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gst/rtp/: More fixage, set endoder-params correctly in the payloader.
Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process): * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_reset_packet), (gst_rtp_vorbis_pay_init_packet), (gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_parse_id), (gst_rtp_vorbis_pay_handle_buffer): More fixage, set endoder-params correctly in the payloader.
This commit is contained in:
Wim Taymans
parent
e4ba501855
commit
25a44f8e02
5 changed files with 206 additions and 101 deletions
12
ChangeLog
12
ChangeLog
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@ -1,3 +1,15 @@
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2006-09-22 Wim Taymans <wim@fluendo.com>
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* gst/rtp/Makefile.am:
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
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* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
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(gst_rtp_vorbis_pay_reset_packet),
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(gst_rtp_vorbis_pay_init_packet),
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(gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_parse_id),
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(gst_rtp_vorbis_pay_handle_buffer):
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More fixage, set endoder-params correctly in the payloader.
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2006-09-22 Tim-Philipp Müller <tim at centricular dot net>
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* gst/autodetect/gstautoaudiosink.c:
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@ -2,7 +2,6 @@ plugin_LTLIBRARIES = libgstrtp.la
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libgstrtp_la_SOURCES = \
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gstrtp.c \
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gstrtpasfdepay.c \
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gstrtpdepay.c \
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gstrtpmpadepay.c \
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gstrtpmpapay.c \
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@ -44,7 +43,6 @@ libgstrtp_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) \
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libgstrtp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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noinst_HEADERS = \
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gstrtpasfdepay.h \
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gstrtpL16depay.h \
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gstrtpL16pay.h \
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gstrtpamrdepay.h \
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@ -21,7 +21,6 @@
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#include "config.h"
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#endif
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#include "gstrtpasfdepay.h"
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#include "gstrtpdepay.h"
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#include "gstrtppcmupay.h"
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#include "gstrtppcmapay.h"
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@ -52,9 +51,6 @@
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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if (!gst_rtp_asf_depay_plugin_init (plugin))
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return FALSE;
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if (!gst_rtp_depay_plugin_init (plugin))
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return FALSE;
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@ -201,7 +201,7 @@ gst_rtp_vorbis_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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GST_DEBUG_OBJECT (depayload, "got RTP packet of size %d", payload_len);
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/* we need at least 4 bytes for the packet header */
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if (payload_len < 4)
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if (G_UNLIKELY (payload_len < 4))
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goto packet_short;
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payload = gst_rtp_buffer_get_payload (buf);
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@ -220,7 +220,7 @@ gst_rtp_vorbis_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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* pkts: number of packets.
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*/
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VDT = (header & 0x30) >> 4;
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if (VDT == 3)
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if (G_UNLIKELY (VDT == 3))
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goto ignore_reserved;
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ident = (header >> 8) & 0xffffff;
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@ -235,8 +235,8 @@ gst_rtp_vorbis_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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payload += 4;
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payload_len -= 4;
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GST_DEBUG_OBJECT (depayload, "ident: %u, F: %d, packets: %d", ident, F,
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packets);
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GST_DEBUG_OBJECT (depayload, "ident: %u, F: %d, VDT: %d, packets: %d", ident,
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F, VDT, packets);
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/* fragmented packets, assemble */
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if (F != 0) {
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@ -268,6 +268,7 @@ gst_rtp_vorbis_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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/* construct assembled buffer */
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payload_len = gst_adapter_available (rtpvorbisdepay->adapter);
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payload = gst_adapter_take (rtpvorbisdepay->adapter, payload_len);
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/* fix the length */
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payload[0] = ((payload_len - 2) >> 8) & 0xff;
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payload[1] = (payload_len - 2) & 0xff;
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to_free = payload;
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@ -307,11 +308,11 @@ gst_rtp_vorbis_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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payload_len);
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/* skip packet if something odd happens */
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if (length > payload_len)
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if (G_UNLIKELY (length > payload_len))
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goto length_short;
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/* create buffer for packet */
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if (to_free) {
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if (G_UNLIKELY (to_free)) {
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outbuf = gst_buffer_new ();
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GST_BUFFER_DATA (outbuf) = payload;
