gstreamer/ext/flac/gstflacdec.c

/* GStreamer
 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
 * Copyright (C) <2006,2011> Tim-Philipp Müller <tim centricular net>
 * Copyright (C) <2006> Jan Schmidt <thaytan at mad scientist com>
 *
 * 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.
 */

/**
 * SECTION:element-flacdec
 * @see_also: #GstFlacEnc
 *
 * flacdec decodes FLAC streams.
 * <ulink url="http://flac.sourceforge.net/">FLAC</ulink>
 * is a Free Lossless Audio Codec.
 *
 * <refsect2>
 * <title>Example launch line</title>
 * |[
 * gst-launch-0.11 filesrc location=media/small/dark.441-16-s.flac ! flacparse ! flacdec ! audioconvert ! audioresample ! autoaudiosink
 * ]|
 * |[
 * gst-launch-0.11 souphttpsrc location=http://gstreamer.freedesktop.org/media/small/dark.441-16-s.flac ! flacparse ! flacdec ! audioconvert ! audioresample ! queue min-threshold-buffers=10 ! autoaudiosink
 * ]|
 * </refsect2>
 */

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>

#include "gstflacdec.h"
#include <gst/gst-i18n-plugin.h>
#include <gst/audio/multichannel.h>
#include <gst/tag/tag.h>

/* Taken from http://flac.sourceforge.net/format.html#frame_header */
static const GstAudioChannelPosition channel_positions[8][8] = {
  {GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
  {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
      GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
      GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
      GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
      GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
        GST_AUDIO_CHANNEL_POSITION_LFE,
        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
      GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
  /* FIXME: 7/8 channel layouts are not defined in the FLAC specs */
  {
        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
        GST_AUDIO_CHANNEL_POSITION_LFE,
      GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
        GST_AUDIO_CHANNEL_POSITION_LFE,
        GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
      GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}
};

GST_DEBUG_CATEGORY_STATIC (flacdec_debug);
#define GST_CAT_DEFAULT flacdec_debug

static FLAC__StreamDecoderReadStatus
gst_flac_dec_read_stream (const FLAC__StreamDecoder * decoder,
    FLAC__byte buffer[], size_t * bytes, void *client_data);
static FLAC__StreamDecoderWriteStatus
gst_flac_dec_write_stream (const FLAC__StreamDecoder * decoder,
    const FLAC__Frame * frame,
    const FLAC__int32 * const buffer[], void *client_data);
static void gst_flac_dec_metadata_cb (const FLAC__StreamDecoder *
    decoder, const FLAC__StreamMetadata * metadata, void *client_data);
static void gst_flac_dec_error_cb (const FLAC__StreamDecoder *
    decoder, FLAC__StreamDecoderErrorStatus status, void *client_data);

static void gst_flac_dec_flush (GstAudioDecoder * audio_dec, gboolean hard);
static gboolean gst_flac_dec_set_format (GstAudioDecoder * dec, GstCaps * caps);
static gboolean gst_flac_dec_start (GstAudioDecoder * dec);
static gboolean gst_flac_dec_stop (GstAudioDecoder * dec);
static GstFlowReturn gst_flac_dec_handle_frame (GstAudioDecoder * audio_dec,
    GstBuffer * buf);

G_DEFINE_TYPE (GstFlacDec, gst_flac_dec, GST_TYPE_AUDIO_DECODER);

#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define FORMATS "{ S8LE, S16LE, S32LE } "
#else
#define FORMATS "{ S8BE, S16BE, S32BE } "
#endif

/* FIXME 0.11: Use width=32 for all depths and let audioconvert
 * handle the conversions instead of doing it ourself.
 */
#define GST_FLAC_DEC_SRC_CAPS                             \
    "audio/x-raw, "                                       \
    "format = (string) " FORMATS ", "                     \
    "rate = (int) [ 1, 655350 ], "                        \
    "channels = (int) [ 1, 8 ]"

