static const GstElementDetails gst_rtp_amrdepay_details =
GST_ELEMENT_DETAILS ("RTP packet depayloader",
"Codec/Depayloader/Network",
- "Extracts AMR audio from RTP packets (RFC 3267)",
+ "Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)",
"Wim Taymans <wim@fluendo.com>");
/* RtpAMRDepay signals and args */
* params see RFC 3267, section 8.1
*/
static GstStaticPadTemplate gst_rtp_amr_depay_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
* GstCaps mapping. */
"octet-align = (string) \"1\", "
"crc = (string) { \"0\", \"1\" }, "
+ "robust-sorting = (string) \"0\", " "interleaving = (string) \"0\";"
+ /* following options are not needed for a decoder
+ *
+ "mode-set = (int) [ 0, 7 ], "
+ "mode-change-period = (int) [ 1, MAX ], "
+ "mode-change-neighbor = (boolean) { TRUE, FALSE }, "
+ "maxptime = (int) [ 20, MAX ], "
+ "ptime = (int) [ 20, MAX ]"
+ */
+ "application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) 16000, "
+ "encoding-name = (string) \"AMR-WB\", "
+ "encoding-params = (string) \"1\", "
+ /* NOTE that all values must be strings in orde to be able to do SDP <->
+ * GstCaps mapping. */
+ "octet-align = (string) \"1\", "
+ "crc = (string) { \"0\", \"1\" }, "
"robust-sorting = (string) \"0\", " "interleaving = (string) \"0\""
/* following options are not needed for a decoder
*
);
static GstStaticPadTemplate gst_rtp_amr_depay_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
+ GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000")
+ GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000;"
+ "audio/AMR-WB, " "channels = (int) 1," "rate = (int) 16000")
);
static gboolean gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps *srccaps;
GstRtpAMRDepay *rtpamrdepay;
const gchar *params;
- const gchar *str;
- gint clock_rate = 8000; /* default */
+ const gchar *str, *type;
+ gint clock_rate, need_clock_rate;
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
+ /* figure out the mode first and set the clock rates */
+ 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;
+ type = "audio/AMR";
+ } else if (strcmp (str, "AMR-WB") == 0) {
+ rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB;
+ clock_rate = need_clock_rate = 16000;
+ type = "audio/AMR-WB";
+ } else
+ goto invalid_mode;
+ } else
+ goto invalid_mode;
+
if (!(str = gst_structure_get_string (structure, "octet-align")))
rtpamrdepay->octet_align = FALSE;
else
* no robust sorting, no interleaving for now */
if (rtpamrdepay->channels != 1)
return FALSE;
- if (clock_rate != 8000)
+ if (clock_rate != need_clock_rate)
return FALSE;
if (rtpamrdepay->octet_align != TRUE)
return FALSE;
if (rtpamrdepay->interleaving != FALSE)
return FALSE;
- srccaps = gst_caps_new_simple ("audio/AMR",
+ 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);
gst_caps_unref (srccaps);
rtpamrdepay->negotiated = TRUE;
return TRUE;
+
+ /* ERRORS */
+invalid_mode:
+ {
+ GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name");
+ return FALSE;
+ }
}
/* -1 is invalid */
-static gint frame_size[16] = {
+static gint nb_frame_size[16] = {
12, 13, 15, 17, 19, 20, 26, 31,
5, -1, -1, -1, -1, -1, -1, 0
};
+static gint wb_frame_size[16] = {
+ 17, 23, 32, 36, 40, 46, 50, 58,
+ 60, -1, -1, -1, -1, -1, -1, 0
+};
static GstBuffer *
gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
GstRtpAMRDepay *rtpamrdepay;
GstBuffer *outbuf = NULL;
gint payload_len;
+ gint *frame_size;
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
if (!gst_rtp_buffer_validate (buf))
goto invalid_packet;
- /* when we get here, 1 channel, 8000 Hz, octet aligned, no CRC,
+ /* setup frame size pointer */
+ if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB)
+ frame_size = nb_frame_size;
+ else
+ frame_size = wb_frame_size;
+
+ /* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC,
* no robust sorting, no interleaving data is to be depayloaded */
{
guint8 *payload, *p, *dp;
* RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
* Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive
* Multi-Rate Wideband (AMR-WB) Audio Codecs.
