1 files changed, 0 insertions, 1221 deletions
diff --git a/src/plugins/previewopus_extractor.c b/src/plugins/previewopus_extractor.c
deleted file mode 100644
index 45ea420..0000000
--- a/src/plugins/previewopus_extractor.c
+++ /dev/null
@@ -1,1221 +0,0 @@
-/*
-     This file is part of libextractor.
-     Copyright Copyright (C) 2008, 2013 Bruno Cabral and Christian Grothoff
-     libextractor is free software; you can redistribute it and/or modify
-     it under the terms of the GNU General Public License as published
-     by the Free Software Foundation; either version 3, or (at your
-     option) any later version.
-     libextractor 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
-     General Public License for more details.
-     You should have received a copy of the GNU General Public License
-     along with libextractor; see the file COPYING.  If not, write to the
-     Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
-     Boston, MA 02110-1301, USA.
- */
-/**
- * @file previewopus_extractor.c
- * @author Bruno Cabral
- * @author Christian Grothoff
- * @brief this extractor produces a binary encoded
- * audio snippet of music/video files using ffmpeg libs.
- *
- * Based on ffmpeg samples.
- *
- * Note that ffmpeg has a few issues:
- * (1) there are no recent official releases of the ffmpeg libs
- * (2) ffmpeg has a history of having security issues (parser is not robust)
- *
- *  So this plugin cannot be recommended for system with high security
- *requirements.
- */
-#include "platform.h"
-#include "extractor.h"
-#include <magic.h>
-#include <libavutil/avutil.h>
-#include <libavutil/audio_fifo.h>
-#include <libavutil/opt.h>
-#include <libavutil/mathematics.h>
-#include <libavformat/avformat.h>
-#include <libavcodec/avcodec.h>
-#include <libswscale/swscale.h>
-#include <libavresample/avresample.h>
-/**
- * Set to 1 to enable debug output.
- */
-#define DEBUG 0
-/**
- * Set to 1 to enable a output file for testing.
- */
-#define OUTPUT_FILE 0
-/**
- * Maximum size in bytes for the preview.
- */
-#define MAX_SIZE (28 * 1024)
-/**
- * HardLimit for file
- */
-#define HARD_LIMIT_SIZE (50 * 1024)
-/** The output bit rate in kbit/s */
-#define OUTPUT_BIT_RATE 28000
-/** The number of output channels */
-#define OUTPUT_CHANNELS 2
-/** The audio sample output format */
-#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
-/** Our output buffer*/
-static unsigned char *buffer;
-/** Actual output buffer size */
-static int totalSize;
-/**
- * Convert an error code into a text message.
- * @param error Error code to be converted
- * @return Corresponding error text (not thread-safe)
- */
-static char *const
-get_error_text (const int error)
-{
-  static char error_buffer[255];
-  av_strerror (error, error_buffer, sizeof(error_buffer));
-  return error_buffer;
-}
-/**
- * Read callback.
- *
- * @param opaque the 'struct EXTRACTOR_ExtractContext'
- * @param buf where to write data
- * @param buf_size how many bytes to read
- * @return -1 on error (or for unknown file size)
- */
-static int
-read_cb (void *opaque,
-         uint8_t *buf,
-         int buf_size)
-{
-  struct EXTRACTOR_ExtractContext *ec = opaque;
-  void *data;
-  ssize_t ret;
-  ret = ec->read (ec->cls, &data, buf_size);
-  if (ret <= 0)
-    return ret;
-  memcpy (buf, data, ret);
-  return ret;
-}
-/**
- * Seek callback.
- *
- * @param opaque the 'struct EXTRACTOR_ExtractContext'
- * @param offset where to seek
- * @param whence how to seek; AVSEEK_SIZE to return file size without seeking
- * @return -1 on error (or for unknown file size)
- */
-static int64_t
-seek_cb (void *opaque,
-         int64_t offset,
-         int whence)
-{
-  struct EXTRACTOR_ExtractContext *ec = opaque;
-  if (AVSEEK_SIZE == whence)
-    return ec->get_size (ec->cls);
-  return ec->seek (ec->cls, offset, whence);
-}
-/**
- * write callback.
