/* GStreamer
+ * Copyright (C) <2007> Nokia Corporation
+ * Copyright (C) <2007> Collabora Ltd
+ * @author: Olivier Crete <olivier.crete@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* Boston, MA 02111-1307, USA.
*/
+/*
+ * This payloader assumes that the data will ALWAYS come as zero or more
+ * 10 bytes frame of audio followed by 0 or 1 2 byte frame of silence.
+ * Any other buffer format won't work
+ */
+
#ifdef HAVE_CONFIG_H
-#include "config.h"
+#include <config.h>
#endif
-#include "gstrtpg729pay.h"
+#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/base/gstadapter.h>
+
+#include "gstrtpg729pay.h"
+
+/* TODO: fix gstrtpbuffer.h */
+#undef GST_RTP_PAYLOAD_G729
+#define GST_RTP_PAYLOAD_G729 18
+#undef GST_RTP_PAYLOAD_G729_STRING
+#define GST_RTP_PAYLOAD_G729_STRING "18"
+
+#define G729_FRAME_SIZE 10
+#define G729B_CN_FRAME_SIZE 2
+#define G729_FRAME_DURATION (10 * GST_MSECOND)
+#define G729_FRAME_DURATION_MS (10)
+
+static gboolean
+gst_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps);
+static GstFlowReturn
+gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf);
-/* elementfactory information */
-static GstElementDetails gst_rtpg729pay_details = {
- "RTP Payloader for G729 Audio",
- "Codec/Payloader/Network",
- "Packetize G729 audio streams into RTP packets",
- "Laurent Glayal <spglegle@yahoo.fr>"
-};
-GST_DEBUG_CATEGORY_STATIC (rtpg729pay_debug);
-#define GST_CAT_DEFAULT (rtpg729pay_debug)
+static const GstElementDetails gst_rtp_g729_pay_details =
+GST_ELEMENT_DETAILS ("G729 RTP packet payloader",
+ "Codec/Payloader/Network",
+ "Packetize G729 audio into RTP packets",
+ "Olivier Crete <olivier.crete@collabora.co.uk>");
-static GstStaticPadTemplate gst_rtpg729pay_sink_template =
+static GstStaticPadTemplate gst_rtp_g729_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/G729, channels=(int)1, rate=(int)8000")
+ GST_STATIC_CAPS ("audio/G729, " /* according to RFC 3555 */
+ "channels = (int) 1, " "rate = (int) 8000")
);
-static GstStaticPadTemplate gst_rtpg729pay_src_template =
+static GstStaticPadTemplate gst_rtp_g729_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) 8000, " "encoding-name = (string) \"G729\";"
+ "payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
+ "clock-rate = (int) 8000, "
+ "encoding-name = (string) \"G729\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
- "clock-rate = (int) 8000")
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) 8000, " "encoding-name = (string) \"G729\"")
);
-static gboolean gst_rtpg729pay_setcaps (GstBaseRTPPayload * payload,
- GstCaps * caps);
+static void
+gst_rtp_g729_pay_init (GstRTPG729Pay * pay, GstRTPG729PayClass * klass);
-GST_BOILERPLATE (GstRtpG729Pay, gst_rtpg729pay, GstBaseRTPAudioPayload,
+GST_BOILERPLATE (GstRTPG729Pay, gst_rtp_g729_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
-gst_rtpg729pay_base_init (gpointer klass)
+gst_rtp_g729_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_rtpg729pay_sink_template));
+ gst_static_pad_template_get (&gst_rtp_g729_pay_sink_template));
gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtpg729pay_src_template));
- gst_element_class_set_details (element_class, &gst_rtpg729pay_details);
+ gst_static_pad_template_get (&gst_rtp_g729_pay_src_template));
+ gst_element_class_set_details (element_class, &gst_rtp_g729_pay_details);
}
static void
-gst_rtpg729pay_class_init (GstRtpG729PayClass * klass)
+gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass)
{
- GObjectClass *gobject_class;
- GstElementClass *gstelement_class;
- GstBaseRTPPayloadClass *gstbasertppayload_class;
+ GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
- gobject_class = (GObjectClass *) klass;
- gstelement_class = (GstElementClass *) klass;
- gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
-
- parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
-
- gstbasertppayload_class->set_caps = gst_rtpg729pay_setcaps;
-
- GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
- "G729 audio RTP payloader");
+ payload_class->set_caps = gst_rtp_g729_pay_set_caps;
+ payload_class->handle_buffer = gst_rtp_g729_pay_handle_buffer;
}
static void
-gst_rtpg729pay_init (GstRtpG729Pay * rtpg729pay, GstRtpG729PayClass * klass)
+gst_rtp_g729_pay_init (GstRTPG729Pay * pay, GstRTPG729PayClass * klass)
{
- GstBaseRTPPayload *basertppayload;
- GstBaseRTPAudioPayload *basertpaudiopayload;
+ GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
+ GstBaseRTPAudioPayload *audiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (pay);
- basertppayload = GST_BASE_RTP_PAYLOAD (rtpg729pay);
- basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg729pay);
+ payload->pt = GST_RTP_PAYLOAD_G729;
+ gst_basertppayload_set_options (payload, "audio", FALSE, "G729", 8000);
- /* we don't set the payload type, it should be set by the application using
- * the pt property or the default 96 will be used */
- basertppayload->clock_rate = 8000;
+ gst_base_rtp_audio_payload_set_frame_based (audiopayload);
+ gst_base_rtp_audio_payload_set_frame_options (audiopayload,
+ G729_FRAME_DURATION_MS, G729_FRAME_SIZE);
- /* tell basertpaudiopayload that this is a frame based codec */
- gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
- gst_basertppayload_set_options (basertppayload, "audio", FALSE, "G729", 8000);
- gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 10, 10);
}
static gboolean
-gst_rtpg729pay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
+gst_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
{
- GstRtpG729Pay *rtpg729pay;
- GstBaseRTPAudioPayload *basertpaudiopayload;
- gboolean ret;
GstStructure *structure;
- const char *payload_name;
-
- rtpg729pay = GST_RTP_G729_PAY (basertppayload);
- basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
+ gint pt;
structure = gst_caps_get_structure (caps, 0);
+ if (!gst_structure_get_int (structure, "payload", &pt))
+ pt = GST_RTP_PAYLOAD_G729;
+
+ payload->pt = pt;
+ payload->dynamic = pt != GST_RTP_PAYLOAD_G729;
+
+ gst_basertppayload_set_outcaps (payload, NULL);
+
+ return TRUE;
+}
+
+static GstFlowReturn
+gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
+{
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstBaseRTPAudioPayload *basertpaudiopayload =
+ GST_BASE_RTP_AUDIO_PAYLOAD (payload);
+ GstAdapter *adapter = NULL;
+ guint payload_len;
+ const guint8 *data = NULL;
+ guint available;
+ guint maxptime_octets = G_MAXUINT;
+ guint minptime_octets = 0;
+ guint min_payload_len;
+ guint max_payload_len;
+ gboolean use_adapter = FALSE;
+
+ available = GST_BUFFER_SIZE (buf);
- payload_name = gst_structure_get_name (structure);
- if (g_strcasecmp ("audio/G729", payload_name) != 0)
- goto wrong_name;
+ if (available % G729_FRAME_SIZE != 0 &&
+ available % G729_FRAME_SIZE != G729B_CN_FRAME_SIZE)
+ goto invalid_size;
- ret = gst_basertppayload_set_outcaps (basertppayload, NULL);
+ /* max number of bytes based on given ptime, has to be multiple of
+ * frame_duration */
+ if (payload->max_ptime != -1) {
+ guint ptime_ms = payload->max_ptime / 1000000;
+
+ maxptime_octets = G729_FRAME_SIZE *
+ (int) (ptime_ms / G729_FRAME_DURATION_MS);
+
+ if (maxptime_octets < G729_FRAME_SIZE) {
+ GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %d is smaller than"
