gstreamer/ext/gsm/gstgsmdec.c

/*
 * Farsight
 * GStreamer GSM encoder
 * Copyright (C) 2005 Philippe Khalaf <burger@speedy.org>
 *
 * 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 "gstgsmdec.h"

GST_DEBUG_CATEGORY_STATIC (gsmdec_debug);
#define GST_CAT_DEFAULT (gsmdec_debug)

/* GSMDec signals and args */
enum
{
  /* FILL ME */
  LAST_SIGNAL
};

enum
{
  /* FILL ME */
  ARG_0
};

static gboolean gst_gsmdec_start (GstAudioDecoder * dec);
static gboolean gst_gsmdec_stop (GstAudioDecoder * dec);
static gboolean gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
static GstFlowReturn gst_gsmdec_parse (GstAudioDecoder * dec,
    GstAdapter * adapter, gint * offset, gint * length);
static GstFlowReturn gst_gsmdec_handle_frame (GstAudioDecoder * dec,
    GstBuffer * in_buf);

/*static guint gst_gsmdec_signals[LAST_SIGNAL] = { 0 }; */

#define ENCODED_SAMPLES	160

static GstStaticPadTemplate gsmdec_sink_template =
    GST_STATIC_PAD_TEMPLATE ("sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/x-gsm, rate = (int) 8000, channels = (int) 1; "
        "audio/ms-gsm, rate = (int) [1, MAX], channels = (int) 1")
    );

static GstStaticPadTemplate gsmdec_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
    GST_PAD_SRC,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/x-raw, "
        "format = (string) " GST_AUDIO_NE (S16) ", "
        "layout = (string) interleaved, "
        "rate = (int) [1, MAX], channels = (int) 1")
    );

G_DEFINE_TYPE (GstGSMDec, gst_gsmdec, GST_TYPE_AUDIO_DECODER);
GST_ELEMENT_REGISTER_DEFINE (gsmdec, "gsmdec", GST_RANK_PRIMARY,
    GST_TYPE_GSMDEC);

static void
gst_gsmdec_class_init (GstGSMDecClass * klass)
{
  GstElementClass *element_class;
  GstAudioDecoderClass *base_class;

  element_class = (GstElementClass *) klass;
  base_class = (GstAudioDecoderClass *) klass;

  gst_element_class_add_static_pad_template (element_class,
      &gsmdec_sink_template);
  gst_element_class_add_static_pad_template (element_class,
      &gsmdec_src_template);
  gst_element_class_set_static_metadata (element_class, "GSM audio decoder",
      "Codec/Decoder/Audio", "Decodes GSM encoded audio",
      "Philippe Khalaf <burger@speedy.org>");

  base_class->start = GST_DEBUG_FUNCPTR (gst_gsmdec_start);
  base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmdec_stop);
  base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmdec_set_format);
  base_class->parse = GST_DEBUG_FUNCPTR (gst_gsmdec_parse);
  base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmdec_handle_frame);

  GST_DEBUG_CATEGORY_INIT (gsmdec_debug, "gsmdec", 0, "GSM Decoder");
}

static void
gst_gsmdec_init (GstGSMDec * gsmdec)
{
  gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (gsmdec), TRUE);
  gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
      (gsmdec), TRUE);
  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (gsmdec));
}

static gboolean
gst_gsmdec_start (GstAudioDecoder * dec)
{
  GstGSMDec *gsmdec = GST_GSMDEC (dec);

  GST_DEBUG_OBJECT (dec, "start");

  gsmdec->state = gsm_create ();

  return TRUE;
}

static gboolean
gst_gsmdec_stop (GstAudioDecoder * dec)
{
  GstGSMDec *gsmdec = GST_GSMDEC (dec);

  GST_DEBUG_OBJECT (dec, "stop");

  gsm_destroy (gsmdec->state);

  return TRUE;
}

static gboolean
gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
{
  GstGSMDec *gsmdec;
  GstStructure *s;
  gboolean ret = FALSE;
  gint rate;
  GstAudioInfo info;

  gsmdec = GST_GSMDEC (dec);

  s = gst_caps_get_structure (caps, 0);
  if (s == NULL)
    goto wrong_caps;

