gstreamer/ext/faac/gstfaac.c

/* GStreamer FAAC (Free AAC Encoder) plugin
 * Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
 *
 * 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 "gstfaac.h"

GST_PAD_TEMPLATE_FACTORY (src_template,
  "src",
  GST_PAD_SRC,
  GST_PAD_ALWAYS,
  GST_CAPS_NEW (
    "faac_mpeg_templ",
    "audio/mpeg",
      "systemstream", GST_PROPS_BOOLEAN (FALSE),
      "mpegversion",  GST_PROPS_LIST (
                        GST_PROPS_INT (4), /* we prefer 4 */
                        GST_PROPS_INT (2)
                      ),
      "channels",     GST_PROPS_INT_RANGE (1, 6),
      "samplerate",   GST_PROPS_INT_RANGE (8000, 96000)
  )
);

GST_PAD_TEMPLATE_FACTORY (sink_template,
  "sink",
  GST_PAD_SINK,
  GST_PAD_ALWAYS,
  GST_CAPS_NEW (
    "faac_int16_templ",
    "audio/x-raw-int",
      "endianness", GST_PROPS_INT (G_BYTE_ORDER),
      "signed",     GST_PROPS_BOOLEAN (TRUE),
      "width",      GST_PROPS_INT (16),
      "depth",      GST_PROPS_INT (16),
      "rate",       GST_PROPS_INT_RANGE (8000, 96000),
      "channels",   GST_PROPS_INT_RANGE (1, 6)
  ),
  GST_CAPS_NEW (
    "faac_int24_templ",
    "audio/x-raw-int",
      "endianness", GST_PROPS_INT (G_BYTE_ORDER),
      "signed",     GST_PROPS_BOOLEAN (TRUE),
      "width",      GST_PROPS_INT (32),
      "depth",      GST_PROPS_INT (24),
      "rate",       GST_PROPS_INT_RANGE (8000, 96000),
      "channels",   GST_PROPS_INT_RANGE (1, 6)
  ),
  GST_CAPS_NEW (
    "faac_float_templ",
    "audio/x-raw-float",
      "endianness", GST_PROPS_INT (G_BYTE_ORDER),
      "depth",      GST_PROPS_INT (32), /* float  */
      "rate",       GST_PROPS_INT_RANGE (8000, 96000),
      "channels",   GST_PROPS_INT_RANGE (1, 6)
  )
);

enum {
  ARG_0,
  ARG_BITRATE,
  ARG_PROFILE,
  ARG_TNS,
  ARG_MIDSIDE,
  ARG_SHORTCTL
  /* FILL ME */
};

static void     gst_faac_base_init    (GstFaacClass *klass);
static void     gst_faac_class_init   (GstFaacClass *klass);
static void     gst_faac_init         (GstFaac      *faac);

static void     gst_faac_set_property (GObject      *object,
				       guint         prop_id, 
				       const GValue *value,
				       GParamSpec   *pspec);
static void     gst_faac_get_property (GObject      *object,
				       guint         prop_id, 
				       GValue       *value,
				       GParamSpec   *pspec);

static GstPadLinkReturn
                gst_faac_sinkconnect  (GstPad       *pad,
				       GstCaps      *caps);
static GstPadLinkReturn
                gst_faac_srcconnect   (GstPad       *pad,
				       GstCaps      *caps);
static void     gst_faac_chain        (GstPad       *pad,
				       GstData      *data);
static GstElementStateReturn
                gst_faac_change_state (GstElement   *element);

static GstElementClass *parent_class = NULL;
/* static guint gst_faac_signals[LAST_SIGNAL] = { 0 }; */

GType
gst_faac_get_type (void)
{
  static GType gst_faac_type = 0;

  if (!gst_faac_type) {
    static const GTypeInfo gst_faac_info = {
      sizeof (GstFaacClass),      
      (GBaseInitFunc) gst_faac_base_init,
      NULL,
      (GClassInitFunc) gst_faac_class_init,
      NULL,
      NULL,
      sizeof(GstFaac),
      0,
      (GInstanceInitFunc) gst_faac_init,
    };

    gst_faac_type = g_type_register_static (GST_TYPE_ELEMENT,
					    "GstFaac",
					    &gst_faac_info, 0);
  }

  return gst_faac_type;
}

static void
gst_faac_base_init (GstFaacClass *klass)
{
  GstElementDetails gst_faac_details = {
    "Free AAC Encoder (FAAC)",
    "Codec/Audio/Encoder",
    "Free MPEG-2/4 AAC encoder",
    "Ronald Bultje <rbultje@ronald.bitfreak.net>",
  };
  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

