#include "gstrtpg726pay.h"
-static const GstElementDetails gst_rtp_g726_pay_details =
-GST_ELEMENT_DETAILS ("RTP packet payloader",
- "Codec/Payloader/Network",
- "Payload-encodes G.726 audio into a RTP packet",
- "Axis Communications <dev-gstreamer@axis.com>");
+GST_DEBUG_CATEGORY_STATIC (rtpg726pay_debug);
+#define GST_CAT_DEFAULT (rtpg726pay_debug)
+
+#define DEFAULT_FORCE_AAL2 TRUE
+
+enum
+{
+ PROP_0,
+ PROP_FORCE_AAL2,
+ PROP_LAST
+};
static GstStaticPadTemplate gst_rtp_g726_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 8000, "
- "encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\" } ")
+ "encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\", "
+ " \"AAL2-G726-16\", \"AAL2-G726-24\", \"AAL2-G726-32\", \"AAL2-G726-40\" } ")
);
-static gboolean gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload,
+static void gst_rtp_g726_pay_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtp_g726_pay_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+
+static gboolean gst_rtp_g726_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
+static GstFlowReturn gst_rtp_g726_pay_handle_buffer (GstRTPBasePayload *
+ payload, GstBuffer * buffer);
-GST_BOILERPLATE (GstRtpG726Pay, gst_rtp_g726_pay, GstBaseRTPAudioPayload,
- GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
-
-static void
-gst_rtp_g726_pay_base_init (gpointer klass)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_g726_pay_sink_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_g726_pay_src_template));
- gst_element_class_set_details (element_class, &gst_rtp_g726_pay_details);
-}
+#define gst_rtp_g726_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpG726Pay, gst_rtp_g726_pay,
+ GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
static void
gst_rtp_g726_pay_class_init (GstRtpG726PayClass * 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;
+
+ gobject_class->set_property = gst_rtp_g726_pay_set_property;
+ gobject_class->get_property = gst_rtp_g726_pay_get_property;
+
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_FORCE_AAL2,
+ g_param_spec_boolean ("force-aal2", "Force AAL2",
+ "Force AAL2 encoding for compatibility with bad depayloaders",
+ DEFAULT_FORCE_AAL2, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- parent_class = g_type_class_peek_parent (klass);
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtp_g726_pay_sink_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtp_g726_pay_src_template));
- gstbasertppayload_class->set_caps = gst_rtp_g726_pay_setcaps;
+ gst_element_class_set_details_simple (gstelement_class, "RTP G.726 payloader",
+ "Codec/Payloader/Network/RTP",
+ "Payload-encodes G.726 audio into a RTP packet",
+ "Axis Communications <dev-gstreamer@axis.com>");
+
+ gstrtpbasepayload_class->set_caps = gst_rtp_g726_pay_setcaps;
+ gstrtpbasepayload_class->handle_buffer = gst_rtp_g726_pay_handle_buffer;
+
+ GST_DEBUG_CATEGORY_INIT (rtpg726pay_debug, "rtpg726pay", 0,
+ "G.726 RTP Payloader");
}
static void
-gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay, GstRtpG726PayClass * klass)
+gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay)
{
- GstBaseRTPAudioPayload *basertpaudiopayload;
+ GstRTPBaseAudioPayload *rtpbaseaudiopayload;
- basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg726pay);
+ rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg726pay);
- GST_BASE_RTP_PAYLOAD (rtpg726pay)->clock_rate = 8000;
+ GST_RTP_BASE_PAYLOAD (rtpg726pay)->clock_rate = 8000;
+
+ rtpg726pay->force_aal2 = DEFAULT_FORCE_AAL2;
/* sample based codec */
- gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
+ gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
}
static gboolean
-gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+gst_rtp_g726_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gchar *encoding_name;
- GstStructure *structure = gst_caps_get_structure (caps, 0);
- GstBaseRTPAudioPayload *basertpaudiopayload;
- gint bitrate;
+ GstStructure *structure;
+ GstRTPBaseAudioPayload *rtpbaseaudiopayload;
+ GstRtpG726Pay *pay;
+ GstCaps *peercaps;
+ gboolean res;
+
+ rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (payload);
+ pay = GST_RTP_G726_PAY (payload);
- basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (payload);
+ structure = gst_caps_get_structure (caps, 0);
- if (!gst_structure_get_int (structure, "bitrate", &bitrate))
