static gboolean
gst_rtp_amr_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpAMRPay *rtpamrpay;
gboolean res;
const GstStructure *s;
const gchar *str;
rtpamrpay = GST_RTP_AMR_PAY (basepayload);
/* figure out the mode Narrow or Wideband */
s = gst_caps_get_structure (caps, 0);
if ((str = gst_structure_get_name (s))) {
if (strcmp (str, "audio/AMR") == 0)
rtpamrpay->mode = GST_RTP_AMR_P_MODE_NB;
else if (strcmp (str, "audio/AMR-WB") == 0)
rtpamrpay->mode = GST_RTP_AMR_P_MODE_WB;
else
goto wrong_type;
} else
goto wrong_type;
if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
else
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "AMR-WB",
16000);
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, "1", "octet-align", G_TYPE_STRING, "1",
/* don't set the defaults
*
* "crc", G_TYPE_STRING, "0",
* "robust-sorting", G_TYPE_STRING, "0",
* "interleaving", G_TYPE_STRING, "0",
*/
NULL);
return res;
/* ERRORS */
wrong_type:
{
GST_ERROR_OBJECT (rtpamrpay, "unsupported media type '%s'",
GST_STR_NULL (str));
return FALSE;
}
}
static gboolean
gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
const char *stname;
GstStructure *structure;
gboolean res;
structure = gst_caps_get_structure (caps, 0);
stname = gst_structure_get_name (structure);
if (strcmp ("audio/x-gsm", stname))
goto invalid_type;
gst_rtp_base_payload_set_options (payload, "audio",
payload->pt != GST_RTP_PAYLOAD_GSM, "GSM", 8000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
/* ERRORS */
invalid_type:
{
GST_WARNING_OBJECT (payload, "invalid media type received");
return FALSE;
}
}
static gboolean
gst_rtp_h263p_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
GstCaps *peercaps;
gchar *encoding_name = NULL;
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
peercaps =
gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), NULL);
if (peercaps) {
GstCaps *intersect = gst_caps_intersect (peercaps,
gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload)));
gst_caps_unref (peercaps);
if (!gst_caps_is_empty (intersect)) {
GstStructure *s = gst_caps_get_structure (intersect, 0);
encoding_name = g_strdup (gst_structure_get_string (s, "encoding-name"));
}
gst_caps_unref (intersect);
}
if (!encoding_name)
encoding_name = g_strdup ("H263-1998");
gst_rtp_base_payload_set_options (payload, "video", TRUE,
(gchar *) encoding_name, 90000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
g_free (encoding_name);
return res;
}
static gboolean
gst_rtp_mpv_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gst_rtp_base_payload_set_options (payload, "video",
payload->pt != GST_RTP_PAYLOAD_MPV, "MPV", 90000);
return gst_rtp_base_payload_set_outcaps (payload, NULL);
}
static gboolean
gst_rtp_asf_pay_set_caps (GstRTPBasePayload * rtppay, GstCaps * caps)
{
/* FIXME change application for the actual content */
gst_rtp_base_payload_set_options (rtppay, "application", TRUE, "X-ASF-PF",
1000);
return TRUE;
}
static gboolean
gst_rtp_klv_pay_setcaps (GstRTPBasePayload * pay, GstCaps * caps)
{
/* FIXME: allow other clock rates */
gst_rtp_base_payload_set_options (pay, "application", TRUE, "SMPTE336M",
90000);
return gst_rtp_base_payload_set_outcaps (pay, NULL);
}
static void
gst_rtp_g723_pay_init (GstRTPG723Pay * pay)
{
GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
pay->adapter = gst_adapter_new ();
payload->pt = GST_RTP_PAYLOAD_G723;
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "G723", 8000);
}
static gboolean
gst_rtp_L16_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpL16Pay *rtpL16pay;
gboolean res;
gchar *params;
GstAudioInfo *info;
const GstRTPChannelOrder *order;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
rtpL16pay = GST_RTP_L16_PAY (basepayload);
info = &rtpL16pay->info;
