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Diffstat (limited to 'gst/rtp/gstrtpmp4apay.c')
-rwxr-xr-xgst/rtp/gstrtpmp4apay.c452
1 files changed, 452 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpmp4apay.c b/gst/rtp/gstrtpmp4apay.c
new file mode 100755
index 0000000..17a80a8
--- /dev/null
+++ b/gst/rtp/gstrtpmp4apay.c
@@ -0,0 +1,452 @@
+/* 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, 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)
+
+/* FIXME: add framed=(boolean)true once our encoders have this field set
+ * on their output caps */
+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, "
+ "stream-format=(string)raw")
+ );
+
+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_finalize (GObject * object);
+
+static gboolean gst_rtp_mp4a_pay_setcaps (GstRTPBasePayload * payload,
+ GstCaps * caps);
+static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload *
+ payload, GstBuffer * buffer);
+
+#define gst_rtp_mp4a_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpMP4APay, gst_rtp_mp4a_pay, GST_TYPE_RTP_BASE_PAYLOAD)
+
+ static void gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstRTPBasePayloadClass *gstrtpbasepayload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
+
+ gobject_class->finalize = gst_rtp_mp4a_pay_finalize;
+
+ gstrtpbasepayload_class->set_caps = gst_rtp_mp4a_pay_setcaps;
+ gstrtpbasepayload_class->handle_buffer = gst_rtp_mp4a_pay_handle_buffer;
+
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtp_mp4a_pay_src_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtp_mp4a_pay_sink_template));
+
+ gst_element_class_set_static_metadata (gstelement_class,
+ "RTP MPEG4 audio payloader", "Codec/Payloader/Network/RTP",
+ "Payload MPEG4 audio as RTP packets (RFC 3016)",
+ "Wim Taymans <wim.taymans@gmail.com>");
+
+ 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 const unsigned int 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)
+{
+ GstMapInfo map;
+ guint8 *data;
+ gsize size;
+ guint8 objectType;
+ guint8 samplingIdx;
+ guint8 channelCfg;
+
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
+ data = map.data;
+ size = map.size;
+
+ 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);
+
+ gst_buffer_unmap (buffer, &map);
+
+ return TRUE;
+
+ /* ERROR */
+too_short:
+ {
+ GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
+ (NULL),
+ ("config string too short, expected 2 bytes, got %" G_GSIZE_FORMAT,
+ size));
+ gst_buffer_unmap (buffer, &map);
+ return FALSE;
+ }
+invalid_object:
+ {
+ GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
+ (NULL), ("invalid object type 0"));
+ gst_buffer_unmap (buffer, &map);
+ return FALSE;
+ }
+wrong_freq:
+ {
+ GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
+ (NULL), ("unsupported frequency index %d", samplingIdx));
+ gst_buffer_unmap (buffer, &map);
+ return FALSE;
+ }
+wrong_channels:
+ {
+ GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
+ (NULL), ("unsupported number of channels %d, must < 8", channelCfg));
+ gst_buffer_unmap (buffer, &map);
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay)
+{
+ gchar *config;
+ GValue v = { 0 };
+ gboolean res;
+
+ g_value_init (&v, GST_TYPE_BUFFER);
+ gst_value_set_buffer (&v, rtpmp4apay->config);
+ config = gst_value_serialize (&v);
+
+ res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4apay),
+ "cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL);
+
+ g_value_unset (&v);
+ g_free (config);
+
+ return res;
+}
+
+static gboolean
+gst_rtp_mp4a_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
+{
+ GstRtpMP4APay *rtpmp4apay;
+ GstStructure *structure;
+ const GValue *codec_data;
+ gboolean res, framed = TRUE;
+ const gchar *stream_format;
+
+ rtpmp4apay = GST_RTP_MP4A_PAY (payload);
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ /* this is already handled by the template caps, but it is better
+ * to leave here to have meaningful warning messages when linking
+ * fails */
+ stream_format = gst_structure_get_string (structure, "stream-format");
+ if (stream_format) {
+ if (strcmp (stream_format, "raw") != 0) {
+ GST_WARNING_OBJECT (rtpmp4apay, "AAC's stream-format must be 'raw', "
+ "%s is not supported", stream_format);
+ return FALSE;
+ }
+ } else {
+ GST_WARNING_OBJECT (rtpmp4apay, "AAC's stream-format not specified, "
+ "assuming 'raw'");
+ }
+
+ 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;
+ GstMapInfo map;
+ GstMapInfo cmap;
+ guint i;
+
+ 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;
+
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
+
+ /* make the StreamMuxConfig, we need 15 bits for the header */
+ cbuffer = gst_buffer_new_and_alloc (map.size + 2);
+ gst_buffer_map (cbuffer, &cmap, GST_MAP_WRITE);
+
+ memset (cmap.data, 0, map.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)
+ */
+ cmap.data[0] = 0x40;
+ cmap.data[1] = 0x00;
+
+ /* append the config bits, shifting them 1 bit left */
+ for (i = 0; i < map.size; i++) {
+ cmap.data[i + 1] |= ((map.data[i] & 0x80) >> 7);
+ cmap.data[i + 2] |= ((map.data[i] & 0x7f) << 1);
+ }
+
+ gst_buffer_unmap (cbuffer, &cmap);
+ gst_buffer_unmap (buffer, &map);
+
+ /* now we can configure the buffer */
+ if (rtpmp4apay->config)
+ gst_buffer_unref (rtpmp4apay->config);
+ rtpmp4apay->config = cbuffer;
+ }
+ }
+
+ if (gst_structure_get_boolean (structure, "framed", &framed) && !framed) {
+ GST_WARNING_OBJECT (payload, "Need framed AAC data as input!");
+ }
+
+ gst_rtp_base_payload_set_options (payload, "audio", TRUE, "MP4A-LATM",
+ rtpmp4apay->rate);
+
+ res = gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
+
+ return res;
+
+ /* 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 (GstRTPBasePayload * basepayload,
+ GstBuffer * buffer)
+{
+ GstRtpMP4APay *rtpmp4apay;
+ GstFlowReturn ret;
+ GstBuffer *outbuf;
+ guint count, mtu;
+ GstMapInfo map;
+ gsize size;
+ guint8 *data;
+ gboolean fragmented;
+ GstClockTime timestamp;
+
+ ret = GST_FLOW_OK;
+
+ rtpmp4apay = GST_RTP_MP4A_PAY (basepayload);
+
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
+ size = map.size;
+ data = map.data;
+
+ timestamp = GST_BUFFER_TIMESTAMP (buffer);
+
+ fragmented = FALSE;
+ mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp4apay);
+
+ while (size > 0) {
+ guint towrite;
+ guint8 *payload;
+ guint payload_len;
+ guint packet_len;
+ GstRTPBuffer rtp = { NULL };
+
+ /* 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 %" G_GSIZE_FORMAT ", 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 */
+ gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
+ payload = gst_rtp_buffer_get_payload (&rtp);
+
+ 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 (&rtp, size == 0);
+
+ gst_rtp_buffer_unmap (&rtp);
+
+ /* copy incomming timestamp (if any) to outgoing buffers */
+ GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+
+ ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp4apay), outbuf);
+
+ fragmented = TRUE;
+ }
+
+ gst_buffer_unmap (buffer, &map);
+ gst_buffer_unref (buffer);
+
+ return ret;
+}
+
+gboolean
+gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpmp4apay",
+ GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_PAY);
+}
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-24 11:34:14 +0000