--- /dev/null
+/* GStreamer
+ * Copyright (C) <2008> 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
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <string.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpmp4apay.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpmp4apay_debug);
+#define GST_CAT_DEFAULT (rtpmp4apay_debug)
+
+/* elementfactory information */
+static const GstElementDetails gst_rtp_mp4apay_details =
+GST_ELEMENT_DETAILS ("RTP packet payloader",
+ "Codec/Payloader/Network",
+ "Payload MPEG4 audio as RTP packets (RFC 3016)",
+ "Wim Taymans <wim.taymans@gmail.com>");
+
+static GstStaticPadTemplate gst_rtp_mp4a_pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/mpeg," "mpegversion=(int) 4")
+ );
+
+static GstStaticPadTemplate gst_rtp_mp4a_pay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) [1, MAX ], "
+ "encoding-name = (string) \"MP4A-LATM\""
+ /* All optional parameters
+ *
+ * "cpresent = (string) \"0\""
+ * "config="
+ */
+ )
+ );
+
+
+static void gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass);
+static void gst_rtp_mp4a_pay_base_init (GstRtpMP4APayClass * klass);
+static void gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay);
+static void gst_rtp_mp4a_pay_finalize (GObject * object);
+
+static gboolean gst_rtp_mp4a_pay_setcaps (GstBaseRTPPayload * payload,
+ GstCaps * caps);
+static GstStateChangeReturn gst_rtp_mp4a_pay_change_state (GstElement * element,
+ GstStateChange transition);
+static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstBaseRTPPayload *
+ payload, GstBuffer * buffer);
+
+static GstBaseRTPPayloadClass *parent_class = NULL;
+
+static GType
+gst_rtp_mp4a_pay_get_type (void)
+{
+ static GType rtpmp4apay_type = 0;
+
+ if (!rtpmp4apay_type) {
+ static const GTypeInfo rtpmp4apay_info = {
+ sizeof (GstRtpMP4APayClass),
+ (GBaseInitFunc) gst_rtp_mp4a_pay_base_init,
+ NULL,
+ (GClassInitFunc) gst_rtp_mp4a_pay_class_init,
+ NULL,
+ NULL,
+ sizeof (GstRtpMP4APay),
+ 0,
+ (GInstanceInitFunc) gst_rtp_mp4a_pay_init,
+ };
+
+ rtpmp4apay_type =
+ g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMP4APay",
+ &rtpmp4apay_info, 0);
+ }
+ return rtpmp4apay_type;
+}
+
+static void
+gst_rtp_mp4a_pay_base_init (GstRtpMP4APayClass * klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_mp4a_pay_src_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_mp4a_pay_sink_template));
+
+ gst_element_class_set_details (element_class, &gst_rtp_mp4apay_details);
+}
+
+static void
+gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * 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_mp4a_pay_finalize;
+
+ gstelement_class->change_state = gst_rtp_mp4a_pay_change_state;
+
+ gstbasertppayload_class->set_caps = gst_rtp_mp4a_pay_setcaps;
+ gstbasertppayload_class->handle_buffer = gst_rtp_mp4a_pay_handle_buffer;
+
+ GST_DEBUG_CATEGORY_INIT (rtpmp4apay_debug, "rtpmp4apay", 0,
+ "MP4A-LATM RTP Payloader");
+}
+
+static void
+gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay)
+{
+ rtpmp4apay->rate = 90000;
+ rtpmp4apay->profile = g_strdup ("1");
+}
+
+static void
+gst_rtp_mp4a_pay_finalize (GObject * object)
+{
+ GstRtpMP4APay *rtpmp4apay;
+
+ rtpmp4apay = GST_RTP_MP4A_PAY (object);
+
+ g_free (rtpmp4apay->params);
+ rtpmp4apay->params = NULL;
+
+ if (rtpmp4apay->config)
+ gst_buffer_unref (rtpmp4apay->config);
+ rtpmp4apay->config = NULL;
+
+ g_free (rtpmp4apay->profile);
+ rtpmp4apay->profile = NULL;
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static unsigned sampling_table[16] = {
+ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+ 16000, 12000, 11025, 8000, 7350, 0, 0, 0