- *
- * @param opaque NULL
- * @param pBuffer to write
- * @param pBufferSize , amount to write
- * @return 0 on error
- */
-static int
-writePacket (void *opaque,
-             unsigned char *pBuffer,
-             int pBufferSize)
-{
-  int sizeToCopy = pBufferSize;
-  if ( (totalSize + pBufferSize) > HARD_LIMIT_SIZE)
-    sizeToCopy = HARD_LIMIT_SIZE - totalSize;
-  memcpy (buffer + totalSize, pBuffer, sizeToCopy);
-  totalSize += sizeToCopy;
-  return sizeToCopy;
-}
-/**
- * Open an output file and the required encoder.
- * Also set some basic encoder parameters.
- * Some of these parameters are based on the input file's parameters.
- */
-static int
-open_output_file (
-  AVCodecContext *input_codec_context,
-  AVFormatContext **output_format_context,
-  AVCodecContext **output_codec_context)
-{
-  AVStream *stream               = NULL;
-  AVCodec *output_codec          = NULL;
-  AVIOContext *io_ctx;
-  int error;
-  unsigned char *iob;
-  if (NULL == (iob = av_malloc (16 * 1024)))
-    return AVERROR_EXIT;
-  if (NULL == (io_ctx = avio_alloc_context (iob, 16 * 1024,
-                                            AVIO_FLAG_WRITE, NULL,
-                                            NULL,
-                                            &writePacket /* no writing */,
-                                            NULL)))
-  {
-    av_free (iob);
-    return AVERROR_EXIT;
-  }
-  if (NULL == ((*output_format_context) = avformat_alloc_context ()))
-  {
-    av_free (io_ctx);
-    return AVERROR_EXIT;
-  }
-  (*output_format_context)->pb = io_ctx;
-  /** Guess the desired container format based on the file extension. */
-  if (! ((*output_format_context)->oformat = av_guess_format (NULL,
-                                                              "file.ogg",
-                                                              NULL)))
-  {
-#if DEBUG
-    fprintf (stderr, "Could not find output file format\n");
-#endif
-    error = AVERROR (ENOSYS);
-    goto cleanup;
-  }
-  /** Find the encoder to be used by its name. */
-  if (! (output_codec = avcodec_find_encoder (AV_CODEC_ID_OPUS)))
-  {
-#if DEBUG
-    fprintf (stderr, "Could not find an OPUS encoder.\n");
-#endif
-    error = AVERROR (ENOSYS);
-    goto cleanup;
-  }
-  /** Create a new audio stream in the output file container. */
-  if (! (stream = avformat_new_stream (*output_format_context, output_codec)))
-  {
-#if DEBUG
-    fprintf (stderr, "Could not create new stream\n");
-#endif
-    error = AVERROR (ENOMEM);
-    goto cleanup;
-  }
-  /** Save the encoder context for easiert access later. */
-  *output_codec_context = stream->codec;
-  /**
-   * Set the basic encoder parameters.
-   * The input file's sample rate is used to avoid a sample rate conversion.
-   */
-  (*output_codec_context)->channels       = OUTPUT_CHANNELS;
-  (*output_codec_context)->channel_layout = av_get_default_channel_layout (
-    OUTPUT_CHANNELS);
-  (*output_codec_context)->sample_rate    = 48000; // Opus need 48000
-  (*output_codec_context)->sample_fmt     = AV_SAMPLE_FMT_S16;
-  (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
-  /** Open the encoder for the audio stream to use it later. */
-  if ((error = avcodec_open2 (*output_codec_context, output_codec, NULL)) < 0)
-  {
-#if DEBUG
-    fprintf (stderr, "Could not open output codec (error '%s')\n",
-             get_error_text (error));
-#endif
-    goto cleanup;
-  }
-  return 0;
-cleanup:
-  av_free (io_ctx);
-  return error < 0 ? error : AVERROR_EXIT;
-}
-/** Initialize one data packet for reading or writing. */
-static void
-init_packet (AVPacket *packet)
-{
-  av_init_packet (packet);
-  /** Set the packet data and size so that it is recognized as being empty. */
-  packet->data = NULL;
-  packet->size = 0;
-}
-/** Initialize one audio frame for reading from the input file */
-static int
-init_input_frame (AVFrame **frame)
-{
-  *frame = av_frame_alloc ();
-  if (NULL == *frame)
-  {
-#if DEBUG
-    fprintf (stderr, "Could not allocate input frame\n");
-#endif
-    return AVERROR (ENOMEM);
-  }
-  return 0;
-}
-/**
- * Initialize the audio resampler based on the input and output codec settings.
