H263P_PAY (payload);
size = GST_BUFFER_SIZE (buffer);
- rtph263ppay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+ rtph263ppay->first_timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ rtph263ppay->first_duration = GST_BUFFER_DURATION (buffer);
/* we always encode and flush a full picture */
gst_adapter_push (rtph263ppay->adapter, buffer);
GstBaseRTPPayload payload;
GstAdapter *adapter;
- GstClockTime first_ts;
+ GstClockTime first_timestamp;
+ GstClockTime first_duration;
GstFragmentationMode fragmentation_mode;
};
static gboolean
gst_rtp_h264_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
- GstCaps *srccaps = NULL;
- gint clock_rate = 90000;
+ GstCaps *srccaps;
+ gint clock_rate;
GstStructure *structure = gst_caps_get_structure (caps, 0);
GstRtpH264Depay *rtph264depay;
const gchar *ps, *profile;
GstBuffer *codec_data;
guint8 *b64;
+ gboolean res;
rtph264depay = GST_RTP_H264_DEPAY (depayload);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000;
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("video/x-h264", NULL);
"codec_data", GST_TYPE_BUFFER, codec_data, NULL);
}
- gst_pad_set_caps (depayload->srcpad, srccaps);
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
- return TRUE;
+ return res;
/* ERRORS */
incomplete_caps:
{
GST_DEBUG_OBJECT (depayload, "we have incomplete caps");
+ gst_caps_unref (srccaps);
return FALSE;
}
}
rtph264depay = GST_RTP_H264_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf)) {
gst_adapter_clear (rtph264depay->adapter);
return NULL;
/* ERRORS */
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtph264depay, STREAM, DECODE,
- (NULL), ("Packet did not validate"));
- return NULL;
- }
undefined_type:
{
GST_ELEMENT_WARNING (rtph264depay, STREAM, DECODE,
data += 2;
size -= 2;
+ GST_LOG_OBJECT (rtph264pay, "SPS %d size %d", i, nal_size);
+
if (size < nal_size)
goto avcc_error;
data += 2;
size -= 2;
+ GST_LOG_OBJECT (rtph264pay, "PPS %d size %d", i, nal_size);
+
if (size < nal_size)
goto avcc_error;
static const GstElementDetails gst_rtp_ilbc_depay_details =
GST_ELEMENT_DETAILS ("RTP iLBC packet depayloader",
"Codec/Depayloader/Network",
- "Extracts iLBC audio from RTP packets",
+ "Extracts iLBC audio from RTP packets (RFC 3952)",
"Philippe Kalaf <philippe.kalaf@collabora.co.uk>");
/* RtpiLBCDepay signals and args */
GstCaps *srccaps;
GstStructure *structure;
const gchar *mode_str = NULL;
- gint mode;
+ gint mode, clock_rate;
gboolean ret;
structure = gst_caps_get_structure (caps, 0);
mode = rtpilbcdepay->mode;
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 8000;
+ depayload->clock_rate = clock_rate;
+
/* parse mode, if we can */
mode_str = gst_structure_get_string (structure, "mode");
if (mode_str) {
GST_DEBUG ("set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
gst_caps_unref (srccaps);
- /* always fixed clock rate of 8000 */
- depayload->clock_rate = 8000;
-
return ret;
}
gst_rtp_ilbc_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf;
+ gboolean marker;
+
+ marker = gst_rtp_buffer_get_marker (buf);
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
- GST_BUFFER_SIZE (buf),
- gst_rtp_buffer_get_marker (buf),
+ GST_BUFFER_SIZE (buf), marker,
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+ if (marker) {
+ /* mark start of talkspurt with DISCONT */
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ }
+
return outbuf;
}
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "mode", &mode);
- if (mode != 20 && mode != 30)
- goto wrong_mode;
-
payload_name = gst_structure_get_name (structure);
if (g_ascii_strcasecmp ("audio/x-iLBC", payload_name))
goto wrong_caps;
+ if (!gst_structure_get_int (structure, "mode", &mode))
+ goto no_mode;
