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gst/rtp/: Add support for AMR-WB.
Original commit message from CVS: Based on Patch by: Daniel Charles <dcharles at ti dot com> * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process): * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_base_init), (gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_init), (gst_rtp_amr_pay_setcaps), (gst_rtp_amr_pay_handle_buffer): * gst/rtp/gstrtpamrpay.h: Add support for AMR-WB. Small cleanups such as using BOILERPLATE.
This commit is contained in:
Daniel Charles
Wim Taymans
parent
0b2e6f1c90
commit
89ae9b40f9
5 changed files with 168 additions and 54 deletions
14
ChangeLog
14
ChangeLog
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@ -1,3 +1,17 @@
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2007-06-01 Wim Taymans <wim@fluendo.com>
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Based on Patch by: Daniel Charles <dcharles at ti dot com>
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* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
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(gst_rtp_amr_depay_process):
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* gst/rtp/gstrtpamrdepay.h:
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* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_base_init),
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(gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_init),
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(gst_rtp_amr_pay_setcaps), (gst_rtp_amr_pay_handle_buffer):
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* gst/rtp/gstrtpamrpay.h:
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Add support for AMR-WB.
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Small cleanups such as using BOILERPLATE.
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2007-05-31 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream):
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@ -38,7 +38,7 @@
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static const GstElementDetails gst_rtp_amrdepay_details =
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GST_ELEMENT_DETAILS ("RTP packet depayloader",
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"Codec/Depayloader/Network",
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"Extracts AMR audio from RTP packets (RFC 3267)",
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"Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)",
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"Wim Taymans <wim@fluendo.com>");
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/* RtpAMRDepay signals and args */
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@ -58,7 +58,7 @@ enum
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* params see RFC 3267, section 8.1
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*/
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static GstStaticPadTemplate gst_rtp_amr_depay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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@ -71,6 +71,25 @@ GST_STATIC_PAD_TEMPLATE ("sink",
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* GstCaps mapping. */
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"octet-align = (string) \"1\", "
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"crc = (string) { \"0\", \"1\" }, "
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"robust-sorting = (string) \"0\", " "interleaving = (string) \"0\";"
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/* following options are not needed for a decoder
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*
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"mode-set = (int) [ 0, 7 ], "
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"mode-change-period = (int) [ 1, MAX ], "
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"mode-change-neighbor = (boolean) { TRUE, FALSE }, "
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"maxptime = (int) [ 20, MAX ], "
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"ptime = (int) [ 20, MAX ]"
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*/
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 16000, "
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"encoding-name = (string) \"AMR-WB\", "
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"encoding-params = (string) \"1\", "
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/* NOTE that all values must be strings in orde to be able to do SDP <->
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* GstCaps mapping. */
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"octet-align = (string) \"1\", "
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"crc = (string) { \"0\", \"1\" }, "
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"robust-sorting = (string) \"0\", " "interleaving = (string) \"0\""
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/* following options are not needed for a decoder
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*
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@ -84,10 +103,11 @@ GST_STATIC_PAD_TEMPLATE ("sink",
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);
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static GstStaticPadTemplate gst_rtp_amr_depay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000")
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GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000;"
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"audio/AMR-WB, " "channels = (int) 1," "rate = (int) 16000")
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);
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static gboolean gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload,
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@ -146,13 +166,28 @@ gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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GstCaps *srccaps;
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GstRtpAMRDepay *rtpamrdepay;
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const gchar *params;
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const gchar *str;
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gint clock_rate = 8000; /* default */
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const gchar *str, *type;
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gint clock_rate, need_clock_rate;
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rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
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structure = gst_caps_get_structure (caps, 0);
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/* figure out the mode first and set the clock rates */
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if ((str = gst_structure_get_string (structure, "encoding-name"))) {
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if (strcmp (str, "AMR") == 0) {
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rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_NB;
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clock_rate = need_clock_rate = 8000;
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type = "audio/AMR";
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} else if (strcmp (str, "AMR-WB") == 0) {
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rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB;
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clock_rate = need_clock_rate = 16000;
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type = "audio/AMR-WB";
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} else
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goto invalid_mode;
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} else
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goto invalid_mode;
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if (!(str = gst_structure_get_string (structure, "octet-align")))
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rtpamrdepay->octet_align = FALSE;
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else
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@ -201,7 +236,7 @@ gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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* no robust sorting, no interleaving for now */
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if (rtpamrdepay->channels != 1)
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return FALSE;
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if (clock_rate != 8000)
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if (clock_rate != need_clock_rate)
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return FALSE;