- * If the input and output sample formats differ, a conversion is required
- * libavresample takes care of this, but requires initialization.
- */
-static int
-init_resampler (AVCodecContext *input_codec_context,
-                AVCodecContext *output_codec_context,
-                AVAudioResampleContext  **resample_context)
-{
-  /**
-   * Only initialize the resampler if it is necessary, i.e.,
-   * if and only if the sample formats differ.
-   */
-  if ((input_codec_context->sample_fmt != output_codec_context->sample_fmt) ||
-      (input_codec_context->channels != output_codec_context->channels) )
-  {
-    int error;
-    /** Create a resampler context for the conversion. */
-    if (! (*resample_context = avresample_alloc_context ()))
-    {
-#if DEBUG
-      fprintf (stderr, "Could not allocate resample context\n");
-#endif
-      return AVERROR (ENOMEM);
-    }
-    /**
-     * Set the conversion parameters.
-     * Default channel layouts based on the number of channels
-     * are assumed for simplicity (they are sometimes not detected
-     * properly by the demuxer and/or decoder).
-     */av_opt_set_int (*resample_context, "in_channel_layout",
-                    av_get_default_channel_layout (
-                      input_codec_context->channels), 0);
-    av_opt_set_int (*resample_context, "out_channel_layout",
-                    av_get_default_channel_layout (
-                      output_codec_context->channels), 0);
-    av_opt_set_int (*resample_context, "in_sample_rate",
-                    input_codec_context->sample_rate, 0);
-    av_opt_set_int (*resample_context, "out_sample_rate",
-                    output_codec_context->sample_rate, 0);
-    av_opt_set_int (*resample_context, "in_sample_fmt",
-                    input_codec_context->sample_fmt, 0);
-    av_opt_set_int (*resample_context, "out_sample_fmt",
-                    output_codec_context->sample_fmt, 0);
-    /** Open the resampler with the specified parameters. */
-    if ((error = avresample_open (*resample_context)) < 0)
-    {
-#if DEBUG
-      fprintf (stderr, "Could not open resample context\n");
-#endif
-      avresample_free (resample_context);
-      return error;
-    }
-  }
-  return 0;
-}
-/** Initialize a FIFO buffer for the audio samples to be encoded. */
-static int
-init_fifo (AVAudioFifo **fifo)
-{
-  /** Create the FIFO buffer based on the specified output sample format. */
-  if (! (*fifo = av_audio_fifo_alloc (OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS,
-                                      1)))
-  {
-#if DEBUG
-    fprintf (stderr, "Could not allocate FIFO\n");
-#endif
-    return AVERROR (ENOMEM);
-  }
-  return 0;
-}
-/** Write the header of the output file container. */
-static int
-write_output_file_header (AVFormatContext *output_format_context)
-{
-  int error;
-  if ((error = avformat_write_header (output_format_context, NULL)) < 0)
-  {
-#if DEBUG
-    fprintf (stderr, "Could not write output file header (error '%s')\n",
-             get_error_text (error));
-#endif
-    return error;
-  }
-  return 0;
-}
-/** Decode one audio frame from the input file. */
-static int
-decode_audio_frame (AVFrame *frame,
-                    AVFormatContext *input_format_context,
-                    AVCodecContext *input_codec_context, int audio_stream_index,
-                    int *data_present, int *finished)
-{
-  /** Packet used for temporary storage. */
-  AVPacket input_packet;
-  int error;
-  init_packet (&input_packet);
-  /** Read one audio frame from the input file into a temporary packet. */
-  while (1)
-  {
-    if ((error = av_read_frame (input_format_context, &input_packet)) < 0)
-    {
-      /** If we are the the end of the file, flush the decoder below. */
-      if (error == AVERROR_EOF)
-      {
-#if DEBUG
-        fprintf (stderr, "EOF in decode_audio\n");
-#endif
-        *finished = 1;
-      }
-      else
-      {
-#if DEBUG
-        fprintf (stderr, "Could not read frame (error '%s')\n",
-                 get_error_text (error));
-#endif
-        return error;
-      }
-    }
-    if (input_packet.stream_index == audio_stream_index)
-      break;
-  }
-  /**
-   * Decode the audio frame stored in the temporary packet.