- bitrate = 32000;
+ if (!gst_structure_get_int (structure, "bitrate", &pay->bitrate))
+ pay->bitrate = 32000;
- switch (bitrate) {
+ GST_DEBUG_OBJECT (payload, "using bitrate %d", pay->bitrate);
+
+ pay->aal2 = FALSE;
+
+ /* first see what we can do with the bitrate */
+ switch (pay->bitrate) {
case 16000:
encoding_name = g_strdup ("G726-16");
- gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
+ gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
2);
break;
case 24000:
encoding_name = g_strdup ("G726-24");
- gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
+ gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
3);
break;
case 32000:
encoding_name = g_strdup ("G726-32");
- gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
+ gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
4);
break;
case 40000:
encoding_name = g_strdup ("G726-40");
- gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
+ gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
5);
break;
default:
goto invalid_bitrate;
}
- gst_basertppayload_set_options (payload, "audio", TRUE, encoding_name, 8000);
- gst_basertppayload_set_outcaps (payload, NULL);
+ GST_DEBUG_OBJECT (payload, "selected base encoding %s", encoding_name);
+
+ /* now see if we need to produce AAL2 or not */
+ peercaps = gst_pad_peer_query_caps (payload->srcpad, NULL);
+ if (peercaps) {
+ GstCaps *filter, *intersect;
+ gchar *capsstr;
+
+ GST_DEBUG_OBJECT (payload, "have peercaps %" GST_PTR_FORMAT, peercaps);
+
+ capsstr = g_strdup_printf ("application/x-rtp, "
+ "media = (string) \"audio\", "
+ "clock-rate = (int) 8000, "
+ "encoding-name = (string) %s; "
+ "application/x-rtp, "
+ "media = (string) \"audio\", "
+ "clock-rate = (int) 8000, "
+ "encoding-name = (string) AAL2-%s", encoding_name, encoding_name);
+ filter = gst_caps_from_string (capsstr);
+ g_free (capsstr);
+
+ /* intersect to filter */
+ intersect = gst_caps_intersect (peercaps, filter);
+ gst_caps_unref (peercaps);
+
+ GST_DEBUG_OBJECT (payload, "intersected to %" GST_PTR_FORMAT, intersect);
+
+ if (!intersect)
+ goto no_format;
+ if (gst_caps_is_empty (intersect)) {
+ gst_caps_unref (intersect);
+ goto no_format;
+ }
+
+ structure = gst_caps_get_structure (intersect, 0);
+
+ /* now see what encoding name we settled on, we need to dup because the
+ * string goes away when we unref the intersection below. */
+ encoding_name =
+ g_strdup (gst_structure_get_string (structure, "encoding-name"));
+
+ /* if we managed to negotiate to AAL2, we definatly are going to do AAL2
+ * encoding. Else we only encode AAL2 when explicitly set by the
+ * property. */
+ if (g_str_has_prefix (encoding_name, "AAL2-"))
+ pay->aal2 = TRUE;
+ else
+ pay->aal2 = pay->force_aal2;
+
+ GST_DEBUG_OBJECT (payload, "final encoding %s, AAL2 %d", encoding_name,
+ pay->aal2);
+
+ gst_caps_unref (intersect);
+ } else {
+ /* downstream can do anything but we prefer the better supported non-AAL2 */
+ pay->aal2 = pay->force_aal2;
+ GST_DEBUG_OBJECT (payload, "no peer caps, AAL2 %d", pay->aal2);
+ }
+
+ gst_rtp_base_payload_set_options (payload, "audio", TRUE, encoding_name,
+ 8000);
+ res = gst_rtp_base_payload_set_outcaps (payload, NULL);
g_free (encoding_name);
- return TRUE;
+ return res;
/* ERRORS */
invalid_bitrate:
{
- GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", bitrate);
+ GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", pay->bitrate);
return FALSE;
}
+no_format:
+ {
+ GST_ERROR_OBJECT (payload, "could not negotiate format");
+ return FALSE;
+ }
+}
+
+static GstFlowReturn
+gst_rtp_g726_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
+{
+ GstFlowReturn res;
+ GstRtpG726Pay *pay;
+
+ pay = GST_RTP_G726_PAY (payload);
+
+ if (!pay->aal2) {
+ GstMapInfo map;
+ guint8 *data, tmp;
+ gsize size;
+
+ /* for non AAL2, we need to reshuffle the bytes, we can do this in-place
+ * when the buffer is writable. */
+ buffer = gst_buffer_make_writable (buffer);
+
+ gst_buffer_map (buffer, &map, GST_MAP_READWRITE);
+ data = map.data;
+ size = map.size;
+
+ GST_LOG_OBJECT (pay, "packing %" G_GSIZE_FORMAT " bytes of data", map.size);
+
+ /* we need to reshuffle the bytes, output is of the form:
+ * A B C D .. with the number of bits depending on the bitrate. */
+ switch (pay->bitrate) {
+ case 16000:
+ {
+ /* 0
+ * 0 1 2 3 4 5 6 7
+ * +-+-+-+-+-+-+-+-+-
+ * |D D|C C|B B|A A| ...