  gst_element_class_add_pad_template (element_class,
	GST_PAD_TEMPLATE_GET (src_template));
  gst_element_class_add_pad_template (element_class,
	GST_PAD_TEMPLATE_GET (sink_template));

  gst_element_class_set_details (element_class, &gst_faac_details);
}

#define GST_TYPE_FAAC_PROFILE (gst_faac_profile_get_type ())
static GType
gst_faac_profile_get_type (void)
{
  static GType gst_faac_profile_type = 0;

  if (!gst_faac_profile_type) {
    static GEnumValue gst_faac_profile[] = {
      { MAIN, "MAIN", "Main profile"                   },
      { LOW,  "LOW",  "Low complexity profile"         },
      { SSR,  "SSR",  "Scalable sampling rate profile" },
      { LTP,  "LTP",  "Long term prediction profile"   },
      { 0, NULL, NULL },
    };

    gst_faac_profile_type = g_enum_register_static ("GstFaacProfile",
						    gst_faac_profile);
  }

  return gst_faac_profile_type;
}

#define GST_TYPE_FAAC_SHORTCTL (gst_faac_shortctl_get_type ())
static GType
gst_faac_shortctl_get_type (void)
{
  static GType gst_faac_shortctl_type = 0;

  if (!gst_faac_shortctl_type) {
    static GEnumValue gst_faac_shortctl[] = {
      { SHORTCTL_NORMAL,  "SHORTCTL_NORMAL",  "Normal block type" },
      { SHORTCTL_NOSHORT, "SHORTCTL_NOSHORT", "No short blocks"   },
      { SHORTCTL_NOLONG,  "SHORTCTL_NOLONG",  "No long blocks"    },
      { 0, NULL, NULL },
    };

    gst_faac_shortctl_type = g_enum_register_static ("GstFaacShortCtl",
						    gst_faac_shortctl);
  }

  return gst_faac_shortctl_type;
}

static void
gst_faac_class_init (GstFaacClass *klass)
{
  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);

  parent_class = g_type_class_ref (GST_TYPE_ELEMENT);

  /* properties */
  g_object_class_install_property (gobject_class, ARG_BITRATE,
    g_param_spec_int ("bitrate", "Bitrate (bps)", "Bitrate in bits/sec",
                      8 * 1024, 320 * 1024, 128 * 1024, G_PARAM_READWRITE));
  g_object_class_install_property (gobject_class, ARG_PROFILE,
    g_param_spec_enum ("profile", "Profile", "MPEG/AAC encoding profile",
                       GST_TYPE_FAAC_PROFILE, MAIN, G_PARAM_READWRITE));
  g_object_class_install_property (gobject_class, ARG_TNS,
    g_param_spec_boolean ("tns", "TNS", "Use temporal noise shaping",
                          FALSE, G_PARAM_READWRITE));
  g_object_class_install_property (gobject_class, ARG_MIDSIDE,
    g_param_spec_boolean ("midside", "Midside", "Allow mid/side encoding",
                          TRUE, G_PARAM_READWRITE));
  g_object_class_install_property (gobject_class, ARG_SHORTCTL,
    g_param_spec_enum ("shortctl", "Block type",
		       "Block type encorcing",
                       GST_TYPE_FAAC_SHORTCTL, MAIN, G_PARAM_READWRITE));

  /* virtual functions */
  gstelement_class->change_state = gst_faac_change_state;

  gobject_class->set_property = gst_faac_set_property;
  gobject_class->get_property = gst_faac_get_property;
}

static void
gst_faac_init (GstFaac *faac)
{
  faac->handle = NULL;
  faac->samplerate = -1;
  faac->channels = -1;
  faac->cache = NULL;
  faac->cache_time = GST_CLOCK_TIME_NONE;
  faac->cache_duration = 0;

  GST_FLAG_SET (faac, GST_ELEMENT_EVENT_AWARE);

  faac->sinkpad = gst_pad_new_from_template (
	GST_PAD_TEMPLATE_GET (sink_template), "sink");
  gst_element_add_pad (GST_ELEMENT (faac), faac->sinkpad);
  gst_pad_set_chain_function (faac->sinkpad, gst_faac_chain);
  gst_pad_set_link_function (faac->sinkpad, gst_faac_sinkconnect);

  faac->srcpad = gst_pad_new_from_template (
	GST_PAD_TEMPLATE_GET (src_template), "src");
  gst_element_add_pad (GST_ELEMENT (faac), faac->srcpad);
  gst_pad_set_link_function (faac->srcpad, gst_faac_srcconnect);