-   * The input audio stream decoder is used to do this.
-   * If we are at the end of the file, pass an empty packet to the decoder
-   * to flush it.
-   */if ((error = avcodec_decode_audio4 (input_codec_context, frame,
-                                      data_present, &input_packet)) < 0)
-  {
-#if DEBUG
-    fprintf (stderr, "Could not decode frame (error '%s')\n",
-             get_error_text (error));
-#endif
-    av_packet_unref (&input_packet);
-    return error;
-  }
-  /**
-   * If the decoder has not been flushed completely, we are not finished,
-   * so that this function has to be called again.
-   */
-  if (*finished && *data_present)
-    *finished = 0;
-  av_packet_unref (&input_packet);
-  return 0;
-}
-/**
- * Initialize a temporary storage for the specified number of audio samples.
- * The conversion requires temporary storage due to the different format.
- * The number of audio samples to be allocated is specified in frame_size.
- */
-static int
-init_converted_samples (uint8_t ***converted_input_samples, int*out_linesize,
-                        AVCodecContext *output_codec_context,
-                        int frame_size)
-{
-  int error;
-  /**
-   * Allocate as many pointers as there are audio channels.
-   * Each pointer will later point to the audio samples of the corresponding
-   * channels (although it may be NULL for interleaved formats).
-   */if (! (*converted_input_samples = calloc (output_codec_context->channels,
-                                            sizeof(**converted_input_samples))))
-  {
-#if DEBUG
-    fprintf (stderr, "Could not allocate converted input sample pointers\n");
-#endif
-    return AVERROR (ENOMEM);
-  }
-  /**
-   * Allocate memory for the samples of all channels in one consecutive
-   * block for convenience.
-   */
-  if ((error = av_samples_alloc (*converted_input_samples, out_linesize,
-                                 output_codec_context->channels,
-                                 frame_size,
-                                 output_codec_context->sample_fmt, 0)) < 0)
-  {
-#if DEBUG
-    fprintf (stderr,
-             "Could not allocate converted input samples (error '%s')\n",
-             get_error_text (error));
-#endif
-    av_freep (&(*converted_input_samples)[0]);
-    free (*converted_input_samples);
-    return error;
-  }
-  return 0;
-}
-/**
- * Convert the input audio samples into the output sample format.
- * The conversion happens on a per-frame basis, the size of which is specified
- * by frame_size.
- */
-static int
-convert_samples (uint8_t **input_data,
-                 uint8_t **converted_data, const int in_sample, const int
-                 out_sample, const int out_linesize,
-                 AVAudioResampleContext  *resample_context)
-{
-  int error;
-  /** Convert the samples using the resampler. */
-  if ((error = avresample_convert (resample_context, converted_data,
-                                   out_linesize,
-                                   out_sample, input_data, 0, in_sample)) < 0)
-  {
-#if DEBUG
-    fprintf (stderr, "Could not convert input samples (error '%s')\n",
-             get_error_text (error));
-#endif
-    return error;
-  }
-  /**
-   * Perform a sanity check so that the number of converted samples is
-   * not greater than the number of samples to be converted.
-   * If the sample rates differ, this case has to be handled differently
-   */if (avresample_available (resample_context))
-  {
-#if DEBUG
-    fprintf (stderr, "%i Converted samples left over\n",avresample_available (
-               resample_context));
-#endif
-  }
-  return 0;
-}
-/** Add converted input audio samples to the FIFO buffer for later processing. */
-static int
-add_samples_to_fifo (AVAudioFifo *fifo,
-                     uint8_t **converted_input_samples,
-                     const int frame_size)
-{
-  int error;
-  /**
-   * Make the FIFO as large as it needs to be to hold both,
-   * the old and the new samples.