#define GST_FLAC_DEC_SINK_CAPS                            \
    "audio/x-flac, "                                      \
    "framed = (boolean) true, "                           \
    "rate = (int) [ 1, 655350 ], "                        \
    "channels = (int) [ 1, 8 ]"

static GstStaticPadTemplate flac_dec_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
    GST_PAD_SRC,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS (GST_FLAC_DEC_SRC_CAPS));
static GstStaticPadTemplate flac_dec_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS (GST_FLAC_DEC_SINK_CAPS));

static void
gst_flac_dec_class_init (GstFlacDecClass * klass)
{
  GstAudioDecoderClass *audiodecoder_class;
  GstElementClass *gstelement_class;

  audiodecoder_class = (GstAudioDecoderClass *) klass;
  gstelement_class = (GstElementClass *) klass;

  GST_DEBUG_CATEGORY_INIT (flacdec_debug, "flacdec", 0, "flac decoder");

  audiodecoder_class->stop = GST_DEBUG_FUNCPTR (gst_flac_dec_stop);
  audiodecoder_class->start = GST_DEBUG_FUNCPTR (gst_flac_dec_start);
  audiodecoder_class->flush = GST_DEBUG_FUNCPTR (gst_flac_dec_flush);
  audiodecoder_class->set_format = GST_DEBUG_FUNCPTR (gst_flac_dec_set_format);
  audiodecoder_class->handle_frame =
      GST_DEBUG_FUNCPTR (gst_flac_dec_handle_frame);

  gst_element_class_add_pad_template (gstelement_class,
      gst_static_pad_template_get (&flac_dec_src_factory));
  gst_element_class_add_pad_template (gstelement_class,
      gst_static_pad_template_get (&flac_dec_sink_factory));

  gst_element_class_set_details_simple (gstelement_class, "FLAC audio decoder",
      "Codec/Decoder/Audio", "Decodes FLAC lossless audio streams",
      "Tim-Philipp Müller <tim@centricular.net>, "
      "Wim Taymans <wim.taymans@gmail.com>");
}

static void
gst_flac_dec_init (GstFlacDec * flacdec)
{
  /* nothing to do here */
}

static gboolean
gst_flac_dec_start (GstAudioDecoder * audio_dec)
{
  FLAC__StreamDecoderInitStatus s;
  GstFlacDec *dec;

  dec = GST_FLAC_DEC (audio_dec);

  ///////////// FIXME:
  dec->tags = gst_tag_list_new (GST_TAG_AUDIO_CODEC, "FLAC", NULL);

  dec->adapter = gst_adapter_new ();

  dec->decoder = FLAC__stream_decoder_new ();

  /* no point calculating MD5 since it's never checked here */
  FLAC__stream_decoder_set_md5_checking (dec->decoder, false);
  FLAC__stream_decoder_set_metadata_respond (dec->decoder,
      FLAC__METADATA_TYPE_VORBIS_COMMENT);
  FLAC__stream_decoder_set_metadata_respond (dec->decoder,
      FLAC__METADATA_TYPE_PICTURE);

  GST_DEBUG_OBJECT (dec, "initializing decoder");
  s = FLAC__stream_decoder_init_stream (dec->decoder,
      gst_flac_dec_read_stream, NULL, NULL, NULL, NULL,
      gst_flac_dec_write_stream, gst_flac_dec_metadata_cb,
      gst_flac_dec_error_cb, dec);

  if (s != FLAC__STREAM_DECODER_INIT_STATUS_OK) {
    GST_ELEMENT_ERROR (GST_ELEMENT (dec), LIBRARY, INIT, (NULL), (NULL));
    return FALSE;
  }

  dec->got_headers = FALSE;

  return TRUE;
}

static gboolean
gst_flac_dec_stop (GstAudioDecoder * dec)
{
  GstFlacDec *flacdec = GST_FLAC_DEC (dec);

  if (flacdec->decoder) {
    FLAC__stream_decoder_delete (flacdec->decoder);
    flacdec->decoder = NULL;
  }

  if (flacdec->adapter) {
    gst_adapter_clear (flacdec->adapter);
    g_object_unref (flacdec->adapter);
    flacdec->adapter = NULL;
  }

  if (flacdec->tags) {
    gst_tag_list_free (flacdec->tags);
    flacdec->tags = NULL;
  }