  /* default properties */
  faac->bitrate = 1024 * 128;
  faac->profile = MAIN;
  faac->shortctl = SHORTCTL_NORMAL;
  faac->tns = FALSE;
  faac->midside = TRUE;
}

static GstPadLinkReturn
gst_faac_sinkconnect (GstPad  *pad,
		      GstCaps *caps)
{
  GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));

  if (!GST_CAPS_IS_FIXED (caps))
    return GST_PAD_LINK_DELAYED;

  if (faac->handle) {
    faacEncClose (faac->handle);
    faac->handle = NULL;
  }
  if (faac->cache) {
    gst_buffer_unref (faac->cache);
    faac->cache = NULL;
  }

  for (; caps != NULL; caps = caps->next) {
    faacEncHandle *handle;
    gint channels, samplerate, depth;
    gulong samples, bytes, fmt = 0, bps = 0;

    gst_caps_get (caps, "channels", &channels,
			"rate",     &samplerate,
			"depth",    &depth, NULL);

    /* open a new handle to the encoder */
    if (!(handle = faacEncOpen (samplerate, channels,
				&samples, &bytes)))
      continue;

    switch (depth) {
      case 16:
        fmt = FAAC_INPUT_16BIT;
        bps = 2;
        break;
      case 24:
        fmt = FAAC_INPUT_32BIT; /* 24-in-32, actually */
        bps = 4;
        break;
      case 32:
        fmt = FAAC_INPUT_FLOAT; /* see template, this is right */
        bps = 4;
        break;
    }

    if (!fmt) {
      faacEncClose (handle);
      continue;
    }

    faac->format = fmt;
    faac->bps = bps;
    faac->handle = handle;
    faac->bytes = bytes;
    faac->samples = samples;
    faac->channels = channels;
    faac->samplerate = samplerate;

    /* if the other side was already set-up, redo that */
    if (GST_PAD_CAPS (faac->srcpad))
      return gst_faac_srcconnect (faac->srcpad,
				  gst_pad_get_allowed_caps (faac->srcpad));

    /* else, that'll be done later */
    return GST_PAD_LINK_OK;
  }

  return GST_PAD_LINK_REFUSED;
}

static GstPadLinkReturn
gst_faac_srcconnect (GstPad  *pad,
		     GstCaps *caps)
{
  GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
  GstCaps *t;

  if (!faac->handle ||
      (faac->samplerate == -1 || faac->channels == -1)) {
    return GST_PAD_LINK_DELAYED;
  }

  /* we do samplerate/channels ourselves */
  for (t = caps; t != NULL; t = t->next) {
    gst_props_remove_entry_by_name (t->properties, "rate");
    gst_props_remove_entry_by_name (t->properties, "channels");
  }

  /* go through list */
  caps = gst_caps_normalize (caps);
  for ( ; caps != NULL; caps = caps->next) {
    faacEncConfiguration *conf;
    gint mpegversion = 0;
    GstCaps *newcaps;
    GstPadLinkReturn ret;

    gst_caps_get_int (caps, "mpegversion", &mpegversion);

    /* new conf */
    conf = faacEncGetCurrentConfiguration (faac->handle);
    conf->mpegVersion = (mpegversion == 4) ? MPEG4 : MPEG2;
    conf->aacObjectType = faac->profile;
    conf->allowMidside = faac->midside;
    conf->useLfe = 0;
    conf->useTns = faac->tns;
    conf->bitRate = faac->bitrate;
    conf->inputFormat = faac->format;

    /* FIXME: this one here means that we do not support direct
     * "MPEG audio file" output (like mp3). This means we can
     * only mux this into mov/qt (mp4a) or matroska or so. If
     * we want to support direct AAC file output, we need ADTS
     * headers, and we need to find a way in the caps to detect
     * that (that the next element is filesink or any element
     * that does want ADTS headers). */

    conf->outputFormat = 0; /* raw, no ADTS headers */
    conf->shortctl = faac->shortctl;
    if (!faacEncSetConfiguration (faac->handle, conf)) {
      GST_WARNING ("Faac doesn't support the current conf");
      continue;
    }

    newcaps = GST_CAPS_NEW ("faac_mpeg_caps",
			    "audio/mpeg",
			      "systemstream", GST_PROPS_BOOLEAN (FALSE),
			      "mpegversion",  GST_PROPS_INT (mpegversion),
			      "channels",     GST_PROPS_INT (faac->channels),
			      "rate",         GST_PROPS_INT (faac->samplerate));
    ret = gst_pad_try_set_caps (faac->srcpad, newcaps);