-   */
-  if ((error = av_audio_fifo_realloc (fifo, av_audio_fifo_size (fifo)
-                                      + frame_size)) < 0)
-  {
-#if DEBUG
-    fprintf (stderr, "Could not reallocate FIFO\n");
-#endif
-    return error;
-  }
-  /** Store the new samples in the FIFO buffer. */
-  if (av_audio_fifo_write (fifo, (void **) converted_input_samples,
-                           frame_size) < frame_size)
-  {
-#if DEBUG
-    fprintf (stderr, "Could not write data to FIFO\n");
-#endif
-    return AVERROR_EXIT;
-  }
-  return 0;
-}
-/**
- * Read one audio frame from the input file, decodes, converts and stores
- * it in the FIFO buffer.
- */
-static int
-read_decode_convert_and_store (AVAudioFifo *fifo,
-                               AVFormatContext *input_format_context,
-                               AVCodecContext *input_codec_context,
-                               AVCodecContext *output_codec_context,
-                               AVAudioResampleContext  *resampler_context, int
-                               audio_stream_index,
-                               int *finished)
-{
-  /** Temporary storage of the input samples of the frame read from the file. */
-  AVFrame *input_frame = NULL;
-  /** Temporary storage for the converted input samples. */
-  uint8_t **converted_input_samples = NULL;
-  int data_present;
-  int ret = AVERROR_EXIT;
-  /** Initialize temporary storage for one input frame. */
-  if (init_input_frame (&input_frame))
-  {
-#if DEBUG
-    fprintf (stderr, "Failed at init frame\n");
-#endif
-    goto cleanup;
-  }
-  /** Decode one frame worth of audio samples. */
-  if (decode_audio_frame (input_frame, input_format_context,
-                          input_codec_context, audio_stream_index,
-                          &data_present,  finished))
-  {
-#if DEBUG
-    fprintf (stderr, "Failed at decode audio\n");
-#endif
-    goto cleanup;
-  }
-  /**
-   * If we are at the end of the file and there are no more samples
-   * in the decoder which are delayed, we are actually finished.
-   * This must not be treated as an error.
-   */if (*finished && ! data_present)
-  {
-    ret = 0;
-#if DEBUG
-    fprintf (stderr, "Failed at finished or no data\n");
-#endif
-    goto cleanup;
-  }
-  /** If there is decoded data, convert and store it */
-  if (data_present)
-  {
-    int out_linesize;
-    // FIX ME: I'm losing samples, but can't get it to work.
-    int out_samples = avresample_available (resampler_context)
-                      + avresample_get_delay (resampler_context)
-                      + input_frame->nb_samples;
-    // fprintf(stderr, "Input nbsamples %i out_samples: %i \n",input_frame->nb_samples,out_samples);
-    /** Initialize the temporary storage for the converted input samples. */
-    if (init_converted_samples (&converted_input_samples, &out_linesize,
-                                output_codec_context,
-                                out_samples))
-    {
-#if DEBUG
-      fprintf (stderr, "Failed at init_converted_samples\n");
-#endif
-      goto cleanup;
-    }
-    /**
-     * Convert the input samples to the desired output sample format.
-     * This requires a temporary storage provided by converted_input_samples.
-     */
-    if (convert_samples (input_frame->extended_data, converted_input_samples,
-                         input_frame->nb_samples, out_samples, out_linesize,
-                         resampler_context))
-    {
-#if DEBUG
-      fprintf (stderr, "Failed at convert_samples, input frame %i \n",
-               input_frame->nb_samples);
-#endif
-      goto cleanup;
-    }
-    /** Add the converted input samples to the FIFO buffer for later processing. */
-    if (add_samples_to_fifo (fifo, converted_input_samples,
-                             out_samples))
-    {
-#if DEBUG
-      fprintf (stderr, "Failed at add_samples_to_fifo\n");
-#endif
-      goto cleanup;
-    }
-    ret = 0;
-  }
-  ret = 0;
-cleanup:
-  if (converted_input_samples)
-  {
-    av_freep (&converted_input_samples[0]);
-    free (converted_input_samples);
-  }
-  av_frame_free (&input_frame);
-  return ret;
-}
-/**
- * Initialize one input frame for writing to the output file.
- * The frame will be exactly frame_size samples large.
- */
-static int
-init_output_frame (AVFrame **frame,
-                   AVCodecContext *output_codec_context,
-                   int frame_size)
-{
-  int error;
-  /** Create a new frame to store the audio samples. */
-  *frame = av_frame_alloc ();
-  if (NULL == *frame)
-  {
-#if DEBUG
-    fprintf (stderr, "Could not allocate output frame\n");
-#endif
-    return AVERROR_EXIT;
-  }
-  /**
-   * Set the frame's parameters, especially its size and format.