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if (rtpamrdepay->octet_align != TRUE)
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return FALSE;
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@ -210,22 +245,34 @@ gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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if (rtpamrdepay->interleaving != FALSE)
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return FALSE;
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srccaps = gst_caps_new_simple ("audio/AMR",
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srccaps = gst_caps_new_simple (type,
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"channels", G_TYPE_INT, rtpamrdepay->channels,
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"rate", G_TYPE_INT, clock_rate, NULL);
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gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
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gst_caps_unref (srccaps);
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rtpamrdepay->negotiated = TRUE;
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return TRUE;
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/* ERRORS */
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invalid_mode:
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{
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GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name");
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return FALSE;
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}
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}
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/* -1 is invalid */
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static gint frame_size[16] = {
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static gint nb_frame_size[16] = {
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12, 13, 15, 17, 19, 20, 26, 31,
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5, -1, -1, -1, -1, -1, -1, 0
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};
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static gint wb_frame_size[16] = {
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17, 23, 32, 36, 40, 46, 50, 58,
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60, -1, -1, -1, -1, -1, -1, 0
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};
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static GstBuffer *
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gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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@ -233,6 +280,7 @@ gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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GstRtpAMRDepay *rtpamrdepay;
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GstBuffer *outbuf = NULL;
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gint payload_len;
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gint *frame_size;
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rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
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@ -242,7 +290,13 @@ gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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if (!gst_rtp_buffer_validate (buf))
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goto invalid_packet;
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/* when we get here, 1 channel, 8000 Hz, octet aligned, no CRC,
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/* setup frame size pointer */
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if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB)
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frame_size = nb_frame_size;
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else
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frame_size = wb_frame_size;
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/* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC,
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* no robust sorting, no interleaving data is to be depayloaded */
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{
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guint8 *payload, *p, *dp;
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@ -39,11 +39,18 @@ G_BEGIN_DECLS
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typedef struct _GstRtpAMRDepay GstRtpAMRDepay;
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typedef struct _GstRtpAMRDepayClass GstRtpAMRDepayClass;
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typedef enum {
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GST_RTP_AMR_DP_MODE_INVALID = 0,
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GST_RTP_AMR_DP_MODE_NB = 1,
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GST_RTP_AMR_DP_MODE_WB = 2
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} GstRtpAMRDepayMode;
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struct _GstRtpAMRDepay
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{
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GstBaseRTPDepayload depayload;
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gboolean negotiated;
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GstRtpAMRDepayMode mode;
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gboolean octet_align;
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guint8 mode_set;
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@ -35,29 +35,36 @@ GST_DEBUG_CATEGORY_STATIC (rtpamrpay_debug);
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* RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
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* Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive
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* Multi-Rate Wideband (AMR-WB) Audio Codecs.
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*
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* ETSI TS 126 201 V6.0.0 (2004-12) - Digital cellular telecommunications system (Phase 2+);
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* Universal Mobile Telecommunications System (UMTS);
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* AMR speech codec, wideband;
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* Frame structure
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* (3GPP TS 26.201 version 6.0.0 Release 6)
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*/
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/* elementfactory information */
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static const GstElementDetails gst_rtp_amrpay_details =
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GST_ELEMENT_DETAILS ("RTP packet payloader",
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"Codec/Payloader/Network",
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"Payload-encode AMR audio into RTP packets (RFC 3267)",
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"Payload-encode AMR or AMR-WB audio into RTP packets (RFC 3267)",
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"Wim Taymans <wim@fluendo.com>");
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static GstStaticPadTemplate gst_rtp_amr_pay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/AMR, channels=(int)1, rate=(int)8000")
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GST_STATIC_CAPS ("audio/AMR, channels=(int)1, rate=(int)8000; "
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"audio/AMR-WB, channels=(int)1, rate=(int)16000")
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);
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static GstStaticPadTemplate gst_rtp_amr_pay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) [ 96, 127 ], "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) \"AMR\", "
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"encoding-params = (string) \"1\", "
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@ -68,47 +75,33 @@ GST_STATIC_PAD_TEMPLATE ("src",
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"mode-set = (int) [ 0, 7 ], "
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"mode-change-period = (int) [ 1, MAX ], "
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"mode-change-neighbor = (string) { \"0\", \"1\" }, "
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"maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ];"
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 16000, "
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"encoding-name = (string) \"AMR-WB\", "
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"encoding-params = (string) \"1\", "
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"octet-align = (string) \"1\", "
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"crc = (string) \"0\", "
