diff --git a/ChangeLog b/ChangeLog index a6249870eb..9f81bf7430 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,11 @@ +2006-04-12 Philippe Kalaf + + * gst-libs/gst/rtp/gstrtpbuffer.h: + Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children + * gst-libs/gst/rtp/gstbasertpaudiopayload.c: + * gst-libs/gst/rtp/gstbasertpaudiopayload.h: + New RTP audio base payloader class. Supports frame or sample based codecs + 2006-04-12 Thomas Vander Stichele * configure.ac: diff --git a/gst-libs/gst/rtp/Makefile.am b/gst-libs/gst/rtp/Makefile.am index ea02606c58..d1a7f21e80 100644 --- a/gst-libs/gst/rtp/Makefile.am +++ b/gst-libs/gst/rtp/Makefile.am @@ -2,12 +2,14 @@ libgstrtpincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/rtp libgstrtpinclude_HEADERS = gstrtpbuffer.h \ gstbasertppayload.h \ + gstbasertpaudiopayload.h \ gstbasertpdepayload.h lib_LTLIBRARIES = libgstrtp-@GST_MAJORMINOR@.la libgstrtp_@GST_MAJORMINOR@_la_SOURCES = gstrtpbuffer.c \ gstbasertppayload.c \ + gstbasertpaudiopayload.c \ gstbasertpdepayload.c libgstrtp_@GST_MAJORMINOR@_la_CFLAGS = $(GST_CFLAGS) diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.c b/gst-libs/gst/rtp/gstbasertpaudiopayload.c new file mode 100644 index 0000000000..f8c7a410d5 --- /dev/null +++ b/gst-libs/gst/rtp/gstbasertpaudiopayload.c @@ -0,0 +1,429 @@ +/* GStreamer + * Copyright (C) <2006> Philippe Khalaf + * + * 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 +#include +#include +#include + +#include "gstbasertpaudiopayload.h" + +GST_DEBUG_CATEGORY (basertpaudiopayload_debug); +#define GST_CAT_DEFAULT (basertpaudiopayload_debug) + +/* let us define a minimum of 10 ms for sample based codecs */ +#define GST_RTP_MIN_PTIME_MS 10 + +static void gst_basertpaudiopayload_finalize (GObject * object); + +static GstFlowReturn +gst_basertpaudiopayload_push (GstBaseRTPPayload * basepayload, guint8 * data, + guint payload_len, GstClockTime timestamp); + +static GstFlowReturn gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload * + payload, GstBuffer * buffer); + +static GstFlowReturn +gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload * + basepayload, GstBuffer * buffer); + +static GstFlowReturn +gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload * + basepayload, GstBuffer * buffer); + +GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_basertpaudiopayload, + GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD); + +static void +gst_basertpaudiopayload_base_init (gpointer klass) +{ +} + +static void +gst_basertpaudiopayload_class_init (GstBaseRTPAudioPayloadClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseRTPPayloadClass *gstbasertppayload_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; + gobject_class->finalize = gst_basertpaudiopayload_finalize; + + parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD); + + gstbasertppayload_class->handle_buffer = + gst_basertpaudiopayload_handle_buffer; + + GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0, + "base audio RTP payloader"); +} + +static void +gst_basertpaudiopayload_init (GstBaseRTPAudioPayload * basertpaudiopayload, + GstBaseRTPAudioPayloadClass * klass) +{ + basertpaudiopayload->adapter = gst_adapter_new (); + basertpaudiopayload->adapter_base_ts = 0; + + basertpaudiopayload->type = AUDIO_CODEC_TYPE_NONE; + + /* these need to be set by child object if frame based */ + basertpaudiopayload->frame_size = 0; + basertpaudiopayload->frame_duration = 0; + + /* these need to be set by child object if sample based */ + basertpaudiopayload->sample_size = 0; +} + +static void +gst_basertpaudiopayload_finalize (GObject * object) +{ + GstBaseRTPAudioPayload *basertpaudiopayload; + + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (object); + g_object_unref (basertpaudiopayload->adapter); + basertpaudiopayload->adapter = NULL; + + GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object)); +} + +void +gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload * + basertpaudiopayload) +{ + g_return_if_fail (basertpaudiopayload != NULL); + + if (basertpaudiopayload->type != AUDIO_CODEC_TYPE_NONE) { + GST_ERROR_OBJECT (basertpaudiopayload, + "Codec type already set! You should only set this once!"); + } + basertpaudiopayload->type = AUDIO_CODEC_TYPE_FRAME_BASED; +} + +void +gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload * + basertpaudiopayload) +{ + g_return_if_fail (basertpaudiopayload != NULL); + + if (basertpaudiopayload->type != AUDIO_CODEC_TYPE_NONE) { + GST_ERROR_OBJECT (basertpaudiopayload, + "Codec type already set! You should only set this once!"); + } + basertpaudiopayload->type = AUDIO_CODEC_TYPE_SAMPLE_BASED; +} + +/* These are options that need to be set for frame based audio codecs */ +void +gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload + * basertpaudiopayload, gint frame_duration, gint frame_size) +{ + g_return_if_fail (basertpaudiopayload != NULL); + + basertpaudiopayload->frame_size = frame_size; + basertpaudiopayload->frame_duration = frame_duration; +} + +void +gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload + * basertpaudiopayload, gint sample_size) +{ + g_return_if_fail (basertpaudiopayload != NULL); + + basertpaudiopayload->sample_size = sample_size; +} + +static GstFlowReturn +gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload * basepayload, + GstBuffer * buffer) +{ + GstFlowReturn ret; + GstBaseRTPAudioPayload *basertpaudiopayload; + + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); + + ret = GST_FLOW_ERROR; + + if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_FRAME_BASED) { + ret = gst_basertpaudiopayload_handle_frame_based_buffer (basepayload, + buffer); + } else if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) { + ret = gst_basertpaudiopayload_handle_sample_based_buffer (basepayload, + buffer); + } else { + GST_DEBUG_OBJECT (basertpaudiopayload, "Audio codec type not set"); + } + + return ret; +} + +/* this assumes all frames have a constant duration and a constant size */ +static GstFlowReturn +gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload * + basepayload, GstBuffer * buffer) +{ + GstBaseRTPAudioPayload *basertpaudiopayload; + guint payload_len; + guint8 *data; + GstFlowReturn ret; + guint available; + gint frame_size, frame_duration; + + guint maxptime_octets = G_MAXUINT; + + ret = GST_FLOW_ERROR; + + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); + + if (basertpaudiopayload->frame_size == 0 || + basertpaudiopayload->frame_duration == 0) { + GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set"); + gst_buffer_unref (buffer); + return GST_FLOW_ERROR; + } + frame_size = basertpaudiopayload->frame_size; + frame_duration = basertpaudiopayload->frame_duration; + + /* If buffer fits on an RTP packet, let's just push it through without using + * the adapter */ + /* this will check again max_ptime and max_mtu */ + if (!gst_basertppayload_is_filled (basepayload, + gst_rtp_buffer_calc_packet_len (GST_BUFFER_SIZE (buffer), 0, 0), + GST_BUFFER_DURATION (buffer))) { + ret = gst_basertpaudiopayload_push (basepayload, GST_BUFFER_DATA (buffer), + GST_BUFFER_SIZE (buffer), GST_BUFFER_TIMESTAMP (buffer)); + gst_buffer_unref (buffer); + + return ret; + } + + /* TODO : would be nice if we had some property that told the payloader to put + * just 1 frame per RTP packet, for the moment we can set the ptime to 0 or + * something smaller or equal to a frame duration */ + + /* max number of bytes based on given ptime, has to be multiple of + * frame_duration */ + if (basepayload->max_ptime != -1) { + guint ptime_ms = basepayload->max_ptime / 1000000; + + maxptime_octets = frame_size * (int) (ptime_ms / frame_duration); + if (maxptime_octets == 0) { + GST_WARNING_OBJECT (basertpaudiopayload, + "Given ptime %d is smaller than minimum %d ms, overwriting to minimum", + ptime_ms, frame_duration); + maxptime_octets = frame_size; + } + } + + /* if the adapter is empty (should be), let's set the base timestamp */ + if (gst_adapter_available (basertpaudiopayload->adapter) == 0) { + basertpaudiopayload->adapter_base_ts = GST_BUFFER_TIMESTAMP (buffer); + } else { + GST_ERROR_OBJECT (basertpaudiopayload, + "Adapter should be empty but is not!"); + return GST_FLOW_ERROR; + } + + gst_adapter_push (basertpaudiopayload->adapter, buffer); + + available = gst_adapter_available (basertpaudiopayload->adapter); + + /* as long as we have full frames */ + /* this loop will always empty the adapter till the last frame */ + /* TODO Make it possible to set a minimum size per packet, this way the + * algorithm doesn't empty the adapter if there is too little data