  return TRUE;
}

static gboolean
gst_flac_dec_set_format (GstAudioDecoder * dec, GstCaps * caps)
{
  /* if stream headers are present we could process them here already */
#if 0
  ///gst_adapter_push (dec->adapter, gst_buffer_ref (buf)); // for all stream headers
  /* The first time we get audio data, we know we got all the headers.
   * We then loop until all the metadata is processed, then do an extra
   * "process_single" step for the audio frame. */
  GST_DEBUG_OBJECT (dec,
      "First audio frame, ensuring all metadata is processed");
  if (!FLAC__stream_decoder_process_until_end_of_metadata (dec->decoder)) {
    GST_DEBUG_OBJECT (dec, "process_until_end_of_metadata failed");
  }
  GST_DEBUG_OBJECT (dec, "All headers and metadata are now processed");
#endif
  /* FIXME: refuse caps is there are no stream headers */
  GST_LOG_OBJECT (dec, "sink caps: %" GST_PTR_FORMAT, caps);
  return TRUE;
}

static gboolean
gst_flac_dec_update_metadata (GstFlacDec * flacdec,
    const FLAC__StreamMetadata * metadata)
{
  GstTagList *list;
  guint num, i;

  if (flacdec->tags)
    list = flacdec->tags;
  else
    flacdec->tags = list = gst_tag_list_new_empty ();

  num = metadata->data.vorbis_comment.num_comments;
  GST_DEBUG_OBJECT (flacdec, "%u tag(s) found", num);

  for (i = 0; i < num; ++i) {
    gchar *vc, *name, *value;

    vc = g_strndup ((gchar *) metadata->data.vorbis_comment.comments[i].entry,
        metadata->data.vorbis_comment.comments[i].length);

    if (gst_tag_parse_extended_comment (vc, &name, NULL, &value, TRUE)) {
      GST_DEBUG_OBJECT (flacdec, "%s : %s", name, value);
      if (value && strlen (value))
        gst_vorbis_tag_add (list, name, value);
      g_free (name);
      g_free (value);
    }

    g_free (vc);
  }

  return TRUE;
}

/* CRC-8, poly = x^8 + x^2 + x^1 + x^0, init = 0 */
static const guint8 crc8_table[256] = {
  0x00, 0x07, 0x0E, 0x09, 0x1C, 0x1B, 0x12, 0x15,
  0x38, 0x3F, 0x36, 0x31, 0x24, 0x23, 0x2A, 0x2D,
  0x70, 0x77, 0x7E, 0x79, 0x6C, 0x6B, 0x62, 0x65,
  0x48, 0x4F, 0x46, 0x41, 0x54, 0x53, 0x5A, 0x5D,
  0xE0, 0xE7, 0xEE, 0xE9, 0xFC, 0xFB, 0xF2, 0xF5,
  0xD8, 0xDF, 0xD6, 0xD1, 0xC4, 0xC3, 0xCA, 0xCD,
  0x90, 0x97, 0x9E, 0x99, 0x8C, 0x8B, 0x82, 0x85,
  0xA8, 0xAF, 0xA6, 0xA1, 0xB4, 0xB3, 0xBA, 0xBD,
  0xC7, 0xC0, 0xC9, 0xCE, 0xDB, 0xDC, 0xD5, 0xD2,
  0xFF, 0xF8, 0xF1, 0xF6, 0xE3, 0xE4, 0xED, 0xEA,
  0xB7, 0xB0, 0xB9, 0xBE, 0xAB, 0xAC, 0xA5, 0xA2,
  0x8F, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9D, 0x9A,
  0x27, 0x20, 0x29, 0x2E, 0x3B, 0x3C, 0x35, 0x32,
  0x1F, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0D, 0x0A,
  0x57, 0x50, 0x59, 0x5E, 0x4B, 0x4C, 0x45, 0x42,
  0x6F, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7D, 0x7A,
  0x89, 0x8E, 0x87, 0x80, 0x95, 0x92, 0x9B, 0x9C,
  0xB1, 0xB6, 0xBF, 0xB8, 0xAD, 0xAA, 0xA3, 0xA4,
  0xF9, 0xFE, 0xF7, 0xF0, 0xE5, 0xE2, 0xEB, 0xEC,
  0xC1, 0xC6, 0xCF, 0xC8, 0xDD, 0xDA, 0xD3, 0xD4,
  0x69, 0x6E, 0x67, 0x60, 0x75, 0x72, 0x7B, 0x7C,
  0x51, 0x56, 0x5F, 0x58, 0x4D, 0x4A, 0x43, 0x44,
  0x19, 0x1E, 0x17, 0x10, 0x05, 0x02, 0x0B, 0x0C,
  0x21, 0x26, 0x2F, 0x28, 0x3D, 0x3A, 0x33, 0x34,
  0x4E, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5C, 0x5B,
  0x76, 0x71, 0x78, 0x7F, 0x6A, 0x6D, 0x64, 0x63,
  0x3E, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2C, 0x2B,
  0x06, 0x01, 0x08, 0x0F, 0x1A, 0x1D, 0x14, 0x13,
  0xAE, 0xA9, 0xA0, 0xA7, 0xB2, 0xB5, 0xBC, 0xBB,
  0x96, 0x91, 0x98, 0x9F, 0x8A, 0x8D, 0x84, 0x83,
  0xDE, 0xD9, 0xD0, 0xD7, 0xC2, 0xC5, 0xCC, 0xCB,
  0xE6, 0xE1, 0xE8, 0xEF, 0xFA, 0xFD, 0xF4, 0xF3
};