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GST_BUFFER_MALLOCDATA (outbuf) = to_free;
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@ -123,40 +123,41 @@ gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
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gst_basertppayload_set_options (basepayload, "audio", TRUE, "vorbis", 8000);
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gst_basertppayload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, "1",
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/* don't set the defaults
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*/
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NULL);
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return TRUE;
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}
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static void
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gst_rtp_vorbis_pay_init_packet (GstRtpVorbisPay * rtpvorbispay)
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gst_rtp_vorbis_pay_reset_packet (GstRtpVorbisPay * rtpvorbispay, guint8 VDT)
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{
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guint payload_len;
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if (rtpvorbispay->packet)
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gst_buffer_unref (rtpvorbispay->packet);
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GST_DEBUG_OBJECT (rtpvorbispay, "reset packet");
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GST_DEBUG_OBJECT (rtpvorbispay, "starting new packet");
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/* new packet allocate max packet size */
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rtpvorbispay->packet =
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gst_rtp_buffer_new_allocate_len (GST_BASE_RTP_PAYLOAD_MTU
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(rtpvorbispay), 0, 0);
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rtpvorbispay->payload_pos = 4;
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payload_len = gst_rtp_buffer_get_payload_len (rtpvorbispay->packet);
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rtpvorbispay->payload_left = payload_len - 4;
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rtpvorbispay->payload_duration = 0;
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rtpvorbispay->payload_ident = 0;
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rtpvorbispay->payload_F = 0;
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rtpvorbispay->payload_VDT = 0;
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rtpvorbispay->payload_VDT = VDT;
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rtpvorbispay->payload_pkts = 0;
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}
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static void
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gst_rtp_vorbis_pay_init_packet (GstRtpVorbisPay * rtpvorbispay, guint8 VDT)
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{
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GST_DEBUG_OBJECT (rtpvorbispay, "starting new packet, VDT: %d", VDT);
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if (rtpvorbispay->packet)
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gst_buffer_unref (rtpvorbispay->packet);
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/* new packet allocate max packet size */
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rtpvorbispay->packet =
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gst_rtp_buffer_new_allocate_len (GST_BASE_RTP_PAYLOAD_MTU
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(rtpvorbispay), 0, 0);
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gst_rtp_vorbis_pay_reset_packet (rtpvorbispay, VDT);
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}
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static GstFlowReturn
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gst_rtp_vorbis_pay_flush_packet (GstRtpVorbisPay * rtpvorbispay)
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{
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@ -165,7 +166,7 @@ gst_rtp_vorbis_pay_flush_packet (GstRtpVorbisPay * rtpvorbispay)
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guint hlen;
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/* check for empty packet */
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if (!rtpvorbispay || rtpvorbispay->payload_pos <= 4)
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if (!rtpvorbispay->packet || rtpvorbispay->payload_pos <= 4)
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return GST_FLOW_OK;
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GST_DEBUG_OBJECT (rtpvorbispay, "flushing packet");
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@ -200,44 +201,162 @@ gst_rtp_vorbis_pay_flush_packet (GstRtpVorbisPay * rtpvorbispay)
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rtpvorbispay->packet);
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rtpvorbispay->packet = NULL;
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/* prepare new packet */
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gst_rtp_vorbis_pay_init_packet (rtpvorbispay);
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return ret;
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}
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static gboolean
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gst_rtp_vorbis_pay_parse_id (GstBaseRTPPayload * basepayload, guint8 * data,
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guint size)
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{
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guint8 channels;
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gint32 rate, version;
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gchar *cstr;
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if (G_UNLIKELY (size < 16))
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goto too_short;
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if (G_UNLIKELY (memcmp (data, "\001vorbis", 7)))
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goto invalid_start;
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data += 7;
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if (G_UNLIKELY ((version = GST_READ_UINT32_LE (data)) != 0))
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goto invalid_version;
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data += 4;
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if (G_UNLIKELY ((channels = *data++) < 1))
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goto invalid_channels;
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if (G_UNLIKELY ((rate = GST_READ_UINT32_LE (data)) < 1))
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goto invalid_rate;
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cstr = g_strdup_printf ("%d", channels);
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gst_basertppayload_set_options (basepayload, "audio", TRUE, "vorbis", rate);
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gst_basertppayload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, cstr,
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/* don't set the defaults
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*/
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NULL);
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g_free (cstr);
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return TRUE;
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/* ERRORS */