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"robust-sorting = (string) \"0\", "
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"interleaving = (string) \"0\", "
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"mode-set = (int) [ 0, 7 ], "
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"mode-change-period = (int) [ 1, MAX ], "
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"mode-change-neighbor = (string) { \"0\", \"1\" }, "
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"maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ]")
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);
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static void gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * klass);
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static void gst_rtp_amr_pay_base_init (GstRtpAMRPayClass * klass);
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static void gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay);
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static gboolean gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * pad,
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GstBuffer * buffer);
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static GstBaseRTPPayloadClass *parent_class = NULL;
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static GType
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gst_rtp_amr_pay_get_type (void)
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{
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static GType rtpamrpay_type = 0;
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if (!rtpamrpay_type) {
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static const GTypeInfo rtpamrpay_info = {
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sizeof (GstRtpAMRPayClass),
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(GBaseInitFunc) gst_rtp_amr_pay_base_init,
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NULL,
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(GClassInitFunc) gst_rtp_amr_pay_class_init,
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NULL,
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NULL,
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sizeof (GstRtpAMRPay),
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0,
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(GInstanceInitFunc) gst_rtp_amr_pay_init,
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};
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rtpamrpay_type =
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g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpAMRPay",
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&rtpamrpay_info, 0);
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}
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return rtpamrpay_type;
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}
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GST_BOILERPLATE (GstRtpAMRPay, gst_rtp_amr_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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static void
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gst_rtp_amr_pay_base_init (GstRtpAMRPayClass * klass)
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gst_rtp_amr_pay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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@ -137,12 +130,11 @@ gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * klass)
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gstbasertppayload_class->handle_buffer = gst_rtp_amr_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtpamrpay_debug, "rtpamrpay", 0,
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"AMR RTP Payloader");
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"AMR/AMR-WB RTP Payloader");
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}
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static void
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gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay)
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gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay, GstRtpAMRPayClass * klass)
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{
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}
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@ -150,10 +142,29 @@ static gboolean
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gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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{
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GstRtpAMRPay *rtpamrpay;
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const GstStructure *s;
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const gchar *str;
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rtpamrpay = GST_RTP_AMR_PAY (basepayload);
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/* figure out the mode Narrow or Wideband */
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s = gst_caps_get_structure (caps, 0);
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if ((str = gst_structure_get_name (s))) {
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if (strcmp (str, "audio/AMR") == 0)
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rtpamrpay->mode = GST_RTP_AMR_P_MODE_NB;
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else if (strcmp (str, "audio/AMR-WB") == 0)
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rtpamrpay->mode = GST_RTP_AMR_P_MODE_WB;
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else
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goto wrong_type;
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} else
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goto wrong_type;
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if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
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gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
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else
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gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR-WB",
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16000);
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gst_basertppayload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, "1", "octet-align", G_TYPE_STRING, "1",
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/* don't set the defaults
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|
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@ -165,13 +176,25 @@ gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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NULL);
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return TRUE;
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/* ERRORS */
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wrong_type:
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{
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GST_ERROR_OBJECT (rtpamrpay, "unsupported media type '%s'",
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GST_STR_NULL (str));
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return FALSE;
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}
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}
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/* -1 is invalid */
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static gint frame_size[16] = {
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static gint nb_frame_size[16] = {
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12, 13, 15, 17, 19, 20, 26, 31,
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5, -1, -1, -1, -1, -1, -1, 0
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};
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static gint wb_frame_size[16] = {
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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,
|
||||
|
|
@ -186,6 +209,7 @@ 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);
|
||||
|
|
@ -194,12 +218,19 @@ gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * basepayload,
|
|||
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++) {
|
||||
|
|
|
|||
|
|
@ -40,9 +40,17 @@ G_BEGIN_DECLS
|
|||
typedef struct _GstRtpAMRPay GstRtpAMRPay;
|
||||
typedef struct _GstRtpAMRPayClass GstRtpAMRPayClass;
|
||||
|
||||
typedef enum {
|
||||
GST_RTP_AMR_P_MODE_INVALID = 0,
|
||||
GST_RTP_AMR_P_MODE_NB = 1,
|
||||
GST_RTP_AMR_P_MODE_WB = 2
|
||||
} GstRtpAMRPayMode;
|
||||
|
||||
struct _GstRtpAMRPay
|
||||
{
|
||||
GstBaseRTPPayload payload;
|
||||
|
||||
GstRtpAMRPayMode mode;
|
||||
};
|
||||
|
||||
struct _GstRtpAMRPayClass
|
||||
|
|
|
|||
Loading…
Reference in a new issue