+};
+
+static gboolean
+gst_rtp_mp4a_pay_parse_audio_config (GstRtpMP4APay * rtpmp4apay,
+ GstBuffer * buffer)
+{
+ guint8 *data;
+ guint size;
+ guint8 objectType;
+ guint8 samplingIdx;
+ guint8 channelCfg;
+
+ data = GST_BUFFER_DATA (buffer);
+ size = GST_BUFFER_SIZE (buffer);
+
+ if (size < 2)
+ goto too_short;
+
+ /* any object type is fine, we need to copy it to the profile-level-id field. */
+ objectType = (data[0] & 0xf8) >> 3;
+ if (objectType == 0)
+ goto invalid_object;
+
+ samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
+ /* only fixed values for now */
+ if (samplingIdx > 12 && samplingIdx != 15)
+ goto wrong_freq;
+
+ channelCfg = ((data[1] & 0x78) >> 3);
+ if (channelCfg > 7)
+ goto wrong_channels;
+
+ /* rtp rate depends on sampling rate of the audio */
+ if (samplingIdx == 15) {
+ if (size < 5)
+ goto too_short;
+
+ /* index of 15 means we get the rate in the next 24 bits */
+ rtpmp4apay->rate = ((data[1] & 0x7f) << 17) |
+ ((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7);
+ } else {
+ /* else use the rate from the table */
+ rtpmp4apay->rate = sampling_table[samplingIdx];
+ }
+ /* extra rtp params contain the number of channels */
+ g_free (rtpmp4apay->params);
+ rtpmp4apay->params = g_strdup_printf ("%d", channelCfg);
+ /* audio stream type */
+ rtpmp4apay->streamtype = "5";
+ /* profile */
+ g_free (rtpmp4apay->profile);
+ rtpmp4apay->profile = g_strdup_printf ("%d", objectType);
+
+ GST_DEBUG_OBJECT (rtpmp4apay,
+ "objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType,
+ samplingIdx, rtpmp4apay->rate, channelCfg);
+
+ return TRUE;
+
+ /* ERROR */
+too_short:
+ {
+ GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
+ (NULL), ("config string too short, expected 2 bytes, got %d", size));
+ return FALSE;
+ }
+invalid_object:
+ {
+ GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
+ (NULL), ("invalid object type 0"));
+ return FALSE;
+ }
+wrong_freq:
+ {
+ GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
+ (NULL), ("unsupported frequency index %d", samplingIdx));
+ return FALSE;
+ }
+wrong_channels:
+ {
+ GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
+ (NULL), ("unsupported number of channels %d, must < 8", channelCfg));
+ return FALSE;
+ }
+}
+
+static void
+gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay)
+{
+ gchar *config;
+ GValue v = { 0 };
+
+ g_value_init (&v, GST_TYPE_BUFFER);
+ gst_value_set_buffer (&v, rtpmp4apay->config);
+ config = gst_value_serialize (&v);
+
+ gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4apay),
+ "cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL);
+
+ g_value_unset (&v);
+ g_free (config);
+}
+
+static gboolean
+gst_rtp_mp4a_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+ GstRtpMP4APay *rtpmp4apay;
+ GstStructure *structure;
+ const GValue *codec_data;
+
+ rtpmp4apay = GST_RTP_MP4A_PAY (payload);
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ codec_data = gst_structure_get_value (structure, "codec_data");
+ if (codec_data) {
+ GST_LOG_OBJECT (rtpmp4apay, "got codec_data");
+ if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
+ GstBuffer *buffer, *cbuffer;
+ guint8 *config;
+ guint8 *data;
+ guint size, i;
+ gboolean res;
+
+ buffer = gst_value_get_buffer (codec_data);
+ GST_LOG_OBJECT (rtpmp4apay, "configuring codec_data");
+
+ /* parse buffer */
+ res = gst_rtp_mp4a_pay_parse_audio_config (rtpmp4apay, buffer);
+
+ if (!res)
+ goto config_failed;
+
+ size = GST_BUFFER_SIZE (buffer);
+ data = GST_BUFFER_DATA (buffer);
+
+ /* make the StreamMuxConfig, we need 15 bits for the header */