+ *
+ * ETSI TS 126 201 V6.0.0 (2004-12) - Digital cellular telecommunications system (Phase 2+);
+ * Universal Mobile Telecommunications System (UMTS);
+ * AMR speech codec, wideband;
+ * Frame structure
+ * (3GPP TS 26.201 version 6.0.0 Release 6)
*/
/* elementfactory information */
static const GstElementDetails gst_rtp_amrpay_details =
GST_ELEMENT_DETAILS ("RTP packet payloader",
"Codec/Payloader/Network",
- "Payload-encode AMR audio into RTP packets (RFC 3267)",
+ "Payload-encode AMR or AMR-WB audio into RTP packets (RFC 3267)",
"Wim Taymans <wim@fluendo.com>");
static GstStaticPadTemplate gst_rtp_amr_pay_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/AMR, channels=(int)1, rate=(int)8000")
+ GST_STATIC_CAPS ("audio/AMR, channels=(int)1, rate=(int)8000; "
+ "audio/AMR-WB, channels=(int)1, rate=(int)16000")
);
static GstStaticPadTemplate gst_rtp_amr_pay_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
+ GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) [ 96, 127 ], "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"AMR\", "
"encoding-params = (string) \"1\", "
"mode-set = (int) [ 0, 7 ], "
"mode-change-period = (int) [ 1, MAX ], "
"mode-change-neighbor = (string) { \"0\", \"1\" }, "
+ "maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ];"
+ "application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) 16000, "
+ "encoding-name = (string) \"AMR-WB\", "
+ "encoding-params = (string) \"1\", "
+ "octet-align = (string) \"1\", "
+ "crc = (string) \"0\", "
+ "robust-sorting = (string) \"0\", "
+ "interleaving = (string) \"0\", "
+ "mode-set = (int) [ 0, 7 ], "
+ "mode-change-period = (int) [ 1, MAX ], "
+ "mode-change-neighbor = (string) { \"0\", \"1\" }, "
"maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ]")
);
-static void gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * klass);
-static void gst_rtp_amr_pay_base_init (GstRtpAMRPayClass * klass);
-static void gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay);
-
static gboolean gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * pad,
GstBuffer * buffer);
-static GstBaseRTPPayloadClass *parent_class = NULL;
-
-static GType
-gst_rtp_amr_pay_get_type (void)
-{
- static GType rtpamrpay_type = 0;
-
- if (!rtpamrpay_type) {
- static const GTypeInfo rtpamrpay_info = {
- sizeof (GstRtpAMRPayClass),
- (GBaseInitFunc) gst_rtp_amr_pay_base_init,
- NULL,
- (GClassInitFunc) gst_rtp_amr_pay_class_init,
- NULL,
- NULL,
- sizeof (GstRtpAMRPay),
- 0,
- (GInstanceInitFunc) gst_rtp_amr_pay_init,
- };
-
- rtpamrpay_type =
- g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpAMRPay",
- &rtpamrpay_info, 0);
- }
- return rtpamrpay_type;
-}
+GST_BOILERPLATE (GstRtpAMRPay, gst_rtp_amr_pay, GstBaseRTPPayload,
+ GST_TYPE_BASE_RTP_PAYLOAD);
static void
-gst_rtp_amr_pay_base_init (GstRtpAMRPayClass * klass)
+gst_rtp_amr_pay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gstbasertppayload_class->handle_buffer = gst_rtp_amr_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpamrpay_debug, "rtpamrpay", 0,
- "AMR RTP Payloader");
-
+ "AMR/AMR-WB RTP Payloader");
}
static void
-gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay)
+gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay, GstRtpAMRPayClass * klass)
{
}
gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpAMRPay *rtpamrpay;
+ const GstStructure *s;
+ const gchar *str;
rtpamrpay = GST_RTP_AMR_PAY (basepayload);
- gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
+ /* 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_basertppayload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
+ else
+ gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR-WB",
+ 16000);
+
gst_basertppayload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, "1", "octet-align", G_TYPE_STRING, "1",
/* don't set the defaults
NULL);
return TRUE;
+
+ /* ERRORS */
+wrong_type:
+ {
+ GST_ERROR_OBJECT (rtpamrpay, "unsupported media type '%s'",
+ GST_STR_NULL (str));
+ return FALSE;
+ }
}
/* -1 is invalid */
-static gint frame_size[16] = {
+static gint nb_frame_size[16] = {
12, 13, 15, 17, 19, 20, 26, 31,
5, -1, -1, -1, -1, -1, -1, 0
};
+static gint wb_frame_size[16] = {
+ 17, 23, 32, 36, 40, 46, 50, 58,
+ 60, -1, -1, -1, -1, -1, -1, 0
+};
static GstFlowReturn
gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * basepayload,
guint packet_len, mtu;
gint i, num_packets, num_nonempty_packets;
gint amr_len;
+ gint *frame_size;
rtpamrpay = GST_RTP_AMR_PAY (basepayload);
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpamrpay);
data = GST_BUFFER_DATA (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
- /* FIXME, only
- * octet aligned, no interleaving, single channel, no CRC,
- * no robust-sorting. */
+ /* setup frame size pointer */
+ if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
+ frame_size = nb_frame_size;
+ else
+ frame_size = wb_frame_size;
GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
+ /* FIXME, only
+ * octet aligned, no interleaving, single channel, no CRC,
+ * no robust-sorting. To fix this you need to implement the downstream
+ * negotiation function. */
+
/* first count number of packets and total amr frame size */
amr_len = num_packets = num_nonempty_packets = 0;
for (i = 0; i < size; i++) {
projects / platform / upstream / gst-plugins-good.git / commitdiff