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gst/rtp/: Fix speex (de)payloader. Fixes #358040.
Original commit message from CVS: * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init), (gst_rtp_speex_depay_get_mode), (gst_rtp_speex_depay_setcaps), (gst_rtp_speex_depay_process): * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_base_init), (gst_rtp_speex_pay_class_init), (gst_rtp_speex_pay_setcaps), (gst_rtp_speex_pay_parse_ident), (gst_rtp_speex_pay_handle_buffer), (gst_rtp_speex_pay_change_state): * gst/rtp/gstrtpspeexpay.h: Fix speex (de)payloader. Fixes #358040.
This commit is contained in:
Wim Taymans
parent
e7495dcfbb
commit
d3948d2323
4 changed files with 252 additions and 21 deletions
12
ChangeLog
12
ChangeLog
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@ -1,3 +1,15 @@
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2007-03-05 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init),
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(gst_rtp_speex_depay_get_mode), (gst_rtp_speex_depay_setcaps),
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(gst_rtp_speex_depay_process):
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* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_base_init),
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(gst_rtp_speex_pay_class_init), (gst_rtp_speex_pay_setcaps),
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(gst_rtp_speex_pay_parse_ident), (gst_rtp_speex_pay_handle_buffer),
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(gst_rtp_speex_pay_change_state):
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* gst/rtp/gstrtpspeexpay.h:
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Fix speex (de)payloader. Fixes #358040.
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2007-03-05 Jan Schmidt <thaytan@mad.scientist.com>
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* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset),
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@ -94,8 +94,6 @@ gst_rtp_speex_depay_class_init (GstRtpSPEEXDepayClass * klass)
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gstelement_class = (GstElementClass *) klass;
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gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gstbasertpdepayload_class->process = gst_rtp_speex_depay_process;
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gstbasertpdepayload_class->set_caps = gst_rtp_speex_depay_setcaps;
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}
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@ -107,37 +105,104 @@ gst_rtp_speex_depay_init (GstRtpSPEEXDepay * rtpspeexdepay,
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GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay)->clock_rate = 8000;
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}
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static gint
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gst_rtp_speex_depay_get_mode (gint rate)
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{
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if (rate > 25000)
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return 2;
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else if (rate > 12500)
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return 1;
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else
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return 0;
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}
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/* len 4 bytes LE,
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* vendor string (len bytes),
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* user_len 4 (0) bytes LE
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*/
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static const gchar gst_rtp_speex_comment[] =
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"\045\0\0\0Depayloaded with GStreamer speexdepay\0\0\0\0";
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static gboolean
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gst_rtp_speex_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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{
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GstCaps *srccaps;
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gboolean ret;
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GstStructure *structure;
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GstRtpSPEEXDepay *rtpspeexdepay;
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gint clock_rate, nb_channels;
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GstBuffer *buf;
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guint8 *data;
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const gchar *params;
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srccaps =
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gst_static_pad_template_get_caps (&gst_rtp_speex_depay_src_template);
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ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
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rtpspeexdepay = GST_RTP_SPEEX_DEPAY (depayload);
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gst_caps_unref (srccaps);
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return ret;
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (structure, "clock-rate", &clock_rate);
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depayload->clock_rate = clock_rate;
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if (!(params = gst_structure_get_string (structure, "encoding-params")))
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nb_channels = 1;
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else {
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nb_channels = atoi (params);
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}
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/* construct minimal header and comment packet for the decoder */
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buf = gst_buffer_new_and_alloc (80);
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data = GST_BUFFER_DATA (buf);
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memcpy (data, "Speex ", 8);
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data += 8;
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memcpy (data, "1.1.12", 7);
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data += 20;
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GST_WRITE_UINT32_LE (data, 1); /* version */
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data += 4;
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GST_WRITE_UINT32_LE (data, 80); /* header_size */
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data += 4;
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GST_WRITE_UINT32_LE (data, clock_rate); /* rate */
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data += 4;
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GST_WRITE_UINT32_LE (data, gst_rtp_speex_depay_get_mode (clock_rate)); /* mode */
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data += 4;
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GST_WRITE_UINT32_LE (data, 4); /* mode_bitstream_version */
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data += 4;
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GST_WRITE_UINT32_LE (data, nb_channels); /* nb_channels */
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data += 4;
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GST_WRITE_UINT32_LE (data, -1); /* bitrate */
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data += 4;
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GST_WRITE_UINT32_LE (data, 0xa0); /* frame_size */
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data += 4;
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GST_WRITE_UINT32_LE (data, 0); /* VBR */
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data += 4;
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GST_WRITE_UINT32_LE (data, 1); /* frames_per_packet */
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data += 4;
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GST_WRITE_UINT32_LE (data, 0); /* extra_headers */
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data += 4;