gst_audio_info_init (info);
if (!gst_audio_info_from_caps (info, caps))
goto invalid_caps;
order = gst_rtp_channels_get_by_pos (info->channels, info->position);
rtpL16pay->order = order;
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L16",
info->rate);
params = g_strdup_printf ("%d", info->channels);
if (!order && info->channels > 2) {
GST_ELEMENT_WARNING (rtpL16pay, STREAM, DECODE,
(NULL), ("Unknown channel order for %d channels", info->channels));
}
if (order && order->name) {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
info->channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
info->channels, NULL);
}
g_free (params);
/* octet-per-sample is 2 * channels for L16 */
gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,
2 * info->channels);
return res;
/* ERRORS */
invalid_caps:
{
GST_DEBUG_OBJECT (rtpL16pay, "invalid caps");
return FALSE;
}
}
static gboolean
gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
gst_rtp_base_payload_set_options (payload, "audio",
payload->pt != GST_RTP_PAYLOAD_G723, "G723", 8000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
}
static gboolean
gst_rtp_mpa_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
gst_rtp_base_payload_set_options (payload, "audio",
payload->pt != GST_RTP_PAYLOAD_MPA, "MPA", 90000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
}
static gboolean
gst_dv_pay_negotiate (GstRTPDVPay * rtpdvpay, guint8 * data, gsize size)
{
const gchar *encode, *media;
gboolean audio_bundled, res;
if ((data[3] & 0x80) == 0) { /* DSF flag */
/* it's an NTSC format */
if ((data[80 * 5 + 48 + 3] & 0x4) && (data[80 * 5 + 48] == 0x60)) { /* 4:2:2 sampling */
/* NTSC 50Mbps */
encode = "314M-25/525-60";
} else { /* 4:1:1 sampling */
/* NTSC 25Mbps */
encode = "SD-VCR/525-60";
}
} else {
/* it's a PAL format */
if ((data[80 * 5 + 48 + 3] & 0x4) && (data[80 * 5 + 48] == 0x60)) { /* 4:2:2 sampling */
/* PAL 50Mbps */
encode = "314M-50/625-50";
} else if ((data[5] & 0x07) == 0) { /* APT flag */
/* PAL 25Mbps 4:2:0 */
encode = "SD-VCR/625-50";
} else
/* PAL 25Mbps 4:1:1 */
encode = "314M-25/625-50";
}
media = "video";
audio_bundled = FALSE;
switch (rtpdvpay->mode) {
case GST_DV_PAY_MODE_AUDIO:
media = "audio";
break;
case GST_DV_PAY_MODE_BUNDLED:
audio_bundled = TRUE;
break;
default:
break;
}
gst_rtp_base_payload_set_options (GST_RTP_BASE_PAYLOAD (rtpdvpay), media,
TRUE, "DV", 90000);
if (audio_bundled) {
res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpdvpay),
"encode", G_TYPE_STRING, encode,
"audio", G_TYPE_STRING, "bundled", NULL);
} else {
res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpdvpay),
"encode", G_TYPE_STRING, encode, NULL);
}
return res;
}
static gboolean
gst_rtp_pcmu_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
payload->pt = GST_RTP_PAYLOAD_PCMU;
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "PCMU", 8000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
}
static gboolean
gst_rtp_ac3_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
gint rate;
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "rate", &rate))
rate = 90000; /* default */
gst_rtp_base_payload_set_options (payload, "audio", TRUE, "AC3", rate);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
}
static gboolean
gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
gchar *capsstr;
capsstr = gst_caps_to_string (caps);
gst_rtp_base_payload_set_options (payload, "audio", FALSE,
"X-GST-OPUS-DRAFT-SPITTKA-00", 48000);
res =
gst_rtp_base_payload_set_outcaps (payload, "caps", G_TYPE_STRING, capsstr,
NULL);
g_free (capsstr);
return res;
}
static gboolean
gst_rtp_siren_pay_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps)
{
GstRTPSirenPay *rtpsirenpay;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