+
+ if (mode != 20 && mode != 30)
+ goto wrong_mode;
+
gst_basertppayload_set_options (basertppayload, "audio", TRUE, "ILBC", 8000);
/* set options for this frame based audio codec */
gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload,
mode, mode == 30 ? 50 : 38);
-
mode_str = g_strdup_printf ("%d", mode);
ret =
gst_basertppayload_set_outcaps (basertppayload, "mode", G_TYPE_STRING,
return ret;
/* ERRORS */
-wrong_mode:
- {
- GST_ERROR_OBJECT (rtpilbcpay, "mode must be 20 or 30, received %d", mode);
- return FALSE;
- }
wrong_caps:
{
GST_ERROR_OBJECT (rtpilbcpay, "expected audio/x-iLBC, received %s",
payload_name);
return FALSE;
}
+no_mode:
+ {
+ GST_ERROR_OBJECT (rtpilbcpay, "did not receive a mode");
+ return FALSE;
+ }
+wrong_mode:
+ {
+ GST_ERROR_OBJECT (rtpilbcpay, "mode must be 20 or 30, received %d", mode);
+ return FALSE;
+ }
mode_changed:
{
GST_ERROR_OBJECT (rtpilbcpay, "Mode has changed from %d to %d! "
GstCaps *srccaps;
GstStructure *structure;
GstRtpMP1SDepay *rtpmp1sdepay;
- gint clock_rate = 90000; /* default */
+ gint clock_rate;
+ gboolean res;
rtpmp1sdepay = GST_RTP_MP1S_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("video/mpeg",
"systemstream", G_TYPE_BOOLEAN, TRUE, NULL);
- gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+ res = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
- return TRUE;
+ return res;
}
static GstBuffer *
{
GstRtpMP1SDepay *rtpmp1sdepay;
GstBuffer *outbuf;
- gint payload_len;
rtpmp1sdepay = GST_RTP_MP1S_DEPAY (depayload);
- if (G_UNLIKELY (!gst_rtp_buffer_validate (buf)))
- goto bad_packet;
-
- payload_len = gst_rtp_buffer_get_payload_len (buf);
-
- outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 0, -1);
- gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
+ outbuf = gst_rtp_buffer_get_payload_buffer (buf);
GST_DEBUG ("gst_rtp_mp1s_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return outbuf;
-
- /* ERRORS */
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtpmp1sdepay, STREAM, DECODE,
- (NULL), ("Packet did not validate"));
- return NULL;
- }
}
static void
GstCaps *srccaps;
GstStructure *structure;
GstRtpMP2TDepay *rtpmp2tdepay;
- gint clock_rate = 90000; /* default */
+ gint clock_rate;
+ gboolean res;
rtpmp2tdepay = GST_RTP_MP2T_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("video/mpegts",
"packetsize", G_TYPE_INT, 188,
"systemstream", G_TYPE_BOOLEAN, TRUE, NULL);
- gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+ res = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
- return TRUE;
+ return res;
}
static GstBuffer *
rtpmp2tdepay = GST_RTP_MP2T_DEPAY (depayload);
- if (G_UNLIKELY (!gst_rtp_buffer_validate (buf)))
- goto bad_packet;
-
payload_len = gst_rtp_buffer_get_payload_len (buf);
if (G_UNLIKELY (payload_len <= rtpmp2tdepay->skip_first_bytes))
outbuf =
gst_rtp_buffer_get_payload_subbuffer (buf, rtpmp2tdepay->skip_first_bytes,
-1);
- gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
GST_DEBUG ("gst_rtp_mp2t_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return outbuf;
/* ERRORS */
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtpmp2tdepay, STREAM, DECODE,
- (NULL), ("Packet did not validate"));
- return NULL;
- }
empty_packet:
{
GST_ELEMENT_WARNING (rtpmp2tdepay, STREAM, DECODE,
GstRtpMP4ADepay *rtpmp4adepay;
GstCaps *srccaps;
const gchar *str;
- gint clock_rate = 90000; /* default */
- gint object_type = 2; /* AAC LC default */
+ gint clock_rate;
+ gint object_type;
gint channels = 2; /* default */
+ gboolean res;
rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
- if (gst_structure_has_field (structure, "clock-rate"))
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