    switch (ret) {
      case GST_PAD_LINK_OK:
      case GST_PAD_LINK_DONE:
        return GST_PAD_LINK_DONE;
      case GST_PAD_LINK_DELAYED:
        return GST_PAD_LINK_DELAYED;
      default:
        break;
    }
  }

  return GST_PAD_LINK_REFUSED;
}

static void
gst_faac_chain (GstPad  *pad,
		GstData *data)
{
  GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
  GstBuffer *inbuf, *outbuf, *subbuf;
  guint size, ret_size, in_size, frame_size;

  if (GST_IS_EVENT (data)) {
    GstEvent *event = GST_EVENT (data);

    switch (GST_EVENT_TYPE (event)) {
      case GST_EVENT_EOS:
        /* flush first */
        while (1) {
          outbuf = gst_buffer_new_and_alloc (faac->bytes);
          if ((ret_size = faacEncEncode (faac->handle,
					 NULL, 0,
					 GST_BUFFER_DATA (outbuf),
					 faac->bytes)) < 0) {
            gst_element_error (GST_ELEMENT (faac), "Error during AAC encoding");
            gst_event_unref (event);
            gst_buffer_unref (outbuf);
            return;
          }

          if (ret_size > 0) {
            GST_BUFFER_SIZE (outbuf) = ret_size;
            GST_BUFFER_TIMESTAMP (outbuf) = 0;
            GST_BUFFER_DURATION (outbuf) = 0;
            gst_pad_push (faac->srcpad, GST_DATA (outbuf));
          } else {
            break;
          }
        }

        gst_element_set_eos (GST_ELEMENT (faac));
        gst_pad_push (faac->srcpad, data);
        return;
      default:
	gst_pad_event_default (pad, event);
        return;
    }
  }

  inbuf = GST_BUFFER (data);

  if (!faac->handle) {
    gst_element_error (GST_ELEMENT (faac),
		       "No input format negotiated");
    gst_buffer_unref (inbuf);
    return;
  }

  if (!GST_PAD_CAPS (faac->srcpad)) {
    if (gst_faac_srcconnect (faac->srcpad,
			     gst_pad_get_allowed_caps (faac->srcpad)) <= 0) {
      gst_element_error (GST_ELEMENT (faac),
			 "Failed to negotiate MPEG/AAC format with next element");
      gst_buffer_unref (inbuf);
      return;
    }
  }

  size = GST_BUFFER_SIZE (inbuf);
  in_size = size;
  if (faac->cache)
    in_size += GST_BUFFER_SIZE (faac->cache);
  frame_size = faac->samples * faac->bps;

  while (1) {
    /* do we have enough data for one frame? */
    if (in_size / faac->bps < faac->samples) {
      if (in_size > size) {
        /* this is panic! we got a buffer, but still don't have enough
         * data. Merge them and retry in the next cycle... */
        faac->cache = gst_buffer_merge (faac->cache, inbuf);
      } else if (in_size == size) {
        /* this shouldn't happen, but still... */
        faac->cache = inbuf;
      } else if (in_size > 0) {
        faac->cache = gst_buffer_create_sub (inbuf, size - in_size,
					     in_size);
        GST_BUFFER_DURATION (faac->cache) =
	  GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (faac->cache) / size;
        GST_BUFFER_TIMESTAMP (faac->cache) =
	  GST_BUFFER_TIMESTAMP (inbuf) + (GST_BUFFER_DURATION (inbuf) *
	    (size - in_size) / size);
        gst_buffer_unref (inbuf);
      } else {
        gst_buffer_unref (inbuf);
      }
          
      return;
    }

    /* create the frame */
    if (in_size > size) {
      /* merge */
      subbuf = gst_buffer_create_sub (inbuf, 0, frame_size - (in_size - size));
      GST_BUFFER_DURATION (subbuf) =
	GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (subbuf) / size;
      subbuf = gst_buffer_merge (faac->cache, subbuf);
      faac->cache = NULL;
    } else {
      subbuf = gst_buffer_create_sub (inbuf, size - in_size, frame_size);
      GST_BUFFER_DURATION (subbuf) =
	GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (subbuf) / size;
      GST_BUFFER_TIMESTAMP (subbuf) =
	GST_BUFFER_TIMESTAMP (inbuf) + (GST_BUFFER_DURATION (inbuf) *
	  (size - in_size) / size);
    }