gint dct_length;
GstStructure *structure;
const char *payload_name;
rtpsirenpay = GST_RTP_SIREN_PAY (rtpbasepayload);
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "dct-length", &dct_length);
if (dct_length != 320)
goto wrong_dct;
payload_name = gst_structure_get_name (structure);
if (g_ascii_strcasecmp ("audio/x-siren", payload_name))
goto wrong_caps;
gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "SIREN",
16000);
/* set options for this frame based audio codec */
gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload, 20, 40);
return gst_rtp_base_payload_set_outcaps (rtpbasepayload, NULL);
/* ERRORS */
wrong_dct:
{
GST_ERROR_OBJECT (rtpsirenpay, "dct-length must be 320, received %d",
dct_length);
return FALSE;
}
wrong_caps:
{
GST_ERROR_OBJECT (rtpsirenpay, "expected audio/x-siren, received %s",
payload_name);
return FALSE;
}
}
static gboolean
gst_rtp_sbc_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
{
GstRtpSBCPay *sbcpay;
gint rate, subbands, channels, blocks, bitpool;
gint frame_len;
const gchar *channel_mode;
GstStructure *structure;
sbcpay = GST_RTP_SBC_PAY (payload);
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "rate", &rate))
return FALSE;
if (!gst_structure_get_int (structure, "channels", &channels))
return FALSE;
if (!gst_structure_get_int (structure, "blocks", &blocks))
return FALSE;
if (!gst_structure_get_int (structure, "bitpool", &bitpool))
return FALSE;
if (!gst_structure_get_int (structure, "subbands", &subbands))
return FALSE;
channel_mode = gst_structure_get_string (structure, "channel-mode");
if (!channel_mode)
return FALSE;
frame_len = gst_rtp_sbc_pay_get_frame_len (subbands, channels, blocks,
bitpool, channel_mode);
sbcpay->frame_length = frame_len;
gst_rtp_base_payload_set_options (payload, "audio", TRUE, "SBC", rate);
GST_DEBUG_OBJECT (payload, "calculated frame length: %d ", frame_len);
return gst_rtp_base_payload_set_outcaps (payload, NULL);
}
static gboolean
gst_rtp_ilbc_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload,
GstCaps * caps)
{
GstRTPILBCPay *rtpilbcpay;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
gboolean ret;
gint mode;
gchar *mode_str;
GstStructure *structure;
const char *payload_name;
rtpilbcpay = GST_RTP_ILBC_PAY (rtpbasepayload);
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
structure = gst_caps_get_structure (caps, 0);
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_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "ILBC",
8000);
/* set options for this frame based audio codec */
gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload,
mode, mode == 30 ? 50 : 38);
mode_str = g_strdup_printf ("%d", mode);
ret =
gst_rtp_base_payload_set_outcaps (rtpbasepayload, "mode", G_TYPE_STRING,
mode_str, NULL);
g_free (mode_str);
if (mode != rtpilbcpay->mode && rtpilbcpay->mode != -1)
goto mode_changed;
rtpilbcpay->mode = mode;
return ret;
/* ERRORS */
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! "
"Mode cannot change while streaming", rtpilbcpay->mode, mode);
return FALSE;
}
}
static gboolean
gst_rtp_mp4g_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
GstRtpMP4GPay *rtpmp4gpay;
GstStructure *structure;
const GValue *codec_data;
const gchar *media_type = NULL;
gboolean res;
rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
structure = gst_caps_get_structure (caps, 0);
codec_data = gst_structure_get_value (structure, "codec_data");
if (codec_data) {
GST_LOG_OBJECT (rtpmp4gpay, "got codec_data");
if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
GstBuffer *buffer;
const gchar *name;
buffer = gst_value_get_buffer (codec_data);
GST_LOG_OBJECT (rtpmp4gpay, "configuring codec_data");
name = gst_structure_get_name (structure);