- if (gst_structure_has_field (structure, "object"))
- gst_structure_get_int (structure, "object", &object_type);
+ if (!gst_structure_get_int (structure, "object", &object_type))
+ object_type = 2; /* AAC LC default */
srccaps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 4,
}
}
bad_config:
- gst_pad_set_caps (depayload->srcpad, srccaps);
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
- return TRUE;
+ return res;
}
static GstBuffer *
rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf)) {
gst_adapter_clear (rtpmp4adepay->adapter);
return NULL;
/* ERRORS */
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtpmp4adepay, STREAM, DECODE,
- ("Packet did not validate"), (NULL));
- return NULL;
- }
wrong_size:
{
GST_ELEMENT_WARNING (rtpmp4adepay, STREAM, DECODE,
}
}
-static void
+static gboolean
gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay)
{
gchar *config;
GValue v = { 0 };
+ gboolean res;
g_value_init (&v, GST_TYPE_BUFFER);
gst_value_set_buffer (&v, rtpmp4apay->config);
config = gst_value_serialize (&v);
- gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4apay),
+ res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4apay),
"cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL);
g_value_unset (&v);
g_free (config);
+
+ return res;
}
static gboolean
GstRtpMP4APay *rtpmp4apay;
GstStructure *structure;
const GValue *codec_data;
+ gboolean res;
rtpmp4apay = GST_RTP_MP4A_PAY (payload);
guint8 *config;
guint8 *data;
guint size, i;
- gboolean res;
buffer = gst_value_get_buffer (codec_data);
GST_LOG_OBJECT (rtpmp4apay, "configuring codec_data");
gst_basertppayload_set_options (payload, "audio", TRUE, "MP4A-LATM",
rtpmp4apay->rate);
- gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
+ res = gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
- return TRUE;
+ return res;
/* ERRORS */
config_failed:
static gboolean
gst_rtp_mp4g_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
-
GstStructure *structure;
GstRtpMP4GDepay *rtpmp4gdepay;
GstCaps *srccaps = NULL;
const gchar *str;
- gint clock_rate = 90000; /* default */
+ gint clock_rate;
gint someint;
+ gboolean res;
rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
if ((str = gst_structure_get_string (structure, "media"))) {
}
}
- gst_pad_set_caps (depayload->srcpad, srccaps);
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
- return TRUE;
+ return res;
/* ERRORS */
unknown_media:
rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf)) {
GST_DEBUG_OBJECT (rtpmp4gdepay, "received DISCONT");
avail = gst_adapter_available (rtpmp4gdepay->adapter);
outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail);
- gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return NULL;
/* ERRORS */
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtpmp4gdepay, STREAM, DECODE,
- ("Packet did not validate."), (NULL));
- return NULL;
- }
short_payload:
{
GST_ELEMENT_WARNING (rtpmp4gdepay, STREAM, DECODE,
g_object_unref (rtpmp4gpay->adapter);
rtpmp4gpay->adapter = NULL;
+
g_free (rtpmp4gpay->params);
rtpmp4gpay->params = NULL;
}
}
-static void
+static gboolean
gst_rtp_mp4g_pay_new_caps (GstRtpMP4GPay * rtpmp4gpay)
{
gchar *config;
GValue v = { 0 };
+ gboolean res;
#define MP4GCAPS \
"streamtype", G_TYPE_STRING, rtpmp4gpay->streamtype, \
/* hmm, silly */
if (rtpmp4gpay->params) {
- gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
+ res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
"encoding-params", G_TYPE_STRING, rtpmp4gpay->params, MP4GCAPS);
} else {
- gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
+ res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
MP4GCAPS);
}
g_free (config);
#undef MP4GCAPS
+ return res;
}