  /* figure out if we deal with plain or MSGSM */
  if (gst_structure_has_name (s, "audio/x-gsm"))
    gsmdec->use_wav49 = 0;
  else if (gst_structure_has_name (s, "audio/ms-gsm"))
    gsmdec->use_wav49 = 1;
  else
    goto wrong_caps;

  gsmdec->needed = 33;

  if (!gst_structure_get_int (s, "rate", &rate)) {
    GST_WARNING_OBJECT (gsmdec, "missing sample rate parameter from sink caps");
    goto beach;
  }

  /* MSGSM needs different framing */
  gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49);

  /* Setting up src caps based on the input sample rate. */
  gst_audio_info_init (&info);
  gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, rate, 1, NULL);

  ret = gst_audio_decoder_set_output_format (dec, &info);

  return ret;

  /* ERRORS */
wrong_caps:

  GST_ERROR_OBJECT (gsmdec, "invalid caps received");

beach:

  return ret;
}

static GstFlowReturn
gst_gsmdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
    gint * offset, gint * length)
{
  GstGSMDec *gsmdec = GST_GSMDEC (dec);
  guint size;

  size = gst_adapter_available (adapter);

  /* if input format is TIME each buffer should be self-contained and
   * the data is presumably packetised, and we should start with a clean
   * slate/state at the beginning of each buffer (for wav49 case) */
  if (dec->input_segment.format == GST_FORMAT_TIME) {
    *offset = 0;
    *length = size;
    gsmdec->needed = 33;
    return GST_FLOW_OK;
  }

  g_return_val_if_fail (size > 0, GST_FLOW_ERROR);

  if (size < gsmdec->needed)
    return GST_FLOW_EOS;

  *offset = 0;
  *length = gsmdec->needed;

  /* WAV49 requires alternating 33 and 32 bytes of input */
  if (gsmdec->use_wav49) {
    gsmdec->needed = (gsmdec->needed == 33 ? 32 : 33);
  }

  return GST_FLOW_OK;
}

static guint
gst_gsmdec_get_frame_count (GstGSMDec * dec, gsize buffer_size)
{
  guint count;

  if (dec->use_wav49) {
    count = (buffer_size / (33 + 32)) * 2;
    if (buffer_size % (33 + 32) >= dec->needed)
      ++count;
  } else {
    count = buffer_size / 33;
  }

  return count;
}

static GstFlowReturn
gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{
  GstGSMDec *gsmdec;
  gsm_signal *out_data;
  gsm_byte *data;
  GstFlowReturn ret = GST_FLOW_OK;
  GstBuffer *outbuf;
  GstMapInfo map, omap;
  gsize outsize;
  guint frames, i, errors = 0;

  /* no fancy draining */
  if (G_UNLIKELY (!buffer))
    return GST_FLOW_OK;

  gsmdec = GST_GSMDEC (dec);

  gst_buffer_map (buffer, &map, GST_MAP_READ);

  frames = gst_gsmdec_get_frame_count (gsmdec, map.size);

  /* always the same amount of output samples (20ms worth per frame) */
  outsize = ENCODED_SAMPLES * frames * sizeof (gsm_signal);
  outbuf = gst_buffer_new_and_alloc (outsize);

  gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
  out_data = (gsm_signal *) omap.data;
  data = (gsm_byte *) map.data;

  for (i = 0; i < frames; ++i) {
    /* now encode frame into the output buffer */
    if (gsm_decode (gsmdec->state, data, out_data) < 0) {
      /* invalid frame */
      GST_AUDIO_DECODER_ERROR (gsmdec, 1, STREAM, DECODE, (NULL),
          ("tried to decode an invalid frame"), ret);
      memset (out_data, 0, ENCODED_SAMPLES * sizeof (gsm_signal));
      ++errors;
    }
    out_data += ENCODED_SAMPLES;
    data += gsmdec->needed;
    if (gsmdec->use_wav49)
      gsmdec->needed = (gsmdec->needed == 33 ? 32 : 33);
  }

  gst_buffer_unmap (outbuf, &omap);
  gst_buffer_unmap (buffer, &map);

  if (errors == frames) {
    gst_buffer_unref (outbuf);
    outbuf = NULL;
  }

  gst_audio_decoder_finish_frame (dec, outbuf, 1);

  return ret;
}