static guint8
gst_flac_calculate_crc8 (guint8 * data, guint length)
{
  guint8 crc = 0;

  while (length--) {
    crc = crc8_table[crc ^ *data];
    ++data;
  }

  return crc;
}

/* FIXME: for our purposes it's probably enough to just check for the sync
 * marker - we just want to know if it's a header frame or not */
static gboolean
gst_flac_dec_scan_got_frame (GstFlacDec * flacdec, guint8 * data, guint size,
    gint64 * last_sample_num)
{
  guint headerlen;
  guint sr_from_end = 0;        /* can be 0, 8 or 16 */
  guint bs_from_end = 0;        /* can be 0, 8 or 16 */
  guint32 val = 0;
  guint8 bs, sr, ca, ss, pb;

  if (size < 10)
    return FALSE;

  /* sync */
  if (data[0] != 0xFF || (data[1] & 0xFC) != 0xF8)
    return FALSE;
  if (data[1] & 1) {
    GST_WARNING_OBJECT (flacdec, "Variable block size FLAC unsupported");
    return FALSE;
  }

  bs = (data[2] & 0xF0) >> 4;   /* blocksize marker   */
  sr = (data[2] & 0x0F);        /* samplerate marker  */
  ca = (data[3] & 0xF0) >> 4;   /* channel assignment */
  ss = (data[3] & 0x0F) >> 1;   /* sample size marker */
  pb = (data[3] & 0x01);        /* padding bit        */

  GST_LOG_OBJECT (flacdec,
      "got sync, bs=%x,sr=%x,ca=%x,ss=%x,pb=%x", bs, sr, ca, ss, pb);

  if (bs == 0 || sr == 0x0F || ca >= 0x0B || ss == 0x03 || ss == 0x07) {
    return FALSE;
  }

  /* read block size from end of header? */
  if (bs == 6)
    bs_from_end = 8;
  else if (bs == 7)
    bs_from_end = 16;

  /* read sample rate from end of header? */
  if (sr == 0x0C)
    sr_from_end = 8;
  else if (sr == 0x0D || sr == 0x0E)
    sr_from_end = 16;

  /* FIXME: This is can be 36 bit if variable block size is used,
   * fortunately not encoder supports this yet and we check for that
   * above.
   */
  val = (guint32) g_utf8_get_char_validated ((gchar *) data + 4, -1);

  if (val == (guint32) - 1 || val == (guint32) - 2) {
    GST_LOG_OBJECT (flacdec, "failed to read sample/frame");
    return FALSE;
  }

  headerlen = 4 + g_unichar_to_utf8 ((gunichar) val, NULL) +
      (bs_from_end / 8) + (sr_from_end / 8);

  if (gst_flac_calculate_crc8 (data, headerlen) != data[headerlen]) {
    GST_LOG_OBJECT (flacdec, "invalid checksum");
    return FALSE;
  }

  if (flacdec->min_blocksize == flacdec->max_blocksize) {
    *last_sample_num = (val + 1) * flacdec->min_blocksize;
  } else {
    *last_sample_num = 0;       /* FIXME: + length of last block in samples */
  }