static gboolean
GstStructure *structure;
const GValue *codec_data;
gchar *media_type = NULL;
+ gboolean res;
rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
GstBuffer *buffer;
const gchar *name;
- gboolean res;
buffer = gst_value_get_buffer (codec_data);
GST_LOG_OBJECT (rtpmp4gpay, "configuring codec_data");
gst_basertppayload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC",
rtpmp4gpay->rate);
- gst_rtp_mp4g_pay_new_caps (rtpmp4gpay);
+ res = gst_rtp_mp4g_pay_new_caps (rtpmp4gpay);
- return TRUE;
+ return res;
/* ERRORS */
config_failed:
GstRtpMP4VDepay *rtpmp4vdepay;
GstCaps *srccaps;
const gchar *str;
- gint clock_rate = 90000; /* default */
+ gint clock_rate;
+ gboolean res;
rtpmp4vdepay = GST_RTP_MP4V_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
- if (gst_structure_has_field (structure, "clock-rate"))
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("video/mpeg",
g_warning ("cannot convert config to buffer");
}
}
- gst_pad_set_caps (depayload->srcpad, srccaps);
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
- return TRUE;
+ return res;
}
static GstBuffer *
rtpmp4vdepay = GST_RTP_MP4V_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf))
gst_adapter_clear (rtpmp4vdepay->adapter);
avail = gst_adapter_available (rtpmp4vdepay->adapter);
outbuf = gst_adapter_take_buffer (rtpmp4vdepay->adapter, avail);
- gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
GST_DEBUG ("gst_rtp_mp4v_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return outbuf;
}
return NULL;
-
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtpmp4vdepay, STREAM, DECODE,
- ("Packet did not validate"), (NULL));
- return NULL;
- }
}
static GstStateChangeReturn
G_OBJECT_CLASS (parent_class)->finalize (object);
}
-static void
+static gboolean
gst_rtp_mp4v_pay_new_caps (GstRtpMP4VPay * rtpmp4vpay)
{
gchar *profile, *config;
GValue v = { 0 };
+ gboolean res;
profile = g_strdup_printf ("%d", rtpmp4vpay->profile);
g_value_init (&v, GST_TYPE_BUFFER);
gst_value_set_buffer (&v, rtpmp4vpay->config);
config = gst_value_serialize (&v);
- gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4vpay),
+ res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4vpay),
"profile-level-id", G_TYPE_STRING, profile,
"config", G_TYPE_STRING, config, NULL);
g_free (profile);
g_free (config);
+
+ return res;
}
static gboolean
GstRtpMP4VPay *rtpmp4vpay;
GstStructure *structure;
const GValue *codec_data;
+ gboolean res;
rtpmp4vpay = GST_RTP_MP4V_PAY (payload);
gst_basertppayload_set_options (payload, "video", TRUE, "MP4V-ES",
rtpmp4vpay->rate);
+ res = TRUE;
+
structure = gst_caps_get_structure (caps, 0);
codec_data = gst_structure_get_value (structure, "codec_data");
if (codec_data) {
if (rtpmp4vpay->config)
gst_buffer_unref (rtpmp4vpay->config);
rtpmp4vpay->config = gst_buffer_copy (buffer);
- gst_rtp_mp4v_pay_new_caps (rtpmp4vpay);
+ res = gst_rtp_mp4v_pay_new_caps (rtpmp4vpay);
}
}
done:
- return TRUE;
+ return res;
}
static void
{
GstStructure *structure;
GstRtpMPADepay *rtpmpadepay;
- gint clock_rate = 90000; /* default */
+ GstCaps *outcaps;
+ gint clock_rate;
+ gboolean res;
rtpmpadepay = GST_RTP_MPA_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000;
depayload->clock_rate = clock_rate;
- return TRUE;
+ outcaps = gst_caps_new_simple ("audio/mpeg", NULL);
+ res = gst_pad_set_caps (depayload->srcpad, outcaps);
+ gst_caps_unref (outcaps);
+
+ return res;
}
static GstBuffer *
rtpmpadepay = GST_RTP_MPA_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
{
gint payload_len;
guint8 *payload;
guint16 frag_offset;
+ gboolean marker;
payload_len = gst_rtp_buffer_get_payload_len (buf);
- payload = gst_rtp_buffer_get_payload (buf);