+ " minimum %d ns, overwriting to minimum",
+ payload->max_ptime, G729_FRAME_DURATION_MS);
+ maxptime_octets = G729_FRAME_SIZE;
+ }
+ }
+
+ max_payload_len = MIN (
+ /* MTU max */
+ (int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
+ (basertpaudiopayload), 0, 0) / G729_FRAME_SIZE) * G729_FRAME_SIZE,
+ /* ptime max */
+ maxptime_octets);
+
+ /* min number of bytes based on a given ptime, has to be a multiple
+ of frame duration */
+ {
+ guint64 min_ptime;
+
+ g_object_get (G_OBJECT (payload), "min-ptime", &min_ptime, NULL);
+
+ min_ptime = min_ptime / 1000000;
+ minptime_octets = G729_FRAME_SIZE *
+ (int) (min_ptime / G729_FRAME_DURATION_MS);
+ }
+
+ min_payload_len = MAX (minptime_octets, G729_FRAME_SIZE);
+
+ if (min_payload_len > max_payload_len) {
+ min_payload_len = max_payload_len;
+ }
+
+ GST_DEBUG_OBJECT (basertpaudiopayload,
+ "Calculated min_payload_len %u and max_payload_len %u",
+ min_payload_len, max_payload_len);
+
+ adapter = gst_base_rtp_audio_payload_get_adapter (basertpaudiopayload);
+
+ if (adapter && gst_adapter_available (adapter)) {
+ /* If there is always data in the adapter, we have to use it */
+ gst_adapter_push (adapter, buf);
+ available = gst_adapter_available (adapter);
+ use_adapter = TRUE;
+ } else {
+ /* let's set the base timestamp */
+ basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
+
+ /* If buffer fits on an RTP packet, let's just push it through */
+ /* this will check against max_ptime and max_mtu */
+ if (GST_BUFFER_SIZE (buf) >= min_payload_len &&
+ GST_BUFFER_SIZE (buf) <= max_payload_len) {
+ ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
+ GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
+ GST_BUFFER_TIMESTAMP (buf));
+ gst_buffer_unref (buf);
+
+ return ret;
+ }
+
+ available = GST_BUFFER_SIZE (buf);
+ data = (guint8 *) GST_BUFFER_DATA (buf);
+ }
+
+ /* as long as we have full frames */
+ /* this loop will push all available buffers till the last frame */
+ while (available >= min_payload_len ||
+ available % G729_FRAME_SIZE == G729B_CN_FRAME_SIZE) {
+ guint num;
+
+ /* We send as much as we can */
+ if (available <= max_payload_len) {
+ payload_len = available;
+ } else {
+ payload_len = MIN (max_payload_len,
+ (available / G729_FRAME_SIZE) * G729_FRAME_SIZE);
+ }
+
+ if (use_adapter) {
+ data = gst_adapter_peek (adapter, payload_len);
+ }
+
+ ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, data,
+ payload_len, basertpaudiopayload->base_ts);
+
+ num = payload_len / G729_FRAME_SIZE;
+ basertpaudiopayload->base_ts += G729_FRAME_DURATION * num;
+
+ if (use_adapter) {
+ gst_adapter_flush (adapter, payload_len);
+ available = gst_adapter_available (adapter);
+ } else {
+ available -= payload_len;
+ data += payload_len;
+ }
+ }
+
+ if (!use_adapter) {
+ if (available != 0 && adapter) {
+ GstBuffer *buf2;
+ buf2 = gst_buffer_create_sub (buf,
+ GST_BUFFER_SIZE (buf) - available, available);
+ gst_adapter_push (adapter, buf2);
+ } else {
+ gst_buffer_unref (buf);
+ }
+ }
+
+ if (adapter) {
+ g_object_unref (adapter);
+ }
return ret;
/* ERRORS */
-wrong_name:
+invalid_size:
{
- GST_ERROR_OBJECT (rtpg729pay, "wrong name, expected 'audio/G729', got '%s'",
- payload_name);
- return FALSE;
+ GST_ELEMENT_ERROR (payload, STREAM, WRONG_TYPE,
+ ("Invalid input buffer size"),
+ ("Invalid buffer size, should be a multiple of"
+ " G729_FRAME_SIZE(10) with an optional G729B_CN_FRAME_SIZE(2)"
+ " added to it, but it is %u", available));
+ gst_buffer_unref (buf);
+ return GST_FLOW_ERROR;
}
}
projects / platform / upstream / gst-plugins-good.git / commitdiff