+ config = g_malloc0 (size + 2);
+
+ /* Create StreamMuxConfig according to ISO/IEC 14496-3:
+ *
+ * audioMuxVersion == 0 (1 bit)
+ * allStreamsSameTimeFraming == 1 (1 bit)
+ * numSubFrames == numSubFrames (6 bits)
+ * numProgram == 0 (4 bits)
+ * numLayer == 0 (3 bits)
+ */
+ config[0] = 0x40;
+ config[1] = 0x00;
+
+ /* append the config bits, shifting them 1 bit left */
+ for (i = 0; i < size; i++) {
+ config[i + 1] |= ((data[i] & 0x80) >> 7);
+ config[i + 2] |= ((data[i] & 0x7f) << 1);
+ }
+
+ cbuffer = gst_buffer_new ();
+ GST_BUFFER_DATA (cbuffer) = config;
+ GST_BUFFER_MALLOCDATA (cbuffer) = config;
+ GST_BUFFER_SIZE (cbuffer) = size + 2;
+
+ /* now we can configure the buffer */
+ if (rtpmp4apay->config)
+ gst_buffer_unref (rtpmp4apay->config);
+ rtpmp4apay->config = cbuffer;
+ }
+ }
+
+ gst_basertppayload_set_options (payload, "audio", TRUE, "MP4A-LATM",
+ rtpmp4apay->rate);
+
+ gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
+
+ return TRUE;
+
+ /* ERRORS */
+config_failed:
+ {
+ GST_DEBUG_OBJECT (rtpmp4apay, "failed to parse config");
+ return FALSE;
+ }
+}
+
+/* we expect buffers as exactly one complete AU
+ */
+static GstFlowReturn
+gst_rtp_mp4a_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+ GstBuffer * buffer)
+{
+ GstRtpMP4APay *rtpmp4apay;
+ GstFlowReturn ret;
+ GstBuffer *outbuf;
+ guint count, mtu, size;
+ guint8 *data;
+ gboolean fragmented;
+
+ ret = GST_FLOW_OK;
+
+ rtpmp4apay = GST_RTP_MP4A_PAY (basepayload);
+
+ size = GST_BUFFER_SIZE (buffer);
+ data = GST_BUFFER_DATA (buffer);
+
+ fragmented = FALSE;
+ mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpmp4apay);
+
+ while (size > 0) {
+ guint towrite;
+ guint8 *payload;
+ guint payload_len;
+ guint packet_len;
+
+ /* this will be the total lenght of the packet */
+ packet_len = gst_rtp_buffer_calc_packet_len (size, 0, 0);
+
+ if (!fragmented) {
+ /* first packet calculate space for the packet including the header */
+ count = size;
+ while (count >= 0xff) {
+ packet_len++;
+ count -= 0xff;
+ }
+ packet_len++;
+ }
+
+ /* fill one MTU or all available bytes */
+ towrite = MIN (packet_len, mtu);
+
+ /* this is the payload length */
+ payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
+
+ GST_DEBUG_OBJECT (rtpmp4apay,
+ "avail %d, towrite %d, packet_len %d, payload_len %d", size, towrite,
+ packet_len, payload_len);
+
+ /* create buffer to hold the payload. */
+ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+
+ /* copy payload */
+ payload = gst_rtp_buffer_get_payload (outbuf);
+
+ if (!fragmented) {
+ /* first packet write the header */
+ count = size;
+ while (count >= 0xff) {
+ *payload++ = 0xff;
+ payload_len--;
+ count -= 0xff;
+ }
+ *payload++ = count;
+ payload_len--;
+ }
+
+ /* copy data to payload */
+ memcpy (payload, data, payload_len);
+ data += payload_len;
+ size -= payload_len;
+
+ /* marker only if the packet is complete */
+ gst_rtp_buffer_set_marker (outbuf, size == 0);
+
+ ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4apay), outbuf);
+
+ fragmented = TRUE;
+ }
+ return ret;
+}
+
+static GstStateChangeReturn
+gst_rtp_mp4a_pay_change_state (GstElement * element, GstStateChange transition)
+{
+ GstRtpMP4APay *rtpmp4apay;
+ GstStateChangeReturn ret;
+
+ rtpmp4apay = GST_RTP_MP4A_PAY (element);
+
+ switch (transition) {
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+ switch (transition) {
+ default:
+ break;
+ }
+ return ret;
+}
+
+
+gboolean
+gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpmp4apay",
+ GST_RANK_NONE, GST_TYPE_RTP_MP4A_PAY);
+}
projects / platform / upstream / gstreamer.git / commitdiff