if (payload_len <= 4)
goto empty_packet;
+ payload = gst_rtp_buffer_get_payload (buf);
/* strip off header
*
* 0 1 2 3
/* subbuffer skipping the 4 header bytes */
outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 4, -1);
+ marker = gst_rtp_buffer_get_marker (buf);
+ if (marker) {
+ /* mark start of talkspurt with discont */
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ }
GST_DEBUG_OBJECT (rtpmpadepay,
"gst_rtp_mpa_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return NULL;
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE,
- ("Packet did not validate."), (NULL));
- return NULL;
- }
-#if 0
-bad_payload:
- {
- GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE,
- ("Unexpected payload type."), (NULL));
- return NULL;
- }
-#endif
+ /* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE,
static gboolean
gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
+ gboolean res;
+
gst_basertppayload_set_options (payload, "audio", TRUE, "MPA", 90000);
- gst_basertppayload_set_outcaps (payload, NULL);
+ res = gst_basertppayload_set_outcaps (payload, NULL);
- return TRUE;
+ return res;
}
static GstFlowReturn
{
GstStructure *structure;
GstRtpMPVDepay *rtpmpvdepay;
- gint clock_rate = 90000; /* default */
+ gint clock_rate;
+ GstCaps *outcaps;
+ gboolean res;
rtpmpvdepay = GST_RTP_MPV_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
- return TRUE;
+ outcaps = gst_caps_new_simple ("video/mpeg",
+ "mpegversion", G_TYPE_INT, 2,
+ "systemstream", G_TYPE_BOOLEAN, FALSE, NULL);
+ res = gst_pad_set_caps (depayload->srcpad, outcaps);
+ gst_caps_unref (outcaps);
+
+ return res;
}
static GstBuffer *
rtpmpvdepay = GST_RTP_MPV_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
{
gint payload_len, payload_header;
guint8 *payload;
return NULL;
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtpmpvdepay, STREAM, DECODE,
- (NULL), ("Packet did not validate."));
- return NULL;
- }
+ /* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpmpvdepay, STREAM, DECODE,
{
const char *stname;
GstStructure *structure;
+ gboolean res;
structure = gst_caps_get_structure (caps, 0);
stname = gst_structure_get_name (structure);
gst_basertppayload_set_options (payload, "video", FALSE, "MPV", 90000);
- gst_basertppayload_set_outcaps (payload, NULL);
+ res = gst_basertppayload_set_outcaps (payload, NULL);
- return TRUE;
+ return res;
}
static GstFlowReturn
GstCaps *srccaps;
GstStructure *structure;
gboolean ret;
- gint clock_rate = 8000; /* default */
+ gint clock_rate;
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 8000; /* default */
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("audio/x-alaw",
static GstBuffer *
gst_rtp_pcma_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
- GstCaps *srccaps;
GstBuffer *outbuf = NULL;
+ gboolean marker;
+ guint len;
+
+ marker = gst_rtp_buffer_get_marker (buf);
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
- GST_BUFFER_SIZE (buf),
- gst_rtp_buffer_get_marker (buf),
+ GST_BUFFER_SIZE (buf), marker,
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
- srccaps = GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
- if (!srccaps) {
- /* Set the default caps */
- srccaps = gst_caps_new_simple ("audio/x-alaw",
- "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
- gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
- gst_caps_unref (srccaps);
- }
-
+ len = gst_rtp_buffer_get_payload_len (buf);
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+ GST_BUFFER_DURATION (outbuf) =
+ gst_util_uint64_scale_int (len, GST_SECOND, depayload->clock_rate);
+
+ if (marker) {
+ /* mark start of talkspurt with DISCONT */
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ }
+
return outbuf;
}
static gboolean
gst_rtp_pcma_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
+ gboolean res;
+
payload->pt = GST_RTP_PAYLOAD_PCMA;
- gst_basertppayload_set_options (payload, "audio", FALSE, "PCMA", 8000);
- gst_basertppayload_set_outcaps (payload, NULL);
+ gst_basertppayload_set_options (payload, "audio", FALSE, "PCMA", 8000);
+ res = gst_basertppayload_set_outcaps (payload, NULL);
- return TRUE;
+ return res;
}
gboolean
GstCaps *srccaps;
GstStructure *structure;
gboolean ret;
- gint clock_rate = 8000; /* default */
+ gint clock_rate;
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 8000; /* default */
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("audio/x-mulaw",
static GstBuffer *
gst_rtp_pcmu_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
- GstCaps *srccaps;
GstBuffer *outbuf = NULL;
guint len;
+ gboolean marker;
+
+ marker = gst_rtp_buffer_get_marker (buf);
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
- GST_BUFFER_SIZE (buf),
- gst_rtp_buffer_get_marker (buf),
+ GST_BUFFER_SIZE (buf), marker,
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
- srccaps = GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
- if (!srccaps) {
- /* Set the default caps */
- srccaps = gst_caps_new_simple ("audio/x-mulaw",
- "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
- gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
- gst_caps_unref (srccaps);
- }
-
len = gst_rtp_buffer_get_payload_len (buf);
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (len, GST_SECOND, depayload->clock_rate);
+ if (marker) {
+ /* mark start of talkspurt with DISCONT */
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ }
+
return outbuf;
}
static gboolean
gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
+ gboolean res;
+
payload->pt = GST_RTP_PAYLOAD_PCMU;
- gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
- gst_basertppayload_set_outcaps (payload, NULL);
+ gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
+ res = gst_basertppayload_set_outcaps (payload, NULL);
- return TRUE;
+ return res;
}
gboolean
gst_rtp_speex_depay_init (GstRtpSPEEXDepay * rtpspeexdepay,
GstRtpSPEEXDepayClass * klass)
{
- GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay)->clock_rate = 8000;
}
static gint
GstBuffer *buf;
guint8 *data;
const gchar *params;
+ GstCaps *srccaps;
+ gboolean res;
rtpspeexdepay = GST_RTP_SPEEX_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ goto no_clockrate;
depayload->clock_rate = clock_rate;
if (!(params = gst_structure_get_string (structure, "encoding-params")))
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* reserved2 */
+ srccaps = gst_caps_new_simple ("audio/x-speex", NULL);
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
+ gst_caps_unref (srccaps);
+
+ gst_buffer_set_caps (buf, GST_PAD_CAPS (depayload->srcpad));
gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay), buf);
buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_speex_comment));
memcpy (GST_BUFFER_DATA (buf), gst_rtp_speex_comment,
sizeof (gst_rtp_speex_comment));
+ gst_buffer_set_caps (buf, GST_PAD_CAPS (depayload->srcpad));
gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay), buf);
- return TRUE;
+ return res;
+
+ /* ERRORS */
+no_clockrate:
+ {
+ GST_DEBUG_OBJECT (depayload, "no clock-rate specified");
+ return FALSE;
+ }
}
static GstBuffer *
/* nothing special to be done */
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+ GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
+
return outbuf;
}
guint32 version, header_size, rate, mode, nb_channels;
GstBaseRTPPayload *payload;
gchar *cstr;
+ gboolean res;
/* we need the header string (8), the version string (20), the version