-   * av_frame_get_buffer needs this to allocate memory for the
-   * audio samples of the frame.
-   * Default channel layouts based on the number of channels
-   * are assumed for simplicity.
-   */(*frame)->nb_samples  = frame_size;
-  (*frame)->channel_layout = output_codec_context->channel_layout;
-  (*frame)->format         = output_codec_context->sample_fmt;
-  (*frame)->sample_rate    = output_codec_context->sample_rate;
-  // fprintf(stderr, "%i %i  \n",frame_size , (*frame)->format,(*frame)->sample_rate);
-  /**
-   * Allocate the samples of the created frame. This call will make
-   * sure that the audio frame can hold as many samples as specified.
-   */
-  if ((error = av_frame_get_buffer (*frame, 0)) < 0)
-  {
-#if DEBUG
-    fprintf (stderr, "Could allocate output frame samples (error '%s')\n",
-             get_error_text (error));
-#endif
-    av_frame_free (frame);
-    return error;
-  }
-  return 0;
-}
-/** Encode one frame worth of audio to the output file. */
-static int
-encode_audio_frame (AVFrame *frame,
-                    AVFormatContext *output_format_context,
-                    AVCodecContext *output_codec_context,
-                    int *data_present)
-{
-  /** Packet used for temporary storage. */
-  AVPacket output_packet;
-  int error;
-  init_packet (&output_packet);
-  /**
-   * Encode the audio frame and store it in the temporary packet.
-   * The output audio stream encoder is used to do this.
-   */
-  if ((error = avcodec_encode_audio2 (output_codec_context, &output_packet,
-                                      frame, data_present)) < 0)
-  {
-#if DEBUG
-    fprintf (stderr, "Could not encode frame (error '%s')\n",
-             get_error_text (error));
-#endif
-    av_packet_unref (&output_packet);
-    return error;
-  }
-  /** Write one audio frame from the temporary packet to the output file. */
-  if (*data_present)
-  {
-    if ((error = av_write_frame (output_format_context, &output_packet)) < 0)
-    {
-#if DEBUG
-      fprintf (stderr, "Could not write frame (error '%s')\n",
-               get_error_text (error));
-#endif
-      av_packet_unref (&output_packet);
-      return error;
-    }
-    av_packet_unref (&output_packet);
-  }
-  return 0;
-}
-/**
- * Load one audio frame from the FIFO buffer, encode and write it to the
- * output file.
- */
-static int
-load_encode_and_write (AVAudioFifo *fifo,
-                       AVFormatContext *output_format_context,
-                       AVCodecContext *output_codec_context)
-{
-  /** Temporary storage of the output samples of the frame written to the file. */
-  AVFrame *output_frame;
-  /**
-   * Use the maximum number of possible samples per frame.
-   * If there is less than the maximum possible frame size in the FIFO
-   * buffer use this number. Otherwise, use the maximum possible frame size
-   */const int frame_size = FFMIN (av_audio_fifo_size (fifo),
-                                output_codec_context->frame_size);
-  int data_written;
-  /** Initialize temporary storage for one output frame. */
-  if (init_output_frame (&output_frame, output_codec_context, frame_size))
-    return AVERROR_EXIT;
-  /**
-   * Read as many samples from the FIFO buffer as required to fill the frame.
-   * The samples are stored in the frame temporarily.
