diff options
Diffstat (limited to 'gst/rtp/gstrtpspeexpay.c')
-rw-r--r-- | gst/rtp/gstrtpspeexpay.c | 164 |
1 files changed, 158 insertions, 6 deletions
diff --git a/gst/rtp/gstrtpspeexpay.c b/gst/rtp/gstrtpspeexpay.c index a102ba7d..2ca72408 100644 --- a/gst/rtp/gstrtpspeexpay.c +++ b/gst/rtp/gstrtpspeexpay.c @@ -27,6 +27,9 @@ #include "gstrtpspeexpay.h" +GST_DEBUG_CATEGORY_STATIC (rtpspeexpay_debug); +#define GST_CAT_DEFAULT (rtpspeexpay_debug) + /* elementfactory information */ static const GstElementDetails gst_rtp_speex_pay_details = GST_ELEMENT_DETAILS ("RTP packet payloader", @@ -48,11 +51,14 @@ GST_STATIC_PAD_TEMPLATE ("src", GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " - "clock-rate = (int) 8000, " + "clock-rate = (int) [ 6000, 48000 ], " "encoding-name = (string) \"SPEEX\", " "encoding-params = (string) \"1\"") ); +static GstStateChangeReturn gst_rtp_speex_pay_change_state (GstElement * + element, GstStateChange transition); + static gboolean gst_rtp_speex_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static GstFlowReturn gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload * @@ -71,6 +77,9 @@ gst_rtp_speex_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_speex_pay_src_template)); gst_element_class_set_details (element_class, &gst_rtp_speex_pay_details); + + GST_DEBUG_CATEGORY_INIT (rtpspeexpay_debug, "rtpspeexpay", 0, + "Speex RTP Payloader"); } static void @@ -84,7 +93,7 @@ gst_rtp_speex_pay_class_init (GstRtpSPEEXPayClass * klass) gstelement_class = (GstElementClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; - parent_class = g_type_class_peek_parent (klass); + gstelement_class->change_state = gst_rtp_speex_pay_change_state; gstbasertppayload_class->set_caps = gst_rtp_speex_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_speex_pay_handle_buffer; @@ -101,10 +110,95 @@ gst_rtp_speex_pay_init (GstRtpSPEEXPay * rtpspeexpay, static gboolean gst_rtp_speex_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) { - gst_basertppayload_set_options (payload, "audio", FALSE, "speex", 8000); - gst_basertppayload_set_outcaps (payload, NULL); + /* don't configure yet, we wait for the ident packet */ + return TRUE; +} + +static gboolean +gst_rtp_speex_pay_parse_ident (GstRtpSPEEXPay * rtpspeexpay, + const guint8 * data, guint size) +{ + guint32 version, header_size, rate, mode, nb_channels; + GstBaseRTPPayload *payload; + gchar *cstr; + + /* 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, "Speex ")) + goto wrong_header; + + /* skip header and version string */ + data += 28; + + version = GST_READ_UINT32_LE (data); + if (version != 1) + goto wrong_version; + + data += 4; + /* ensure sizes */ + header_size = GST_READ_UINT32_LE (data); + if (header_size < 80) + goto header_too_small; + + if (size < header_size) + goto payload_too_small; + + data += 4; + rate = GST_READ_UINT32_LE (data); + data += 4; + mode = GST_READ_UINT32_LE (data); + data += 8; + nb_channels = GST_READ_UINT32_LE (data); + + GST_DEBUG_OBJECT (rtpspeexpay, "rate %d, mode %d, nb_channels %d", + rate, mode, nb_channels); + + payload = GST_BASE_RTP_PAYLOAD (rtpspeexpay); + + gst_basertppayload_set_options (payload, "audio", FALSE, "SPEEX", rate); + cstr = g_strdup_printf ("%d", nb_channels); + gst_basertppayload_set_outcaps (payload, "encoding-params", + G_TYPE_STRING, cstr, NULL); + g_free (cstr); return TRUE; + + /* ERRORS */ +too_small: + { + GST_DEBUG_OBJECT (rtpspeexpay, + "ident packet too small, need at least 32 bytes"); + return FALSE; + } +wrong_header: + { + GST_DEBUG_OBJECT (rtpspeexpay, + "ident packet does not start with \"Speex \""); + return FALSE; + } +wrong_version: + { + GST_DEBUG_OBJECT (rtpspeexpay, "can only handle version 1, have version %d", + version); + return FALSE; + } +header_too_small: + { + GST_DEBUG_OBJECT (rtpspeexpay, + "header size too small, need at least 80 bytes, " "got only %d", + header_size); + return FALSE; + } +payload_too_small: + { + GST_DEBUG_OBJECT (rtpspeexpay, + "payload too small, need at least %d bytes, got only %d", header_size, + size); + return FALSE; + } } static GstFlowReturn @@ -121,6 +215,26 @@ gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload * basepayload, rtpspeexpay = GST_RTP_SPEEX_PAY (basepayload); size = GST_BUFFER_SIZE (buffer); + data = GST_BUFFER_DATA (buffer); + + switch (rtpspeexpay->packet) { + case 0: + /* ident packet. We need to parse the headers to construct the RTP + * properties. */ + if (!gst_rtp_speex_pay_parse_ident (rtpspeexpay, data, size)) + goto parse_error; + + ret = GST_FLOW_OK; + goto done; + case 1: + /* comment packet, we ignore it */ + ret = GST_FLOW_OK; + goto done; + default: + /* other packets go in the payload */ + break; + } + timestamp = GST_BUFFER_TIMESTAMP (buffer); /* FIXME, only one SPEEX frame per RTP packet for now */ @@ -135,8 +249,6 @@ gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload * basepayload, /* get payload */ payload = gst_rtp_buffer_get_payload (outbuf); - data = GST_BUFFER_DATA (buffer); - /* copy data in payload */ memcpy (&payload[0], data, size); @@ -144,6 +256,46 @@ gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload * basepayload, ret = gst_basertppayload_push (basepayload, outbuf); +done: + rtpspeexpay->packet++; + + return ret; + + /* ERRORS */ +parse_error: + { + GST_ELEMENT_ERROR (rtpspeexpay, STREAM, DECODE, (NULL), + ("Error parsing first identification packet.")); + return GST_FLOW_ERROR; + } +} + +static GstStateChangeReturn +gst_rtp_speex_pay_change_state (GstElement * element, GstStateChange transition) +{ + GstRtpSPEEXPay *rtpspeexpay; + GstStateChangeReturn ret; + + rtpspeexpay = GST_RTP_SPEEX_PAY (element); + + switch (transition) { + case GST_STATE_CHANGE_NULL_TO_READY: + break; + case GST_STATE_CHANGE_READY_TO_PAUSED: + rtpspeexpay->packet = 0; + break; + default: + break; + } + + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + + switch (transition) { + case GST_STATE_CHANGE_READY_TO_NULL: + break; + default: + break; + } return ret; } |