* and the header length. */
gst_basertppayload_set_options (payload, "audio", FALSE, "SPEEX", rate);
cstr = g_strdup_printf ("%d", nb_channels);
- gst_basertppayload_set_outcaps (payload, "encoding-params",
+ res = gst_basertppayload_set_outcaps (payload, "encoding-params",
G_TYPE_STRING, cstr, NULL);
g_free (cstr);
- return TRUE;
+ return res;
/* ERRORS */
too_small:
gboolean
gst_rtp_sv3v_depay_setcaps (GstBaseRTPDepayload * filter, GstCaps * caps)
{
-
GstStructure *structure = gst_caps_get_structure (caps, 0);
- gint clock_rate = 90000; // default
-
- if (gst_structure_has_field (structure, "clock-rate"))
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ gint clock_rate;
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; // default
filter->clock_rate = clock_rate;
+ /* will set caps later */
+
return TRUE;
}
-
static GstBuffer *
gst_rtp_sv3v_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
-
GstRtpSV3VDepay *rtpsv3vdepay;
GstBuffer *outbuf;
guint16 seq;
rtpsv3vdepay = GST_RTP_SV3V_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
/* flush on sequence number gaps */
seq = gst_rtp_buffer_get_seq (buf);
if (seq != rtpsv3vdepay->nextseq) {
avail = gst_adapter_available (rtpsv3vdepay->adapter);
outbuf = gst_adapter_take_buffer (rtpsv3vdepay->adapter, avail);
- /* timestamp for complete buffer is that of last buffer as well */
- gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
-
return outbuf;
}
}
bad_packet:
{
GST_ELEMENT_WARNING (rtpsv3vdepay, STREAM, DECODE,
- ("Packet did not validate"), (NULL));
+ (NULL), ("Packet was too short"));
return NULL;
}
}
GstCaps *srccaps;
const gchar *delivery_method;
const gchar *configuration;
+ gboolean res;
rtptheoradepay = GST_RTP_THEORA_DEPAY (depayload);
/* set caps on pad and on header */
srccaps = gst_caps_new_simple ("video/x-theora", NULL);
- gst_pad_set_caps (depayload->srcpad, srccaps);
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
/* Clock rate is always 90000 according to draft-barbato-avt-rtp-theora-01 */
depayload->clock_rate = 90000;
- return TRUE;
+ return res;
/* ERRORS */
unsupported_delivery_method:
rtptheoradepay = GST_RTP_THEORA_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
payload_len = gst_rtp_buffer_get_payload_len (buf);
GST_DEBUG_OBJECT (depayload, "got RTP packet of size %d", payload_len);
return NULL;
}
/* ERORRS */
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtptheoradepay, STREAM, DECODE,
- (NULL), ("Packet did not validate"));
- return NULL;
- }
switch_failed:
{
GST_ELEMENT_ERROR (rtptheoradepay, STREAM, DECODE,
gst_rtp_theora_pay_class_init (GstRtpTheoraPayClass * klass)
{
GObjectClass *gobject_class;
-
GstElementClass *gstelement_class;
-
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gst_rtp_theora_pay_flush_packet (GstRtpTheoraPay * rtptheorapay)
{
GstFlowReturn ret;
-
guint8 *payload;
-
guint hlen;
/* check for empty packet */
encode_base64 (const guint8 * in, guint size, guint * len)
{
gchar *ret, *d;
-
static const gchar *v =
"ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/";
gst_rtp_theora_pay_finish_headers (GstBaseRTPPayload * basepayload)
{
GstRtpTheoraPay *rtptheorapay = GST_RTP_THEORA_PAY (basepayload);
-
GList *walk;
-
guint length, size, n_headers, configlen;
-
gchar *wstr, *hstr, *configuration;
-
guint8 *data, *config;
-
guint32 ident;
+ gboolean res;
GST_DEBUG_OBJECT (rtptheorapay, "finish headers");
wstr = g_strdup_printf ("%d", rtptheorapay->width);
hstr = g_strdup_printf ("%d", rtptheorapay->height);
gst_basertppayload_set_options (basepayload, "video", TRUE, "THEORA", 90000);
- gst_basertppayload_set_outcaps (basepayload,