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too_short:
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{
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GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
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("Identification packet is too short, need at least 16, got %d", size),
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(NULL));
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return FALSE;
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}
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invalid_start:
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{
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GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
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("Invalid header start in identification packet"), (NULL));
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return FALSE;
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}
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invalid_version:
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{
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GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
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("Invalid version, expected 0, got %d", version), (NULL));
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return FALSE;
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}
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invalid_rate:
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{
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GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
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("Invalid rate %d", rate), (NULL));
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return FALSE;
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}
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invalid_channels:
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{
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GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
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("Invalid channels %d", channels), (NULL));
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return FALSE;
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}
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}
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static GstFlowReturn
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gst_rtp_vorbis_pay_append_buffer (GstRtpVorbisPay * rtpvorbispay,
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gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstFlowReturn res;
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guint size;
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GstClockTime duration;
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GstRtpVorbisPay *rtpvorbispay;
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GstFlowReturn ret;
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guint size, newsize;
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guint8 *data;
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guint packet_len;
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GstClockTime duration, newduration;
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gboolean flush;
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guint8 VDT;
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guint plen;
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guint8 *ppos, *payload, *data;
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guint8 *ppos, *payload;
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gboolean fragmented;
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res = GST_FLOW_OK;
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if (rtpvorbispay->payload_left < 2)
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return res;
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rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
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size = GST_BUFFER_SIZE (buffer);
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/* skip packets that are too big */
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if (size > 0xffff)
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return res;
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data = GST_BUFFER_DATA (buffer);
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duration = GST_BUFFER_DURATION (buffer);
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GST_DEBUG_OBJECT (rtpvorbispay, "size %u, duration %" GST_TIME_FORMAT,
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size, GST_TIME_ARGS (duration));
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if (G_UNLIKELY (size < 1 || size > 0xffff))
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goto wrong_size;
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/* find packet type */
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if (data[0] & 1) {
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/* header */
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if (data[0] == 1) {
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/* identification, we need to parse this in order to get the clock rate. */
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if (G_UNLIKELY (!gst_rtp_vorbis_pay_parse_id (basepayload, data, size)))
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goto parse_id_failed;
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VDT = 1;
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} else if (data[0] == 5)
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/* setup */
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VDT = 1;
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else if (data[0] == 3)
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VDT = 2;
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else
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goto unknown_header;
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} else
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/* data */
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VDT = 0;
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/* size increases with packet length and 2 bytes size eader. */
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newduration = rtpvorbispay->payload_duration;
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if (duration != GST_CLOCK_TIME_NONE)
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newduration += duration;
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newsize = rtpvorbispay->payload_pos + 2 + size;
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packet_len = gst_rtp_buffer_calc_packet_len (newsize, 0, 0);
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/* check buffer filled against length and max latency */
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flush = gst_basertppayload_is_filled (basepayload, packet_len, newduration);
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/* we can store up to 15 vorbis packets in one RTP packet. */
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flush |= (rtpvorbispay->payload_pkts == 15);
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/* flush if we have a new VDT */
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if (rtpvorbispay->packet)
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flush |= (rtpvorbispay->payload_VDT != VDT);
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if (flush)