/* parse buffer */
if (!strcmp (name, "audio/mpeg")) {
res = gst_rtp_mp4g_pay_parse_audio_config (rtpmp4gpay, buffer);
media_type = "audio";
} else if (!strcmp (name, "video/mpeg")) {
res = gst_rtp_mp4g_pay_parse_video_config (rtpmp4gpay, buffer);
media_type = "video";
} else {
res = FALSE;
}
if (!res)
goto config_failed;
/* now we can configure the buffer */
if (rtpmp4gpay->config)
gst_buffer_unref (rtpmp4gpay->config);
rtpmp4gpay->config = gst_buffer_copy (buffer);
}
}
if (media_type == NULL)
goto config_failed;
gst_rtp_base_payload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC",
rtpmp4gpay->rate);
res = gst_rtp_mp4g_pay_new_caps (rtpmp4gpay);
return res;
/* ERRORS */
config_failed:
{
GST_DEBUG_OBJECT (rtpmp4gpay, "failed to parse config");
return FALSE;
}
}
static gboolean
gst_rtp_vraw_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
GstRtpVRawPay *rtpvrawpay;
gboolean res;
gint pgroup, xinc, yinc;
const gchar *depthstr, *samplingstr, *colorimetrystr;
gchar *wstr, *hstr;
GstVideoInfo info;
rtpvrawpay = GST_RTP_VRAW_PAY (payload);
if (!gst_video_info_from_caps (&info, caps))
goto invalid_caps;
rtpvrawpay->vinfo = info;
if (gst_video_colorimetry_matches (&info.colorimetry,
GST_VIDEO_COLORIMETRY_BT601)) {
colorimetrystr = "BT601-5";
} else if (gst_video_colorimetry_matches (&info.colorimetry,
GST_VIDEO_COLORIMETRY_BT709)) {
colorimetrystr = "BT709-2";
} else if (gst_video_colorimetry_matches (&info.colorimetry,
GST_VIDEO_COLORIMETRY_SMPTE240M)) {
colorimetrystr = "SMPTE240M";
} else {
colorimetrystr = "SMPTE240M";
}
xinc = yinc = 1;
/* these values are the only thing we can do */
depthstr = "8";
switch (GST_VIDEO_INFO_FORMAT (&info)) {
case GST_VIDEO_FORMAT_RGBA:
samplingstr = "RGBA";
pgroup = 4;
break;
case GST_VIDEO_FORMAT_BGRA:
samplingstr = "BGRA";
pgroup = 4;
break;
case GST_VIDEO_FORMAT_RGB:
samplingstr = "RGB";
pgroup = 3;
break;
case GST_VIDEO_FORMAT_BGR:
samplingstr = "BGR";
pgroup = 3;
break;
case GST_VIDEO_FORMAT_AYUV:
samplingstr = "YCbCr-4:4:4";
pgroup = 3;
break;
case GST_VIDEO_FORMAT_UYVY:
samplingstr = "YCbCr-4:2:2";
pgroup = 4;
xinc = 2;
break;
case GST_VIDEO_FORMAT_Y41B:
samplingstr = "YCbCr-4:1:1";
pgroup = 6;
xinc = 4;
break;
case GST_VIDEO_FORMAT_I420:
samplingstr = "YCbCr-4:2:0";
pgroup = 6;
xinc = yinc = 2;
break;
case GST_VIDEO_FORMAT_UYVP:
samplingstr = "YCbCr-4:2:2";
pgroup = 5;
xinc = 2;
depthstr = "10";
break;
default:
goto unknown_format;
break;
}
if (GST_VIDEO_INFO_IS_INTERLACED (&info)) {
yinc *= 2;
}
rtpvrawpay->pgroup = pgroup;
rtpvrawpay->xinc = xinc;
rtpvrawpay->yinc = yinc;
GST_DEBUG_OBJECT (payload, "width %d, height %d, sampling %s",
GST_VIDEO_INFO_WIDTH (&info), GST_VIDEO_INFO_HEIGHT (&info), samplingstr);
GST_DEBUG_OBJECT (payload, "xinc %d, yinc %d, pgroup %d", xinc, yinc, pgroup);
wstr = g_strdup_printf ("%d", GST_VIDEO_INFO_WIDTH (&info));
hstr = g_strdup_printf ("%d", GST_VIDEO_INFO_HEIGHT (&info));
gst_rtp_base_payload_set_options (payload, "video", TRUE, "RAW", 90000);
if (GST_VIDEO_INFO_IS_INTERLACED (&info)) {
res = gst_rtp_base_payload_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, "interlace", G_TYPE_STRING, "true", NULL);
} else {
res = gst_rtp_base_payload_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 res;
/* ERRORS */
invalid_caps:
{
GST_ERROR_OBJECT (payload, "could not parse caps");
return FALSE;
}
unknown_format:
{
GST_ERROR_OBJECT (payload, "unknown caps format");
return FALSE;
}
}
static gboolean
gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
GstCaps *src_caps;
GstStructure *s;
char *encoding_name;
gint channels, rate;
const char *sprop_stereo = NULL;
char *sprop_maxcapturerate = NULL;
src_caps = gst_pad_get_allowed_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
