/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
+ * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:element-rtpL16pay
+ * @title: rtpL16pay
+ * @see_also: rtpL16depay
+ *
+ * Payload raw audio into RTP packets according to RFC 3551.
+ * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
+ *
+ * ## Example pipeline
+ * |[
+ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink
+ * ]| This example pipeline will payload raw audio. Refer to
+ * the rtpL16depay example to depayload and play the RTP stream.
+ *
*/
#ifdef HAVE_CONFIG_H
#include <string.h>
+#include <gst/audio/audio.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpL16pay.h"
+#include "gstrtpchannels.h"
GST_DEBUG_CATEGORY_STATIC (rtpL16pay_debug);
#define GST_CAT_DEFAULT (rtpL16pay_debug)
-/* elementfactory information */
-static const GstElementDetails gst_rtp_L16_pay_details =
-GST_ELEMENT_DETAILS ("RTP packet payloader",
- "Codec/Payloader/Network",
- "Payload-encode Raw audio into RTP packets (RFC 3551)",
- "Wim Taymans <wim@fluendo.com>");
-
static GstStaticPadTemplate gst_rtp_L16_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "endianness = (int) BIG_ENDIAN, "
- "signed = (boolean) true, "
- "width = (int) 16, "
- "depth = (int) 16, "
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) S16BE, "
+ "layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
"clock-rate = (int) 44100")
);
-static void gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass);
-static void gst_rtp_L16_pay_base_init (GstRtpL16PayClass * klass);
-static void gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay);
-static void gst_rtp_L16_pay_finalize (GObject * object);
-
-static gboolean gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload,
+static gboolean gst_rtp_L16_pay_setcaps (GstRTPBasePayload * basepayload,
GstCaps * caps);
-static GstFlowReturn gst_rtp_L16_pay_handle_buffer (GstBaseRTPPayload * pad,
+static GstCaps *gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload,
+ GstPad * pad, GstCaps * filter);
+static GstFlowReturn
+gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer);
-static GstCaps *gst_rtp_L16_pay_getcaps (GstBaseRTPPayload * rtppayload,
- GstPad * pad);
-static GstBaseRTPPayloadClass *parent_class = NULL;
-
-static GType
-gst_rtp_L16_pay_get_type (void)
-{
- static GType rtpL16pay_type = 0;
-
- if (!rtpL16pay_type) {
- static const GTypeInfo rtpL16pay_info = {
- sizeof (GstRtpL16PayClass),
- (GBaseInitFunc) gst_rtp_L16_pay_base_init,
- NULL,
- (GClassInitFunc) gst_rtp_L16_pay_class_init,
- NULL,
- NULL,
- sizeof (GstRtpL16Pay),
- 0,
- (GInstanceInitFunc) gst_rtp_L16_pay_init,
- };
-
- rtpL16pay_type =
- g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpL16Pay",
- &rtpL16pay_info, 0);
- }
- return rtpL16pay_type;
-}
-
-static void
-gst_rtp_L16_pay_base_init (GstRtpL16PayClass * klass)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_L16_pay_src_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_L16_pay_sink_template));
-
- gst_element_class_set_details (element_class, &gst_rtp_L16_pay_details);
-}
+#define gst_rtp_L16_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpL16Pay, gst_rtp_L16_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
static void
gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass)
{
- GObjectClass *gobject_class;
GstElementClass *gstelement_class;
- GstBaseRTPPayloadClass *gstbasertppayload_class;
+ GstRTPBasePayloadClass *gstrtpbasepayload_class;
- gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
- gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+ gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
- parent_class = g_type_class_peek_parent (klass);
+ gstrtpbasepayload_class->set_caps = gst_rtp_L16_pay_setcaps;
+ gstrtpbasepayload_class->get_caps = gst_rtp_L16_pay_getcaps;
+ gstrtpbasepayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer;
- gobject_class->finalize = gst_rtp_L16_pay_finalize;