  /* FIXME: only valid for fixed block size streams */
  GST_DEBUG_OBJECT (flacdec, "frame number: %" G_GINT64_FORMAT,
      *last_sample_num);

  if (flacdec->sample_rate > 0 && *last_sample_num != 0) {
    GST_DEBUG_OBJECT (flacdec, "last sample %" G_GINT64_FORMAT " = %"
        GST_TIME_FORMAT, *last_sample_num,
        GST_TIME_ARGS (*last_sample_num * GST_SECOND / flacdec->sample_rate));
  }

  return TRUE;
}

/* FIXME: let parser extract tags */
static void
gst_flac_extract_picture_buffer (GstFlacDec * dec,
    const FLAC__StreamMetadata * metadata)
{
  FLAC__StreamMetadata_Picture picture;
  GstTagList *tags;

  g_return_if_fail (metadata->type == FLAC__METADATA_TYPE_PICTURE);

  GST_LOG_OBJECT (dec, "Got PICTURE block");
  picture = metadata->data.picture;

  GST_DEBUG_OBJECT (dec, "declared MIME type is: '%s'",
      GST_STR_NULL (picture.mime_type));
  GST_DEBUG_OBJECT (dec, "image data is %u bytes", picture.data_length);

  tags = gst_tag_list_new_empty ();

  gst_tag_list_add_id3_image (tags, (guint8 *) picture.data,
      picture.data_length, picture.type);

  if (!gst_tag_list_is_empty (tags)) {
    gst_element_found_tags_for_pad (GST_ELEMENT (dec),
        GST_AUDIO_DECODER_SRC_PAD (dec), tags);
  } else {
    GST_DEBUG_OBJECT (dec, "problem parsing PICTURE block, skipping");
    gst_tag_list_free (tags);
  }
}

static void
gst_flac_dec_metadata_cb (const FLAC__StreamDecoder * decoder,
    const FLAC__StreamMetadata * metadata, void *client_data)
{
  GstFlacDec *flacdec = GST_FLAC_DEC (client_data);

  GST_LOG_OBJECT (flacdec, "metadata type: %d", metadata->type);

  switch (metadata->type) {
    case FLAC__METADATA_TYPE_STREAMINFO:{
      gint64 samples;
      guint depth;

      samples = metadata->data.stream_info.total_samples;

      flacdec->min_blocksize = metadata->data.stream_info.min_blocksize;
      flacdec->max_blocksize = metadata->data.stream_info.max_blocksize;
      flacdec->sample_rate = metadata->data.stream_info.sample_rate;
      flacdec->depth = depth = metadata->data.stream_info.bits_per_sample;
      flacdec->channels = metadata->data.stream_info.channels;

      if (depth < 9)
        flacdec->width = 8;
      else if (depth < 17)
        flacdec->width = 16;
      else
        flacdec->width = 32;

      GST_DEBUG_OBJECT (flacdec, "blocksize: min=%u, max=%u",
          flacdec->min_blocksize, flacdec->max_blocksize);
      GST_DEBUG_OBJECT (flacdec, "sample rate: %u, channels: %u",
          flacdec->sample_rate, flacdec->channels);
      GST_DEBUG_OBJECT (flacdec, "depth: %u, width: %u", flacdec->depth,
          flacdec->width);

      GST_DEBUG_OBJECT (flacdec, "total samples = %" G_GINT64_FORMAT, samples);
      break;
    }
    case FLAC__METADATA_TYPE_PICTURE:{
      gst_flac_extract_picture_buffer (flacdec, metadata);
      break;
    }
    case FLAC__METADATA_TYPE_VORBIS_COMMENT:
      gst_flac_dec_update_metadata (flacdec, metadata);
      break;
    default:
      break;
  }
}

static void
gst_flac_dec_error_cb (const FLAC__StreamDecoder * d,
    FLAC__StreamDecoderErrorStatus status, void *client_data)
{
  const gchar *error;
  GstFlacDec *dec;

  dec = GST_FLAC_DEC (client_data);

  switch (status) {
    case FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC:
      /* Ignore this error and keep processing */
      return;
    case FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER:
      error = "bad header";
      break;
    case FLAC__STREAM_DECODER_ERROR_STATUS_FRAME_CRC_MISMATCH:
      error = "CRC mismatch";
      break;
    default:
      error = "unknown error";
      break;
  }

  GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("%s (%d)", error, status));
  dec->last_flow = GST_FLOW_ERROR;
}

static FLAC__StreamDecoderReadStatus
gst_flac_dec_read_stream (const FLAC__StreamDecoder * decoder,
    FLAC__byte buffer[], size_t * bytes, void *client_data)
{
  GstFlacDec *dec = GST_FLAC_DEC (client_data);
  guint len;

  len = MIN (gst_adapter_available (dec->adapter), *bytes);

  if (len == 0) {
    GST_LOG_OBJECT (dec, "0 bytes available at the moment");
    return FLAC__STREAM_DECODER_READ_STATUS_ABORT;
  }

  GST_LOG_OBJECT (dec, "feeding %u bytes to decoder (available=%u, bytes=%u)",
      len, gst_adapter_available (dec->adapter), (guint) * bytes);
  gst_adapter_copy (dec->adapter, buffer, 0, len);
  *bytes = len;

  gst_adapter_flush (dec->adapter, len);

  return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
}

static FLAC__StreamDecoderWriteStatus
gst_flac_dec_write (GstFlacDec * flacdec, const FLAC__Frame * frame,
    const FLAC__int32 * const buffer[])
{
  GstFlowReturn ret = GST_FLOW_OK;
  GstBuffer *outbuf;
  guint depth = frame->header.bits_per_sample;
  guint width;
  guint sample_rate = frame->header.sample_rate;
  guint channels = frame->header.channels;
  guint samples = frame->header.blocksize;
  guint j, i;
  gpointer data;
  gsize size;
  const gchar *format;

  GST_LOG_OBJECT (flacdec, "samples in frame header: %d", samples);

  if (depth == 0) {
    if (flacdec->depth < 4 || flacdec->depth > 32) {
      GST_ERROR_OBJECT (flacdec, "unsupported depth %d from STREAMINFO",
          flacdec->depth);
      ret = GST_FLOW_ERROR;
      goto done;
    }

    depth = flacdec->depth;
    if (depth < 9)
      depth = 8;
    else if (depth < 17)
      depth = 16;
    else
      depth = 32;
  }

  switch (depth) {
    case 8:
      width = 8;
      format = GST_AUDIO_NE (S8);
      break;
    case 12:
    case 16:
      width = 16;
      format = GST_AUDIO_NE (S16);
      break;
    case 20:
    case 24:
    case 32:
      width = 32;
      format = GST_AUDIO_NE (S32);
      break;
    default:
      GST_ERROR_OBJECT (flacdec, "unsupported depth %d", depth);
      ret = GST_FLOW_ERROR;
      goto done;
  }

  if (sample_rate == 0) {
    if (flacdec->sample_rate != 0) {
      sample_rate = flacdec->sample_rate;
    } else {
      GST_ERROR_OBJECT (flacdec, "unknown sample rate");
      ret = GST_FLOW_ERROR;
      goto done;
    }
  }

  if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (flacdec))) {
    GstCaps *caps;

    GST_DEBUG_OBJECT (flacdec, "Negotiating %d Hz @ %d channels",
        frame->header.sample_rate, channels);

    caps = gst_caps_new_simple ("audio/x-raw",
        "format", G_TYPE_STRING, format,
        "rate", G_TYPE_INT, frame->header.sample_rate,
        "channels", G_TYPE_INT, channels, NULL);

    if (channels > 2) {
      GstStructure *s = gst_caps_get_structure (caps, 0);

      gst_audio_set_channel_positions (s, channel_positions[channels - 1]);
    }

    flacdec->depth = depth;
    flacdec->width = width;
    flacdec->channels = channels;
    flacdec->sample_rate = sample_rate;

    gst_audio_decoder_set_outcaps (GST_AUDIO_DECODER (flacdec), caps);
    gst_caps_unref (caps);
  }

  if (flacdec->tags) {
    gst_element_found_tags_for_pad (GST_ELEMENT (flacdec),
        GST_AUDIO_DECODER_SRC_PAD (flacdec), flacdec->tags);
    flacdec->tags = NULL;
  }