-   */
-  if (av_audio_fifo_read (fifo, (void **) output_frame->data, frame_size) <
-      frame_size)
-  {
-#if DEBUG
-    fprintf (stderr, "Could not read data from FIFO\n");
-#endif
-    av_frame_free (&output_frame);
-    return AVERROR_EXIT;
-  }
-  /** Encode one frame worth of audio samples. */
-  if (encode_audio_frame (output_frame, output_format_context,
-                          output_codec_context, &data_written))
-  {
-    av_frame_free (&output_frame);
-    return AVERROR_EXIT;
-  }
-  av_frame_free (&output_frame);
-  return 0;
-}
-/** Write the trailer of the output file container. */
-static int
-write_output_file_trailer (AVFormatContext *output_format_context)
-{
-  int error;
-  if ((error = av_write_trailer (output_format_context)) < 0)
-  {
-#if DEBUG
-    fprintf (stderr, "Could not write output file trailer (error '%s')\n",
-             get_error_text (error));
-#endif
-    return error;
-  }
-  return 0;
-}
-#define ENUM_CODEC_ID enum AVCodecID
-/**
- * Perform the audio snippet extraction
- *
- * @param ec extraction context to use
- */
-static void
-extract_audio (struct EXTRACTOR_ExtractContext *ec)
-{
-  AVIOContext *io_ctx;
-  struct AVFormatContext *format_ctx;
-  AVCodecContext *codec_ctx;
-  AVFormatContext *output_format_context = NULL;
-  AVCodec *codec;
-  AVDictionary *options;
-  AVFrame *frame;
-  AVCodecContext*output_codec_context = NULL;
-  AVAudioResampleContext  *resample_context = NULL;
-  AVAudioFifo *fifo = NULL;
-  int audio_stream_index;
-  int i;
-  int err;
-  int duration;
-  unsigned char *iob;
-  totalSize = 0;
-  if (NULL == (iob = av_malloc (16 * 1024)))
-    return;
-  if (NULL == (io_ctx = avio_alloc_context (iob,
-                                            16 * 1024,
-                                            0, ec,
-                                            &read_cb,
-                                            NULL /* no writing */,
-                                            &seek_cb)))
-  {
-    av_free (iob);
-    return;
-  }
-  if (NULL == (format_ctx = avformat_alloc_context ()))
-  {
-    av_free (io_ctx);
-    return;
-  }
-  format_ctx->pb = io_ctx;
-  options = NULL;
-  if (0 != avformat_open_input (&format_ctx, "<no file>", NULL, &options))
-  {
-    av_free (io_ctx);
-    return;
-  }
-  av_dict_free (&options);
-  if (0 > avformat_find_stream_info (format_ctx, NULL))
-  {
-#if DEBUG
-    fprintf (stderr,
-             "Failed to read stream info\n");
-#endif
-    avformat_close_input (&format_ctx);
-    av_free (io_ctx);
-    return;
-  }
-  codec = NULL;
-  codec_ctx = NULL;
-  audio_stream_index = -1;
-  for (i = 0; i<format_ctx->nb_streams; i++)
-  {
-    codec_ctx = format_ctx->streams[i]->codec;
-    if (AVMEDIA_TYPE_AUDIO != codec_ctx->codec_type)
-      continue;
-    if (NULL == (codec = avcodec_find_decoder (codec_ctx->codec_id)))
-      continue;
-    options = NULL;
-    if (0 != (err = avcodec_open2 (codec_ctx, codec, &options)))
-    {
-      codec = NULL;
-      continue;
-    }
-    av_dict_free (&options);
-    audio_stream_index = i;
-    break;
-  }
-  if ( (-1 == audio_stream_index) ||
-       (0 == codec_ctx->channels) )
-  {
-#if DEBUG
-    fprintf (stderr,
-             "No audio streams or no suitable codec found\n");
-#endif
-    if (NULL != codec)
-      avcodec_close (codec_ctx);
-    avformat_close_input (&format_ctx);
-    av_free (io_ctx);
-    return;
-  }
-  frame = av_frame_alloc ();
-  if (NULL == frame)
-  {
-#if DEBUG
-    fprintf (stderr,
-             "Failed to allocate frame\n");
-#endif
-    avcodec_close (codec_ctx);
-    avformat_close_input (&format_ctx);
-    av_free (io_ctx);
-    return;
-  }
-  if (! (buffer = malloc (HARD_LIMIT_SIZE)))
-    goto cleanup;
-  /** Open the output file for writing. */
-  if (open_output_file (codec_ctx,
-                        &output_format_context,
-                        &output_codec_context))
-    goto cleanup;
-  /** Initialize the resampler to be able to convert audio sample formats. */
-  if (init_resampler (codec_ctx,
-                      output_codec_context,
-                      &resample_context))
-    goto cleanup;
-  /** Initialize the FIFO buffer to store audio samples to be encoded. */
-  if (init_fifo (&fifo))
-    goto cleanup;
-  /** Write the header of the output file container. */
-  if (write_output_file_header (output_format_context))
-    goto cleanup;
-  if (format_ctx->duration == AV_NOPTS_VALUE)
-  {
-    duration = -1;
-#if DEBUG
-    fprintf (stderr,
-             "Duration unknown\n");
-#endif
-  }
-  else
-  {
-    duration = format_ctx->duration;
-#if DEBUG
-    fprintf (stderr,
-             "Duration: %lld\n",
-             format_ctx->duration);
-#endif
-  }
-  /* if duration is known, seek to first tried,
-   * else use 10 sec into stream */
-  if (-1 != duration)
-    err = av_seek_frame (format_ctx, -1, (duration / 3), 0);
-  else
-    err = av_seek_frame (format_ctx, -1, 10 * AV_TIME_BASE, 0);
-  if (err >= 0)
-    avcodec_flush_buffers (codec_ctx);
-  /**
-   * Loop as long as we have input samples to read or output samples
-   * to write; abort as soon as we have neither.