left and + * will wait until the next buffers to arrive */ + while (available >= frame_size) { + /* we need to see how many frames we can get based on maximum MTU, maximum + * ptime and the number of bytes available in the adapter */ + payload_len = MIN (MIN ( + /* MTU max */ + (int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU + (basertpaudiopayload), 0, 0) / frame_size) * frame_size, + /* ptime max */ + maxptime_octets), + /* currently available */ + floor (available / frame_size) * frame_size); + + data = + (guint8 *) gst_adapter_peek (basertpaudiopayload->adapter, payload_len); + ret = + gst_basertpaudiopayload_push (basepayload, data, payload_len, + basertpaudiopayload->adapter_base_ts); + + gst_adapter_flush (basertpaudiopayload->adapter, payload_len); + gfloat ts_inc = (payload_len * frame_duration) / frame_size; + + ts_inc = ts_inc * GST_MSECOND; + basertpaudiopayload->adapter_base_ts += ts_inc; + GST_DEBUG_OBJECT (basertpaudiopayload, "%f %f %d", ts_inc, + ts_inc * GST_MSECOND, (payload_len * frame_duration) / frame_size); + GST_DEBUG_OBJECT (basertpaudiopayload, "Pushing with ts %" GST_TIME_FORMAT, + GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts)); + + available = gst_adapter_available (basertpaudiopayload->adapter); + } + + /* adapter should be freed by now */ + if (available != 0) { + GST_ERROR_OBJECT (basertpaudiopayload, + "Adapter should be empty but is not!"); + return GST_FLOW_ERROR; + } + + return ret; +} + +static GstFlowReturn +gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload * + basepayload, GstBuffer * buffer) +{ + GstBaseRTPAudioPayload *basertpaudiopayload; + guint payload_len; + guint8 *data; + GstFlowReturn ret; + guint available; + + guint maxptime_octets = G_MAXUINT; + + guint minptime_octets = 0; + guint sample_size; + + ret = GST_FLOW_ERROR; + + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); + + if (basertpaudiopayload->sample_size == 0) { + GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set"); + gst_buffer_unref (buffer); + return GST_FLOW_ERROR; + } + sample_size = basertpaudiopayload->sample_size; + + /* If buffer fits on an RTP packet, let's just push it through without using + * the adapter */ + /* this will check again max_ptime and max_mtu */ + if (!gst_basertppayload_is_filled (basepayload, + gst_rtp_buffer_calc_packet_len (GST_BUFFER_SIZE (buffer), 0, 0), + GST_BUFFER_DURATION (buffer))) { + ret = gst_basertpaudiopayload_push (basepayload, GST_BUFFER_DATA (buffer), + GST_BUFFER_SIZE (buffer), GST_BUFFER_TIMESTAMP (buffer)); + gst_buffer_unref (buffer); + + return ret; + } + + /* max number of bytes based on given ptime */ + if (basepayload->max_ptime != -1) { + maxptime_octets = basepayload->max_ptime * basepayload->clock_rate / + (sample_size * GST_SECOND); + minptime_octets = GST_RTP_MIN_PTIME_MS * basepayload->clock_rate / + (sample_size * 1000); + GST_DEBUG_OBJECT (basertpaudiopayload, + "Calculated max_octects %u and min_octets %u", maxptime_octets, + minptime_octets); + if (maxptime_octets < minptime_octets) { + GST_WARNING_OBJECT (basertpaudiopayload, + "Given ptime %d is smaller than minimum %d, replacing by %d", + maxptime_octets, minptime_octets, minptime_octets); + maxptime_octets = minptime_octets; + } + } + + /* if the adapter is empty (should be), let's set the base timestamp */ + if (gst_adapter_available (basertpaudiopayload->adapter) == 0) { + basertpaudiopayload->adapter_base_ts = GST_BUFFER_TIMESTAMP (buffer); + GST_DEBUG_OBJECT (basertpaudiopayload, "Setting to %" GST_TIME_FORMAT, + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer))); + } + + gst_adapter_push (basertpaudiopayload->adapter, buffer); + + available = gst_adapter_available (basertpaudiopayload->adapter); + + /* as long as we have full frames */ + /* this loop will always empty the adapter till the last frame */ + /* TODO Make it possible to set a minimum size per packet, this way the + * algorithm doesn't empty the adapter if there is too little data left and + * will wait until the next buffers to arrive */ + while (available >= minptime_octets) { + /* we need to see how many frames we can get based on maximum MTU, maximum + * ptime and the number of bytes available in the adapter */ + payload_len = MIN (MIN ( + /* MTU max */ + gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU + (basertpaudiopayload), 0, 0), + /* ptime max */ + maxptime_octets), + /* currently available */ + available); + + data = + (guint8 *) gst_adapter_peek (basertpaudiopayload->adapter, payload_len); + GST_DEBUG_OBJECT (basertpaudiopayload, "Pushing with ts %" GST_TIME_FORMAT, + GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts)); + ret = + gst_basertpaudiopayload_push (basepayload, data, payload_len, + basertpaudiopayload->adapter_base_ts); + + gst_adapter_flush (basertpaudiopayload->adapter, payload_len); + gfloat num = payload_len; + gfloat datarate = (sample_size * basepayload->clock_rate); + + basertpaudiopayload->adapter_base_ts += + /* payload_len (bytes) * nsecs/sec / datarate (bytes*sec) */ + num / datarate * GST_SECOND; + GST_DEBUG_OBJECT (basertpaudiopayload, "Calculating ts inc %f %f %f", num, + datarate, num / datarate * GST_SECOND); + GST_DEBUG_OBJECT (basertpaudiopayload, "New ts is %" GST_TIME_FORMAT, + GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts)); + + available = gst_adapter_available (basertpaudiopayload->adapter); + } + + return ret; +} + +static GstFlowReturn +gst_basertpaudiopayload_push (GstBaseRTPPayload * basepayload, guint8 * data, + guint payload_len, GstClockTime timestamp) +{ + GstBuffer *outbuf; + guint8 *payload; + GstFlowReturn ret; + + /* create buffer to hold the payload */ + outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); + + /* copy payload */ + gst_rtp_buffer_set_payload_type (outbuf, basepayload->pt); + payload = gst_rtp_buffer_get_payload (outbuf); + memcpy (payload, data, payload_len); + + GST_BUFFER_TIMESTAMP (outbuf) = timestamp; + ret = gst_basertppayload_push (basepayload, outbuf); + + return ret; +} diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.h b/gst-libs/gst/rtp/gstbasertpaudiopayload.h new file mode 100644 index 0000000000..e2892e7bb0 --- /dev/null +++ b/gst-libs/gst/rtp/gstbasertpaudiopayload.h @@ -0,0 +1,88 @@ +/* GStreamer + * Copyright (C) <2006> Philippe Khalaf + * + * 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. + */ + +#ifndef __GST_BASE_RTP_AUDIO_PAYLOAD_H__ +#define __GST_BASE_RTP_AUDIO_PAYLOAD_H__ + +#include +#include +#include + +G_BEGIN_DECLS + +typedef struct _GstBaseRTPAudioPayload GstBaseRTPAudioPayload; +typedef struct _GstBaseRTPAudioPayloadClass GstBaseRTPAudioPayloadClass; + +#define GST_TYPE_BASE_RTP_AUDIO_PAYLOAD \ + (gst_basertpaudiopayload_get_type()) +#define GST_BASE_RTP_AUDIO_PAYLOAD(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload)) +#define GST_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload)) +#define GST_IS_BASE_RTP_AUDIO_PAYLOAD(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD)) +#define GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS(obj) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD)) + +typedef enum { + AUDIO_CODEC_TYPE_NONE, + AUDIO_CODEC_TYPE_FRAME_BASED, + AUDIO_CODEC_TYPE_SAMPLE_BASED +} AudioCodecType; + +struct _GstBaseRTPAudioPayload +{ + GstBaseRTPPayload payload; + + GstClockTime adapter_base_ts; + GstAdapter *adapter; + gint frame_size; + gint frame_duration; + + gint sample_size; + + AudioCodecType type; +}; + +struct _GstBaseRTPAudioPayloadClass +{ + GstBaseRTPPayloadClass parent_class; +}; + +gboolean gst_basertpaudiopayload_plugin_init (GstPlugin * plugin); + +GType gst_basertpaudiopayload_get_type (void); + +void +gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload); + +void +gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload); + +void +gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload + *basertpaudiopayload, gint frame_duration, gint frame_size); + +void +gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload + *basertpaudiopayload, gint sample_size); + +G_END_DECLS + +#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */ diff --git a/gst-libs/gst/rtp/gstrtpbuffer.h b/gst-libs/gst/rtp/gstrtpbuffer.h index 9fcfbcd626..32ed24a670 100644 --- a/gst-libs/gst/rtp/gstrtpbuffer.h +++ b/gst-libs/gst/rtp/gstrtpbuffer.h @@ -71,6 +71,8 @@ typedef enum #define GST_RTP_PAYLOAD_MPV_STRING "32" #define GST_RTP_PAYLOAD_H263_STRING "34" +#define GST_RTP_PAYLOAD_DYNAMIC_STRING "[96, 127]" + /* creating buffers */ GstBuffer* gst_rtp_buffer_new (void); void gst_rtp_buffer_allocate_data (GstBuffer *buffer, guint payload_len,