+ * |0 1|0 1|0 1|0 1|
+ * +-+-+-+-+-+-+-+-+-
+ */
+ while (size > 0) {
+ tmp = *data;
+ *data++ = ((tmp & 0xc0) >> 6) |
+ ((tmp & 0x30) >> 2) | ((tmp & 0x0c) << 2) | ((tmp & 0x03) << 6);
+ size--;
+ }
+ break;
+ }
+ case 24000:
+ {
+ /* 0 1 2
+ * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
+ * |C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ...
+ * |1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1|
+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
+ */
+ while (size > 2) {
+ tmp = *data;
+ *data++ = ((tmp & 0xc0) >> 6) |
+ ((tmp & 0x38) >> 1) | ((tmp & 0x07) << 5);
+ tmp = *data;
+ *data++ = ((tmp & 0x80) >> 7) |
+ ((tmp & 0x70) >> 3) | ((tmp & 0x0e) << 4) | ((tmp & 0x01) << 7);
+ tmp = *data;
+ *data++ = ((tmp & 0xe0) >> 5) |
+ ((tmp & 0x1c) >> 2) | ((tmp & 0x03) << 6);
+ size -= 3;
+ }
+ break;
+ }
+ case 32000:
+ {
+ /* 0 1
+ * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
+ * |B B B B|A A A A|D D D D|C C C C| ...
+ * |0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3|
+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
+ */
+ while (size > 0) {
+ tmp = *data;
+ *data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4);
+ size--;
+ }
+ break;
+ }
+ case 40000:
+ {
+ /* 0 1 2 3 4
+ * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0
+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
+ * |B B B|A A A A A|D|C C C C C|B B|E E E E|D D D D|G G|F F F F F|E|H H H H H|G G G|
+ * |2 3 4|0 1 2 3 4|4|0 1 2 3 4|0 1|1 2 3 4|0 1 2 3|3 4|0 1 2 3 4|0|0 1 2 3 4|0 1 2|
+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
+ */
+ while (size > 4) {
+ tmp = *data;
+ *data++ = ((tmp & 0xe0) >> 5) | ((tmp & 0x1f) << 3);
+ tmp = *data;
+ *data++ = ((tmp & 0x80) >> 7) |
+ ((tmp & 0x7c) >> 2) | ((tmp & 0x03) << 6);
+ tmp = *data;
+ *data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4);
+ tmp = *data;
+ *data++ = ((tmp & 0xc0) >> 6) |
+ ((tmp & 0x3e) << 2) | ((tmp & 0x01) << 7);
+ tmp = *data;
+ *data++ = ((tmp & 0xf8) >> 3) | ((tmp & 0x07) << 5);
+ size -= 5;
+ }
+ break;
+ }
+ }
+ gst_buffer_unmap (buffer, &map);
+ }
+
+ res =
+ GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (payload,
+ buffer);
+
+ return res;
+}
+
+static void
+gst_rtp_g726_pay_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRtpG726Pay *rtpg726pay;
+
+ rtpg726pay = GST_RTP_G726_PAY (object);
+
+ switch (prop_id) {
+ case PROP_FORCE_AAL2:
+ rtpg726pay->force_aal2 = g_value_get_boolean (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_rtp_g726_pay_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRtpG726Pay *rtpg726pay;
+
+ rtpg726pay = GST_RTP_G726_PAY (object);
+
+ switch (prop_id) {
+ case PROP_FORCE_AAL2:
+ g_value_set_boolean (value, rtpg726pay->force_aal2);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
}
gboolean
gst_rtp_g726_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg726pay",
- GST_RANK_NONE, GST_TYPE_RTP_G726_PAY);
+ GST_RANK_SECONDARY, GST_TYPE_RTP_G726_PAY);
}
projects / platform / upstream / gst-plugins-good.git / blobdiff