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GST_WRITE_UINT32_LE (data, 0); /* reserved1 */
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data += 4;
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GST_WRITE_UINT32_LE (data, 0); /* reserved2 */
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gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay), buf);
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buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_speex_comment));
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memcpy (GST_BUFFER_DATA (buf), gst_rtp_speex_comment,
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sizeof (gst_rtp_speex_comment));
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gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay), buf);
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return TRUE;
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}
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static GstBuffer *
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gst_rtp_speex_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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{
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GstBuffer *outbuf = NULL;
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gint payload_len;
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guint8 *payload;
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GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
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GST_BUFFER_SIZE (buf),
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gst_rtp_buffer_get_marker (buf),
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gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
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payload_len = gst_rtp_buffer_get_payload_len (buf);
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payload = gst_rtp_buffer_get_payload (buf);
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/* nothing special to be done */
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outbuf = gst_rtp_buffer_get_payload_buffer (buf);
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outbuf = gst_buffer_new_and_alloc (payload_len);
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memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
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return outbuf;
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}
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@ -27,6 +27,9 @@
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#include "gstrtpspeexpay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpspeexpay_debug);
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#define GST_CAT_DEFAULT (rtpspeexpay_debug)
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/* elementfactory information */
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static const GstElementDetails gst_rtp_speex_pay_details =
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GST_ELEMENT_DETAILS ("RTP packet payloader",
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@ -48,11 +51,14 @@ GST_STATIC_PAD_TEMPLATE ("src",
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GST_STATIC_CAPS ("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) 8000, "
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"clock-rate = (int) [ 6000, 48000 ], "
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"encoding-name = (string) \"SPEEX\", "
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"encoding-params = (string) \"1\"")
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);
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static GstStateChangeReturn gst_rtp_speex_pay_change_state (GstElement *
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element, GstStateChange transition);
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static gboolean gst_rtp_speex_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload *
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@ -71,6 +77,9 @@ gst_rtp_speex_pay_base_init (gpointer klass)
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_speex_pay_src_template));
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gst_element_class_set_details (element_class, &gst_rtp_speex_pay_details);
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GST_DEBUG_CATEGORY_INIT (rtpspeexpay_debug, "rtpspeexpay", 0,
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"Speex RTP Payloader");
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}
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static void
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@ -84,7 +93,7 @@ gst_rtp_speex_pay_class_init (GstRtpSPEEXPayClass * klass)
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gstelement_class = (GstElementClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gstelement_class->change_state = gst_rtp_speex_pay_change_state;
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gstbasertppayload_class->set_caps = gst_rtp_speex_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_speex_pay_handle_buffer;
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@ -101,10 +110,95 @@ gst_rtp_speex_pay_init (GstRtpSPEEXPay * rtpspeexpay,
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static gboolean
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gst_rtp_speex_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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gst_basertppayload_set_options (payload, "audio", FALSE, "speex", 8000);
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gst_basertppayload_set_outcaps (payload, NULL);
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/* don't configure yet, we wait for the ident packet */
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return TRUE;
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}
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static gboolean
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gst_rtp_speex_pay_parse_ident (GstRtpSPEEXPay * rtpspeexpay,
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const guint8 * data, guint size)
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{
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guint32 version, header_size, rate, mode, nb_channels;
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GstBaseRTPPayload *payload;
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gchar *cstr;
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/* we need the header string (8), the version string (20), the version
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* and the header length. */
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if (size < 36)
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goto too_small;
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if (!g_str_has_prefix ((const gchar *) data, "Speex "))
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goto wrong_header;
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/* skip header and version string */
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data += 28;
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version = GST_READ_UINT32_LE (data);
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if (version != 1)
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goto wrong_version;
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data += 4;
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/* ensure sizes */
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header_size = GST_READ_UINT32_LE (data);
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if (header_size < 80)
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goto header_too_small;
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if (size < header_size)
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goto payload_too_small;
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data += 4;
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rate = GST_READ_UINT32_LE (data);
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data += 4;
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mode = GST_READ_UINT32_LE (data);
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data += 8;
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nb_channels = GST_READ_UINT32_LE (data);