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_L16_pay_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_L16_pay_sink_template);
- gstbasertppayload_class->set_caps = gst_rtp_L16_pay_setcaps;
- gstbasertppayload_class->get_caps = gst_rtp_L16_pay_getcaps;
- gstbasertppayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer;
+ gst_element_class_set_static_metadata (gstelement_class,
+ "RTP audio payloader", "Codec/Payloader/Network/RTP",
+ "Payload-encode Raw audio into RTP packets (RFC 3551)",
+ "Wim Taymans <wim.taymans@gmail.com>");
GST_DEBUG_CATEGORY_INIT (rtpL16pay_debug, "rtpL16pay", 0,
"L16 RTP Payloader");
static void
gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay)
{
- rtpL16pay->adapter = gst_adapter_new ();
-}
-
-static void
-gst_rtp_L16_pay_finalize (GObject * object)
-{
- GstRtpL16Pay *rtpL16pay;
-
- rtpL16pay = GST_RTP_L16_PAY (object);
+ GstRTPBaseAudioPayload *rtpbaseaudiopayload;
- g_object_unref (rtpL16pay->adapter);
- rtpL16pay->adapter = NULL;
+ rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL16pay);
- G_OBJECT_CLASS (parent_class)->finalize (object);
+ /* tell rtpbaseaudiopayload that this is a sample based codec */
+ gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
}
static gboolean
-gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
+gst_rtp_L16_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpL16Pay *rtpL16pay;
- GstStructure *structure;
- gint channels, rate;
gboolean res;
gchar *params;
+ GstAudioInfo *info;
+ const GstRTPChannelOrder *order;
+ GstRTPBaseAudioPayload *rtpbaseaudiopayload;
+ rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
rtpL16pay = GST_RTP_L16_PAY (basepayload);
- structure = gst_caps_get_structure (caps, 0);
+ info = &rtpL16pay->info;
+ gst_audio_info_init (info);
+ if (!gst_audio_info_from_caps (info, caps))
+ goto invalid_caps;
- /* first parse input caps */
- if (!gst_structure_get_int (structure, "rate", &rate))
- goto no_rate;
+ order = gst_rtp_channels_get_by_pos (info->channels, info->position);
+ rtpL16pay->order = order;
- if (!gst_structure_get_int (structure, "channels", &channels))
- goto no_channels;
+ gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L16",
+ info->rate);
+ params = g_strdup_printf ("%d", info->channels);
+ if (!order && info->channels > 2) {
+ GST_ELEMENT_WARNING (rtpL16pay, STREAM, DECODE,
+ (NULL), ("Unknown channel order for %d channels", info->channels));
+ }
+
+ if (order && order->name) {
+ res = gst_rtp_base_payload_set_outcaps (basepayload,
+ "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
+ info->channels, "channel-order", G_TYPE_STRING, order->name, NULL);
+ } else {
+ res = gst_rtp_base_payload_set_outcaps (basepayload,
+ "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
+ info->channels, NULL);
+ }
- gst_basertppayload_set_options (basepayload, "audio", TRUE, "L16", rate);
- params = g_strdup_printf ("%d", channels);
- res = gst_basertppayload_set_outcaps (basepayload,
- "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
- channels, NULL);
g_free (params);
- rtpL16pay->rate = rate;
- rtpL16pay->channels = channels;
+ /* octet-per-sample is 2 * channels for L16 */
+ gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,
+ 2 * info->channels);
return res;
/* ERRORS */
-no_rate:
+invalid_caps:
{
- GST_DEBUG_OBJECT (rtpL16pay, "no rate given");
- return FALSE;
- }
-no_channels:
- {
- GST_DEBUG_OBJECT (rtpL16pay, "no channels given");
+ GST_DEBUG_OBJECT (rtpL16pay, "invalid caps");
return FALSE;
}
}
-static GstFlowReturn
-gst_rtp_L16_pay_flush (GstRtpL16Pay * rtpL16pay, guint len)
-{
- GstBuffer *outbuf;
- guint8 *payload;
- GstFlowReturn ret;
- guint samples;
- GstClockTime duration;
-
- /* now alloc output buffer */
- outbuf = gst_rtp_buffer_new_allocate (len, 0, 0);
-
- /* get payload, this is now writable */
- payload = gst_rtp_buffer_get_payload (outbuf);
-
- /* copy and flush data out of adapter into the RTP payload */