+ res = gst_basertppayload_set_outcaps (basepayload,
"sampling", G_TYPE_STRING, "YCbCr-4:2:0",
"width", G_TYPE_STRING, wstr,
"height", G_TYPE_STRING, hstr,
g_free (hstr);
g_free (configuration);
- return TRUE;
+ return res;
/* ERRORS */
no_headers:
guint size)
{
GstRtpTheoraPay *rtptheorapay;
-
gint width, height;
rtptheorapay = GST_RTP_THEORA_PAY (basepayload);
GstBuffer * buffer)
{
GstRtpTheoraPay *rtptheorapay;
-
GstFlowReturn ret;
-
guint size, newsize;
-
guint8 *data;
-
guint packet_len;
-
GstClockTime duration, newduration, timestamp;
-
gboolean flush;
-
guint8 TDT;
-
guint plen;
-
guint8 *ppos, *payload;
-
gboolean fragmented;
rtptheorapay = GST_RTP_THEORA_PAY (basepayload);
GstCaps *srccaps;
const gchar *configuration;
gint clock_rate;
+ gboolean res;
rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (depayload);
/* set caps on pad and on header */
srccaps = gst_caps_new_simple ("audio/x-vorbis", NULL);
- gst_pad_set_caps (depayload->srcpad, srccaps);
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
- return TRUE;
+ return res;
/* ERRORS */
invalid_configuration:
rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (depayload);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
-
payload_len = gst_rtp_buffer_get_payload_len (buf);
GST_DEBUG_OBJECT (depayload, "got RTP packet of size %d", payload_len);
return NULL;
}
/* ERORRS */
-bad_packet:
- {
- GST_ELEMENT_WARNING (rtpvorbisdepay, STREAM, DECODE,
- (NULL), ("Packet did not validate"));
- return NULL;
- }
switch_failed:
{
GST_ELEMENT_WARNING (rtpvorbisdepay, STREAM, DECODE,
gchar *cstr, *configuration;
guint8 *data, *config;
guint32 ident;
+ gboolean res;
GST_DEBUG_OBJECT (rtpvorbispay, "finish headers");
cstr = g_strdup_printf ("%d", rtpvorbispay->channels);
gst_basertppayload_set_options (basepayload, "audio", TRUE, "VORBIS",
rtpvorbispay->rate);
- gst_basertppayload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING,
- cstr, "configuration", G_TYPE_STRING, configuration, NULL);
+ res =
+ gst_basertppayload_set_outcaps (basepayload, "encoding-params",
+ G_TYPE_STRING, cstr, "configuration", G_TYPE_STRING, configuration, NULL);
g_free (cstr);
g_free (configuration);
- return TRUE;
+ return res;
/* ERRORS */
no_headers:
{
GstStructure *structure;
GstRtpVRawDepay *rtpvrawdepay;
- gint clock_rate = 90000; /* default */
+ gint clock_rate;
const gchar *str, *type;
gint format, width, height, pgroup, xinc, yinc;
guint ystride, uvstride, yp, up, vp, outsize;
GstCaps *srccaps;
guint32 fourcc = 0;
+ gboolean res;
rtpvrawdepay = GST_RTP_VRAW_DEPAY (depayload);
yp = up = vp = uvstride = 0;
xinc = yinc = 1;
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
if (!(str = gst_structure_get_string (structure, "width")))
"format", GST_TYPE_FOURCC, fourcc,
"framerate", GST_TYPE_FRACTION, 0, 1, NULL);
- gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+ res = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
GST_DEBUG_OBJECT (depayload, "width %d, height %d, format %d", width, height,
ystride, uvstride);
GST_DEBUG_OBJECT (depayload, "outsize %u", outsize);
- return TRUE;
+ return res;
/* ERRORS */
no_width:
hstr = g_strdup_printf ("%d", rtpvrawpay->height);
gst_basertppayload_set_options (payload, "video", TRUE, "RAW", 90000);
- gst_basertppayload_set_outcaps (payload, "sampling", G_TYPE_STRING,
+ res = gst_basertppayload_set_outcaps (payload, "sampling", G_TYPE_STRING,
samplingstr, "depth", G_TYPE_STRING, depthstr, "width", G_TYPE_STRING,
wstr, "height", G_TYPE_STRING, hstr, "colorimetry", G_TYPE_STRING,
colorimetrystr, NULL);
g_free (wstr);
g_free (hstr);
- return TRUE;
+ return res;
/* ERRORS */
unknown_mask:
projects / platform / upstream / gstreamer.git / commitdiff