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GST_DEBUG_OBJECT (rtpspeexpay, "rate %d, mode %d, nb_channels %d",
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rate, mode, nb_channels);
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payload = GST_BASE_RTP_PAYLOAD (rtpspeexpay);
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gst_basertppayload_set_options (payload, "audio", FALSE, "SPEEX", rate);
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cstr = g_strdup_printf ("%d", nb_channels);
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gst_basertppayload_set_outcaps (payload, "encoding-params",
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G_TYPE_STRING, cstr, NULL);
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g_free (cstr);
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return TRUE;
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/* ERRORS */
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too_small:
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{
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GST_DEBUG_OBJECT (rtpspeexpay,
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"ident packet too small, need at least 32 bytes");
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return FALSE;
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}
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wrong_header:
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{
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GST_DEBUG_OBJECT (rtpspeexpay,
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"ident packet does not start with \"Speex \"");
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return FALSE;
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}
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wrong_version:
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{
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GST_DEBUG_OBJECT (rtpspeexpay, "can only handle version 1, have version %d",
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version);
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return FALSE;
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}
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header_too_small:
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{
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GST_DEBUG_OBJECT (rtpspeexpay,
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"header size too small, need at least 80 bytes, " "got only %d",
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header_size);
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return FALSE;
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}
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payload_too_small:
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{
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GST_DEBUG_OBJECT (rtpspeexpay,
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"payload too small, need at least %d bytes, got only %d", header_size,
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size);
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return FALSE;
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}
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}
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static GstFlowReturn
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@ -121,6 +215,26 @@ gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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rtpspeexpay = GST_RTP_SPEEX_PAY (basepayload);
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size = GST_BUFFER_SIZE (buffer);
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data = GST_BUFFER_DATA (buffer);
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switch (rtpspeexpay->packet) {
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case 0:
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/* ident packet. We need to parse the headers to construct the RTP
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* properties. */
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if (!gst_rtp_speex_pay_parse_ident (rtpspeexpay, data, size))
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goto parse_error;
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ret = GST_FLOW_OK;
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goto done;
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case 1:
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/* comment packet, we ignore it */
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ret = GST_FLOW_OK;
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goto done;
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default:
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/* other packets go in the payload */
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break;
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}
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timestamp = GST_BUFFER_TIMESTAMP (buffer);
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/* FIXME, only one SPEEX frame per RTP packet for now */
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@ -135,8 +249,6 @@ gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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/* get payload */
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payload = gst_rtp_buffer_get_payload (outbuf);
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data = GST_BUFFER_DATA (buffer);
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/* copy data in payload */
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memcpy (&payload[0], data, size);
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@ -144,6 +256,46 @@ gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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ret = gst_basertppayload_push (basepayload, outbuf);
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done:
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rtpspeexpay->packet++;
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return ret;
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/* ERRORS */
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parse_error:
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{
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GST_ELEMENT_ERROR (rtpspeexpay, STREAM, DECODE, (NULL),
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("Error parsing first identification packet."));
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return GST_FLOW_ERROR;
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}
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}
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static GstStateChangeReturn
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gst_rtp_speex_pay_change_state (GstElement * element, GstStateChange transition)
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{
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GstRtpSPEEXPay *rtpspeexpay;
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GstStateChangeReturn ret;
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rtpspeexpay = GST_RTP_SPEEX_PAY (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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rtpspeexpay->packet = 0;
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_NULL:
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break;
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default:
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break;
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}
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return ret;
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}
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@ -38,6 +38,8 @@ typedef struct _GstRtpSPEEXPayClass GstRtpSPEEXPayClass;
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struct _GstRtpSPEEXPay
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{
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GstBaseRTPPayload payload;
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guint64 packet;
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};
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struct _GstRtpSPEEXPayClass
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