- gst_adapter_copy (rtpL16pay->adapter, payload, 0, len);
- gst_adapter_flush (rtpL16pay->adapter, len);
-
- samples = len / (2 * rtpL16pay->channels);
- duration = gst_util_uint64_scale_int (samples, GST_SECOND, rtpL16pay->rate);
-
- GST_BUFFER_TIMESTAMP (outbuf) = rtpL16pay->first_ts;
- GST_BUFFER_DURATION (outbuf) = duration;
-
- /* increase count (in ts) of data pushed to basertppayload */
- if (GST_CLOCK_TIME_IS_VALID (rtpL16pay->first_ts))
- rtpL16pay->first_ts += duration;
-
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpL16pay), outbuf);
-
- return ret;
-}
-
-static GstFlowReturn
-gst_rtp_L16_pay_handle_buffer (GstBaseRTPPayload * basepayload,
- GstBuffer * buffer)
-{
- GstRtpL16Pay *rtpL16pay;
- GstFlowReturn ret = GST_FLOW_OK;
- guint payload_len;
- GstClockTime timestamp;
- guint mtu, avail;
-
- rtpL16pay = GST_RTP_L16_PAY (basepayload);
- mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpL16pay);
-
- timestamp = GST_BUFFER_TIMESTAMP (buffer);
-
- if (GST_BUFFER_IS_DISCONT (buffer))
- gst_adapter_clear (rtpL16pay->adapter);
-
- avail = gst_adapter_available (rtpL16pay->adapter);
- if (avail == 0) {
- rtpL16pay->first_ts = timestamp;
- }
-
- /* push buffer in adapter */
- gst_adapter_push (rtpL16pay->adapter, buffer);
-
- /* get payload len for MTU */
- payload_len = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
-
- /* flush complete MTU while we have enough data in the adapter */
- while (avail >= payload_len) {
- /* flush payload_len bytes */
- ret = gst_rtp_L16_pay_flush (rtpL16pay, payload_len);
- if (ret != GST_FLOW_OK)
- break;
-
- avail = gst_adapter_available (rtpL16pay->adapter);
- }
- return ret;
-}
-
static GstCaps *
-gst_rtp_L16_pay_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad)
+gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
+ GstCaps * filter)
{
GstCaps *otherpadcaps;
GstCaps *caps;
- otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
- caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
+ caps = gst_pad_get_pad_template_caps (pad);
+ otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
if (otherpadcaps) {
if (!gst_caps_is_empty (otherpadcaps)) {
GstStructure *structure;
gint rate;
structure = gst_caps_get_structure (otherpadcaps, 0);
+ caps = gst_caps_make_writable (caps);
if (gst_structure_get_int (structure, "channels", &channels)) {
gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
} else if (gst_structure_get_int (structure, "payload", &pt)) {
- if (pt == 10)
+ if (pt == GST_RTP_PAYLOAD_L16_STEREO)
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
- else if (pt == 11)
+ else if (pt == GST_RTP_PAYLOAD_L16_MONO)
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
}
if (gst_structure_get_int (structure, "clock-rate", &rate)) {
gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
} else if (gst_structure_get_int (structure, "payload", &pt)) {
- if (pt == 10 || pt == 11)
+ if (pt == GST_RTP_PAYLOAD_L16_STEREO || pt == GST_RTP_PAYLOAD_L16_MONO)
gst_caps_set_simple (caps, "rate", G_TYPE_INT, 44100, NULL);
}
gst_caps_unref (otherpadcaps);
}
+ if (filter) {
+ GstCaps *tcaps = caps;
+
+ caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
+ gst_caps_unref (tcaps);
+ }
+
return caps;
}
+static GstFlowReturn
+gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload,
+ GstBuffer * buffer)
+{
+ GstRtpL16Pay *rtpL16pay;
+
+ rtpL16pay = GST_RTP_L16_PAY (basepayload);
+ buffer = gst_buffer_make_writable (buffer);
+
+ if (rtpL16pay->order &&
+ !gst_audio_buffer_reorder_channels (buffer, rtpL16pay->info.finfo->format,
+ rtpL16pay->info.channels, rtpL16pay->info.position,
+ rtpL16pay->order->pos)) {
+ return GST_FLOW_ERROR;
+ }
+
+ return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,
+ buffer);
+}
+
gboolean
gst_rtp_L16_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpL16pay",
- GST_RANK_NONE, GST_TYPE_RTP_L16_PAY);
+ GST_RANK_SECONDARY, GST_TYPE_RTP_L16_PAY);
}
projects / platform / upstream / gst-plugins-good.git / blobdiff