if (src_caps) {
src_caps = gst_caps_make_writable (src_caps);
src_caps = gst_caps_truncate (src_caps);
s = gst_caps_get_structure (src_caps, 0);
gst_structure_fixate_field_string (s, "encoding-name", "OPUS");
encoding_name = g_strdup (gst_structure_get_string (s, "encoding-name"));
gst_caps_unref (src_caps);
} else {
encoding_name = g_strdup ("X-GST-OPUS-DRAFT-SPITTKA-00");
}
s = gst_caps_get_structure (caps, 0);
if (gst_structure_get_int (s, "channels", &channels)) {
if (channels > 2) {
GST_ERROR_OBJECT (payload,
"More than 2 channels with channel-mapping-family=0 is invalid");
return FALSE;
} else if (channels == 2) {
sprop_stereo = "1";
} else {
sprop_stereo = "0";
}
}
if (gst_structure_get_int (s, "rate", &rate)) {
sprop_maxcapturerate = g_strdup_printf ("%d", rate);
}
gst_rtp_base_payload_set_options (payload, "audio", FALSE,
encoding_name, 48000);
g_free (encoding_name);
if (sprop_maxcapturerate && sprop_stereo) {
res =
gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",
G_TYPE_STRING, sprop_maxcapturerate, "sprop-stereo", G_TYPE_STRING,
sprop_stereo, NULL);
} else if (sprop_maxcapturerate) {
res =
gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",
G_TYPE_STRING, sprop_maxcapturerate, NULL);
} else if (sprop_stereo) {
res =
gst_rtp_base_payload_set_outcaps (payload, "sprop-stereo",
G_TYPE_STRING, sprop_stereo, NULL);
} else {
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
}
g_free (sprop_maxcapturerate);
return res;
}
static gboolean
gst_rtp_celt_pay_parse_ident (GstRtpCELTPay * rtpceltpay,
const guint8 * data, guint size)
{
guint32 version, header_size, rate, nb_channels, frame_size, overlap;
guint32 bytes_per_packet;
GstRTPBasePayload *payload;
gchar *cstr, *fsstr;
gboolean res;
/* we need the header string (8), the version string (20), the version
* and the header length. */
if (size < 36)
goto too_small;
if (!g_str_has_prefix ((const gchar *) data, "CELT "))
goto wrong_header;
/* skip header and version string */
data += 28;
version = GST_READ_UINT32_LE (data);
GST_DEBUG_OBJECT (rtpceltpay, "version %08x", version);
#if 0
if (version != 1)
goto wrong_version;
#endif
data += 4;
/* ensure sizes */
header_size = GST_READ_UINT32_LE (data);
if (header_size < 56)
goto header_too_small;
if (size < header_size)
goto payload_too_small;
data += 4;
rate = GST_READ_UINT32_LE (data);
data += 4;
nb_channels = GST_READ_UINT32_LE (data);
data += 4;
frame_size = GST_READ_UINT32_LE (data);
data += 4;
overlap = GST_READ_UINT32_LE (data);
data += 4;
bytes_per_packet = GST_READ_UINT32_LE (data);
GST_DEBUG_OBJECT (rtpceltpay, "rate %d, nb_channels %d, frame_size %d",
rate, nb_channels, frame_size);
GST_DEBUG_OBJECT (rtpceltpay, "overlap %d, bytes_per_packet %d",
overlap, bytes_per_packet);
payload = GST_RTP_BASE_PAYLOAD (rtpceltpay);
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "CELT", rate);
cstr = g_strdup_printf ("%d", nb_channels);
fsstr = g_strdup_printf ("%d", frame_size);
res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params",
G_TYPE_STRING, cstr, "frame-size", G_TYPE_STRING, fsstr, NULL);
g_free (cstr);
g_free (fsstr);
return res;
/* ERRORS */
too_small:
{
GST_DEBUG_OBJECT (rtpceltpay,
"ident packet too small, need at least 32 bytes");
return FALSE;
}
wrong_header:
{
GST_DEBUG_OBJECT (rtpceltpay,
"ident packet does not start with \"CELT \"");
return FALSE;
}
#if 0
wrong_version:
{
GST_DEBUG_OBJECT (rtpceltpay, "can only handle version 1, have version %d",
version);
return FALSE;
}
#endif
header_too_small:
{
GST_DEBUG_OBJECT (rtpceltpay,
"header size too small, need at least 80 bytes, " "got only %d",
header_size);
return FALSE;
}
payload_too_small:
{
GST_DEBUG_OBJECT (rtpceltpay,
"payload too small, need at least %d bytes, got only %d", header_size,