    outbuf = gst_buffer_new_and_alloc (faac->bytes);
    if ((ret_size = faacEncEncode (faac->handle,
				   (gint32 *) GST_BUFFER_DATA (subbuf),
				   GST_BUFFER_SIZE (subbuf) / faac->bps,
				   GST_BUFFER_DATA (outbuf),
				   faac->bytes)) < 0) {
      gst_element_error (GST_ELEMENT (faac), "Error during AAC encoding");
      gst_buffer_unref (inbuf);
      gst_buffer_unref (subbuf);
      return;
    }

    if (ret_size > 0) {
      GST_BUFFER_SIZE (outbuf) = ret_size;
      if (faac->cache_time != GST_CLOCK_TIME_NONE) {
        GST_BUFFER_TIMESTAMP (outbuf) = faac->cache_time;
        faac->cache_time = GST_CLOCK_TIME_NONE;
      } else
        GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (subbuf);
      GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (subbuf);
      if (faac->cache_duration) {
        GST_BUFFER_DURATION (outbuf) += faac->cache_duration;
        faac->cache_duration = 0;
      }
      gst_pad_push (faac->srcpad, GST_DATA (outbuf));
    } else {
      /* FIXME: what I'm doing here isn't fully correct, but there
       * really isn't a better way yet.
       * Problem is that libfaac caches buffers (for encoding
       * purposes), so the timestamp of the outgoing buffer isn't
       * the same as the timestamp of the data that I pushed in.
       * However, I don't know the delay between those two so I
       * cannot really say aything about it. This is a bad guess. */

      gst_buffer_unref (outbuf);
      if (faac->cache_time != GST_CLOCK_TIME_NONE)
        faac->cache_time = GST_BUFFER_TIMESTAMP (subbuf);
      faac->cache_duration += GST_BUFFER_DURATION (subbuf);
    }

    in_size -= frame_size;
    gst_buffer_unref (subbuf);
  }
}

static void
gst_faac_set_property (GObject      *object,
		       guint         prop_id, 
		       const GValue *value,
		       GParamSpec   *pspec)
{
  GstFaac *faac = GST_FAAC (object);

  switch (prop_id) {
    case ARG_BITRATE:
      faac->bitrate = g_value_get_int (value);
      break;
    case ARG_PROFILE:
      faac->profile = g_value_get_enum (value);
      break;
    case ARG_TNS:
      faac->tns = g_value_get_boolean (value);
      break;
    case ARG_MIDSIDE:
      faac->midside = g_value_get_boolean (value);
      break;
    case ARG_SHORTCTL:
      faac->shortctl = g_value_get_enum (value);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_faac_get_property (GObject    *object,
		       guint       prop_id, 
		       GValue     *value,
		       GParamSpec *pspec)
{
  GstFaac *faac = GST_FAAC (object);

  switch (prop_id) {
    case ARG_BITRATE:
      g_value_set_int (value, faac->bitrate);
      break;
    case ARG_PROFILE:
      g_value_set_enum (value, faac->profile);
      break;
    case ARG_TNS:
      g_value_set_boolean (value, faac->tns);
      break;
    case ARG_MIDSIDE:
      g_value_set_boolean (value, faac->midside);
      break;
    case ARG_SHORTCTL:
      g_value_set_enum (value, faac->shortctl);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static GstElementStateReturn
gst_faac_change_state (GstElement *element)
{
  GstFaac *faac = GST_FAAC (element);

  switch (GST_STATE_TRANSITION (element)) {
    case GST_STATE_PAUSED_TO_READY:
      if (faac->handle) {
        faacEncClose (faac->handle);
        faac->handle = NULL;
      }
      if (faac->cache) {
        gst_buffer_unref (faac->cache);
        faac->cache = NULL;
      }
      faac->cache_time = GST_CLOCK_TIME_NONE;
      faac->cache_duration = 0;
      faac->samplerate = -1;
      faac->channels = -1;
      break;
    default:
      break;
  }

  if (GST_ELEMENT_CLASS (parent_class)->change_state)
    return GST_ELEMENT_CLASS (parent_class)->change_state (element);

  return GST_STATE_SUCCESS;
}

static gboolean
plugin_init (GstPlugin *plugin)
{
  return gst_element_register (plugin, "faac",
			       GST_RANK_NONE,
			       GST_TYPE_FAAC);
}

GST_PLUGIN_DEFINE (
  GST_VERSION_MAJOR,
  GST_VERSION_MINOR,
  "faac",
  "Free AAC Encoder (FAAC)",
  plugin_init,
  VERSION,
  "LGPL",
  GST_PACKAGE,
  GST_ORIGIN
)