  GST_LOG_OBJECT (flacdec, "alloc_buffer_and_set_caps");
  outbuf = gst_buffer_new_allocate (NULL, samples * channels * (width / 8), 0);

  data = gst_buffer_map (outbuf, &size, NULL, GST_MAP_WRITE);
  if (width == 8) {
    gint8 *outbuffer = (gint8 *) data;

    for (i = 0; i < samples; i++) {
      for (j = 0; j < channels; j++) {
        *outbuffer++ = (gint8) buffer[j][i];
      }
    }
  } else if (width == 16) {
    gint16 *outbuffer = (gint16 *) data;

    for (i = 0; i < samples; i++) {
      for (j = 0; j < channels; j++) {
        *outbuffer++ = (gint16) buffer[j][i];
      }
    }
  } else if (width == 32) {
    gint32 *outbuffer = (gint32 *) data;

    for (i = 0; i < samples; i++) {
      for (j = 0; j < channels; j++) {
        *outbuffer++ = (gint32) buffer[j][i];
      }
    }
  } else {
    g_assert_not_reached ();
  }
  gst_buffer_unmap (outbuf, data, size);

  GST_DEBUG_OBJECT (flacdec, "pushing %d samples", samples);

  ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (flacdec), outbuf, 1);

  if (G_UNLIKELY (ret != GST_FLOW_OK)) {
    GST_DEBUG_OBJECT (flacdec, "finish_frame flow %s", gst_flow_get_name (ret));
  }

done:

  /* we act on the flow return value later in the handle_frame function, as we
   * don't want to mess up the internal decoder state by returning ABORT when
   * the error is in fact non-fatal (like a pad in flushing mode) and we want
   * to continue later. So just pretend everything's dandy and act later. */
  flacdec->last_flow = ret;

  return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}

static FLAC__StreamDecoderWriteStatus
gst_flac_dec_write_stream (const FLAC__StreamDecoder * decoder,
    const FLAC__Frame * frame,
    const FLAC__int32 * const buffer[], void *client_data)
{
  return gst_flac_dec_write (GST_FLAC_DEC (client_data), frame, buffer);
}

static void
gst_flac_dec_flush (GstAudioDecoder * audio_dec, gboolean hard)
{
  GstFlacDec *dec = GST_FLAC_DEC (audio_dec);

  if (!hard) {
    guint available = gst_adapter_available (dec->adapter);

    if (available > 0) {
      GST_INFO_OBJECT (dec, "draining, %u bytes left in adapter", available);
      FLAC__stream_decoder_process_until_end_of_stream (dec->decoder);
    }
  }

  FLAC__stream_decoder_flush (dec->decoder);
  gst_adapter_clear (dec->adapter);
}

static GstFlowReturn
gst_flac_dec_handle_frame (GstAudioDecoder * audio_dec, GstBuffer * buf)
{
  GstFlacDec *dec;

  dec = GST_FLAC_DEC (audio_dec);

  /* drain remaining data? */
  if (G_UNLIKELY (buf == NULL)) {
    gst_flac_dec_flush (audio_dec, FALSE);
    return GST_FLOW_OK;
  }

  GST_LOG_OBJECT (dec, "frame: ts %" GST_TIME_FORMAT ", flags 0x%04x, %u bytes",
      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), GST_BUFFER_FLAGS (buf),
      gst_buffer_get_size (buf));

  /* drop any in-stream headers, we've processed those in set_format already */
  if (G_UNLIKELY (!dec->got_headers)) {
    gboolean got_audio_frame;
    gint64 unused;
    guint8 *data;
    gsize size;

    /* check if this is a flac audio frame (rather than a header or junk) */
    data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
    got_audio_frame = gst_flac_dec_scan_got_frame (dec, data, size, &unused);
    gst_buffer_unmap (buf, data, size);

    if (!got_audio_frame) {
      GST_INFO_OBJECT (dec, "dropping in-stream header, %d bytes", size);
      return GST_FLOW_OK;
    }

    GST_INFO_OBJECT (dec, "first audio frame, got all in-stream headers now");
    dec->got_headers = TRUE;
  }

  gst_adapter_push (dec->adapter, gst_buffer_ref (buf));
  buf = NULL;

  dec->last_flow = GST_FLOW_OK;

  /* framed - there should always be enough data to decode something */
  GST_LOG_OBJECT (dec, "%u bytes available",
      gst_adapter_available (dec->adapter));

  if (!FLAC__stream_decoder_process_single (dec->decoder)) {
    GST_INFO_OBJECT (dec, "process_single failed");
  }

  return dec->last_flow;
}