size);
return FALSE;
}
}
static gboolean
gst_rtp_bv_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps)
{
GstRTPBVPay *rtpbvpay;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
gint mode;
GstStructure *structure;
const char *payload_name;
rtpbvpay = GST_RTP_BV_PAY (rtpbasepayload);
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
structure = gst_caps_get_structure (caps, 0);
payload_name = gst_structure_get_name (structure);
if (g_ascii_strcasecmp ("audio/x-bv", payload_name))
goto wrong_caps;
if (!gst_structure_get_int (structure, "mode", &mode))
goto no_mode;
if (mode != 16 && mode != 32)
goto wrong_mode;
if (mode == 16) {
gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV16",
8000);
rtpbasepayload->clock_rate = 8000;
} else {
gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV32",
16000);
rtpbasepayload->clock_rate = 16000;
}
/* set options for this frame based audio codec */
gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload,
mode, mode == 16 ? 10 : 20);
if (mode != rtpbvpay->mode && rtpbvpay->mode != -1)
goto mode_changed;
rtpbvpay->mode = mode;
return TRUE;
/* ERRORS */
wrong_caps:
{
GST_ERROR_OBJECT (rtpbvpay, "expected audio/x-bv, received %s",
payload_name);
return FALSE;
}
no_mode:
{
GST_ERROR_OBJECT (rtpbvpay, "did not receive a mode");
return FALSE;
}
wrong_mode:
{
GST_ERROR_OBJECT (rtpbvpay, "mode must be 16 or 32, received %d", mode);
return FALSE;
}
mode_changed:
{
GST_ERROR_OBJECT (rtpbvpay, "Mode has changed from %d to %d! "
"Mode cannot change while streaming", rtpbvpay->mode, mode);
return FALSE;
}
}
static gboolean
gst_rtp_mpv_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gst_rtp_base_payload_set_options (payload, "video", FALSE, "MPV", 90000);
return gst_rtp_base_payload_set_outcaps (payload, NULL);
}
static gboolean
gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpG722Pay *rtpg722pay;
GstStructure *structure;
gint rate, channels, clock_rate;
gboolean res;
gchar *params;
#if 0
GstAudioChannelPosition *pos;
const GstRTPChannelOrder *order;
#endif
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
rtpg722pay = GST_RTP_G722_PAY (basepayload);
structure = gst_caps_get_structure (caps, 0);
/* first parse input caps */
if (!gst_structure_get_int (structure, "rate", &rate))
goto no_rate;
if (!gst_structure_get_int (structure, "channels", &channels))
goto no_channels;
/* FIXME: Do something with the channel positions */
#if 0
/* get the channel order */
pos = gst_audio_get_channel_positions (structure);
if (pos)
order = gst_rtp_channels_get_by_pos (channels, pos);
else
order = NULL;
#endif
/* Clock rate is always 8000 Hz for G722 according to
* RFC 3551 although the sampling rate is 16000 Hz */
clock_rate = 8000;
gst_rtp_base_payload_set_options (basepayload, "audio",
basepayload->pt != GST_RTP_PAYLOAD_G722, "G722", clock_rate);
params = g_strdup_printf ("%d", channels);
#if 0
if (!order && channels > 2) {
GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
(NULL), ("Unknown channel order for %d channels", channels));
}
if (order && order->name) {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
#endif
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, NULL);
#if 0
}
#endif
g_free (params);
#if 0
g_free (pos);
#endif
rtpg722pay->rate = rate;
rtpg722pay->channels = channels;
/* bits-per-sample is 4 * channels for G722, but as the RTP clock runs at
* half speed (8 instead of 16 khz), pretend it's 8 bits per sample
* channels. */
gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
8 * rtpg722pay->channels);
return res;
/* ERRORS */
no_rate:
{
GST_DEBUG_OBJECT (rtpg722pay, "no rate given");
return FALSE;
}
no_channels:
{
GST_DEBUG_OBJECT (rtpg722pay, "no channels given");
return FALSE;
}
}