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ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
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/* create new packet if we must */
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if (!rtpvorbispay->packet)
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gst_rtp_vorbis_pay_init_packet (rtpvorbispay, VDT);
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payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
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ppos = payload + rtpvorbispay->payload_pos;
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fragmented = FALSE;
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ret = GST_FLOW_OK;
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/* put buffer in packet, it either fits completely or needs to be fragmented
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* over multiple RTP packets. */
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while (size) {
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plen = MIN (rtpvorbispay->payload_left - 2, size);
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GST_DEBUG_OBJECT (rtpvorbispay, "append %u bytes", plen);
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/* data is copied in the payload with a 2 byte length header */
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ppos[0] = (plen >> 8) & 0xff;
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ppos[1] = (plen & 0xff);
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memcpy (&ppos[2], data, plen);
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@ -256,12 +375,7 @@ gst_rtp_vorbis_pay_append_buffer (GstRtpVorbisPay * rtpvorbispay,
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/* fragment continues, set F to 0x2. */
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rtpvorbispay->payload_F = 0x2;
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} else {
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if (size == 0) {
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/* unfragmented packet, update stats for next packet */
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rtpvorbispay->payload_pkts++;
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if (duration != GST_CLOCK_TIME_NONE)
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rtpvorbispay->payload_duration += duration;
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} else {
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if (size > 0) {
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/* fragmented packet starts, set F to 0x1, mark ourselves as
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* fragmented. */
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rtpvorbispay->payload_F = 0x1;
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@ -271,58 +385,42 @@ gst_rtp_vorbis_pay_append_buffer (GstRtpVorbisPay * rtpvorbispay,
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if (fragmented) {
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/* fragmented packets are always flushed and have ptks of 0 */
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rtpvorbispay->payload_pkts = 0;
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res = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
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/* get new pointers */
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ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
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|
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if (size > 0) {
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/* start new packet and get pointers. VDT stays the same. */
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gst_rtp_vorbis_pay_init_packet (rtpvorbispay,
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rtpvorbispay->payload_VDT);
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payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
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ppos = payload + rtpvorbispay->payload_pos;
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}
|
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}
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|
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return res;
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}
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static GstFlowReturn
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gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * basepayload,
|
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GstBuffer * buffer)
|
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{
|
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GstRtpVorbisPay *rtpvorbispay;
|
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GstFlowReturn ret;
|
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guint size, newsize;
|
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guint packet_len;
|
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GstClockTime duration, newduration;
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gboolean flush;
|
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rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
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|
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size = GST_BUFFER_SIZE (buffer);
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duration = GST_BUFFER_DURATION (buffer);
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GST_DEBUG_OBJECT (rtpvorbispay, "size %u, duration %" GST_TIME_FORMAT,
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size, GST_TIME_ARGS (duration));
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if (!rtpvorbispay->packet)
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gst_rtp_vorbis_pay_init_packet (rtpvorbispay);
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/* size increases with packet length and 2 bytes size eader. */
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newduration = rtpvorbispay->payload_duration;
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} else {
|
||||
/* unfragmented packet, update stats for next packet, size == 0 and we
|
||||
* exit the while loop */
|
||||
rtpvorbispay->payload_pkts++;
|
||||
if (duration != GST_CLOCK_TIME_NONE)
|
||||
newduration += duration;
|
||||
|
||||
newsize = rtpvorbispay->payload_pos + 2 + size;
|
||||
packet_len = gst_rtp_buffer_calc_packet_len (newsize, 0, 0);
|
||||
|
||||
/* check buffer filled against length and max latency */
|
||||
flush = gst_basertppayload_is_filled (basepayload, packet_len, newduration);
|
||||
/* we can store up to 15 vorbis packets in one RTP packet. */
|
||||
flush |= (rtpvorbispay->payload_pkts == 15);
|
||||
|
||||
if (flush)
|
||||
ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
|
||||
|
||||
/* put buffer in packet */
|
||||
ret = gst_rtp_vorbis_pay_append_buffer (rtpvorbispay, buffer);
|
||||
|
||||
rtpvorbispay->payload_duration += duration;
|
||||
}
|
||||
}
|
||||
return ret;
|
||||
|
||||
/* ERRORS */
|
||||
wrong_size:
|
||||
{
|
||||
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
|
||||
("Invalid packet size (1 < %d <= 0xffff)", size), (NULL));
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
parse_id_failed:
|
||||
{
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
unknown_header:
|
||||
{
|
||||
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
|
||||
("Ignoring unknown header received"), (NULL));
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
}
|
||||
|
||||
gboolean
|
||||
|
|
|
|||
Loading…
Reference in a new issue