/* 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
#include <string.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/multichannel.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",
+GST_ELEMENT_DETAILS ("RTP audio payloader",
"Codec/Payloader/Network",
"Payload-encode Raw audio into RTP packets (RFC 3551)",
- "Wim Taymans <wim@fluendo.com>");
+ "Wim Taymans <wim.taymans@gmail.com>");
static GstStaticPadTemplate gst_rtp_L16_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
"payload = (int) [ 96, 127 ], "
"clock-rate = (int) [ 1, MAX ], "
"encoding-name = (string) \"L16\", "
- "channels = (int) [ 1, MAX ], "
- "rate = (int) [ 1, MAX ];"
+ "channels = (int) [ 1, MAX ];"
"application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) { " GST_RTP_PAYLOAD_L16_STEREO_STRING ", "
- GST_RTP_PAYLOAD_L16_MONO_STRING " }," "clock-rate = (int) 44100")
+ "encoding-name = (string) \"L16\", "
+ "payload = (int) " GST_RTP_PAYLOAD_L16_STEREO_STRING ", "
+ "clock-rate = (int) 44100;"
+ "application/x-rtp, "
+ "media = (string) \"audio\", "
+ "encoding-name = (string) \"L16\", "
+ "payload = (int) " GST_RTP_PAYLOAD_L16_MONO_STRING ", "
+ "clock-rate = (int) 44100")
);
static void gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass);
GstCaps * caps);
static GstFlowReturn gst_rtp_L16_pay_handle_buffer (GstBaseRTPPayload * pad,
GstBuffer * buffer);
+static GstCaps *gst_rtp_L16_pay_getcaps (GstBaseRTPPayload * rtppayload,
+ GstPad * pad);
static GstBaseRTPPayloadClass *parent_class = NULL;
gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass)
{
GObjectClass *gobject_class;
- GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
- gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_rtp_L16_pay_finalize;
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_DEBUG_CATEGORY_INIT (rtpL16pay_debug, "rtpL16pay", 0,
GstRtpL16Pay *rtpL16pay;
GstStructure *structure;
gint channels, rate;
+ gboolean res;
+ gchar *params;
+ GstAudioChannelPosition *pos;
+ const GstRTPChannelOrder *order;
rtpL16pay = GST_RTP_L16_PAY (basepayload);
if (!gst_structure_get_int (structure, "channels", &channels))
goto no_channels;
+ /* get the channel order */
+ pos = gst_audio_get_channel_positions (structure);
+ if (pos)
+ order = gst_rtp_channels_get_by_pos (channels, pos);
+ else
+ order = NULL;
+
gst_basertppayload_set_options (basepayload, "audio", TRUE, "L16", rate);
- gst_basertppayload_set_outcaps (basepayload,
- "channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, rate, NULL);
+ params = g_strdup_printf ("%d", channels);
+
+ if (!order && channels > 2) {
+ GST_ELEMENT_WARNING (rtpL16pay, STREAM, DECODE,
+ (NULL), ("Unknown channel order for %d channels", channels));
+ }
+
+ if (order && order->name) {
+ res = gst_basertppayload_set_outcaps (basepayload,
+ "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
+ channels, "channel-order", G_TYPE_STRING, order->name, NULL);
+ } else {
+ res = gst_basertppayload_set_outcaps (basepayload,
+ "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
+ channels, NULL);
+ }
+
+ g_free (params);
+ g_free (pos);
rtpL16pay->rate = rate;
rtpL16pay->channels = channels;
- return TRUE;
+ return res;
/* ERRORS */
no_rate:
guint samples;
GstClockTime duration;
+ /* calculate the amount of samples and round down the length */
+ samples = len / (2 * rtpL16pay->channels);
+ len = samples * (2 * rtpL16pay->channels);
+
/* now alloc output buffer */
outbuf = gst_rtp_buffer_new_allocate (len, 0, 0);
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;
return ret;
}
+static GstCaps *
+gst_rtp_L16_pay_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad)
+{
+ GstCaps *otherpadcaps;
+ GstCaps *caps;
+
+ otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
+ caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
+
+ if (otherpadcaps) {
+ if (!gst_caps_is_empty (otherpadcaps)) {
+ GstStructure *structure;
+ gint channels;
+ gint pt;
+ gint rate;
+
+ structure = gst_caps_get_structure (otherpadcaps, 0);
+
+ 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)
+ gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
+ else if (pt == 11)
+ 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)
+ gst_caps_set_simple (caps, "rate", G_TYPE_INT, 44100, NULL);
+ }
+
+ }
+ gst_caps_unref (otherpadcaps);
+ }
+ return caps;
+}
+
gboolean
gst_rtp_L16_pay_plugin_init (GstPlugin * plugin)
{
projects / platform / upstream / gstreamer.git / blobdiff