-   */
-  while (1)
-  {
-    /** Use the encoder's desired frame size for processing. */
-    const int output_frame_size = output_codec_context->frame_size;
-    int finished                = 0;
-    /**
-     * Make sure that there is one frame worth of samples in the FIFO
-     * buffer so that the encoder can do its work.
-     * Since the decoder's and the encoder's frame size may differ, we
-     * need to FIFO buffer to store as many frames worth of input samples
-     * that they make up at least one frame worth of output samples.
-     */while ((av_audio_fifo_size (fifo) < output_frame_size))
-    {
-      /**
-       * Decode one frame worth of audio samples, convert it to the
-       * output sample format and put it into the FIFO buffer.
-       */
-      if (read_decode_convert_and_store (fifo,
-                                         format_ctx,
-                                         codec_ctx,
-                                         output_codec_context,
-                                         resample_context,
-                                         audio_stream_index,
-                                         &finished))
-      {
-        goto cleanup;
-      }
-      /**
-       * If we are at the end of the input file, we continue
-       * encoding the remaining audio samples to the output file.
-       */
-      if (finished)
-        break;
-    }
-    /* Already over our limit*/
-    if (totalSize >= MAX_SIZE)
-      finished = 1;
-    /**
-     * If we have enough samples for the encoder, we encode them.
-     * At the end of the file, we pass the remaining samples to
-     * the encoder.
-     *///
-    while (av_audio_fifo_size (fifo) >= output_frame_size ||
-           (finished && av_audio_fifo_size (fifo) > 0))
-    {
-      /**
-       * Take one frame worth of audio samples from the FIFO buffer,
-       * encode it and write it to the output file.
-       */
-      if (load_encode_and_write (fifo,
-                                 output_format_context,
-                                 output_codec_context))
-        goto cleanup;
-    }
-    /**
-     * If we are at the end of the input file and have encoded
-     * all remaining samples, we can exit this loop and finish.
-     */
-    if (finished)
-    {
-      int data_written;
-      /** Flush the encoder as it may have delayed frames. */
-      do {
-        encode_audio_frame (NULL,
-                            output_format_context,
-                            output_codec_context,
-                            &data_written);
-      } while (data_written);
-      break;
-    }
-  }
-  /** Write the trailer of the output file container. */
-  if (write_output_file_trailer (output_format_context))
-    goto cleanup;
-  ec->proc (ec->cls,
-            "previewopus",
-            EXTRACTOR_METATYPE_AUDIO_PREVIEW,
-            EXTRACTOR_METAFORMAT_BINARY,
-            "audio/opus",
-            buffer,
-            totalSize);
-#if OUTPUT_FILE
-  {
-    FILE *f;
-    f = fopen ("example.opus", "wb");
-    if (! f)
-    {
-      fprintf (stderr, "Could not open %s\n", "file");
-      exit (1);
-    }
-    fwrite (buffer, 1, totalSize, f);
-    fclose (f);
-  }
-#endif
-cleanup:
-  av_free (frame);
-  free (buffer);
-  if (fifo)
-    av_audio_fifo_free (fifo);
-  if (resample_context)
-  {
-    avresample_close (resample_context);
-    avresample_free (&resample_context);
-  }
-  if (output_codec_context)
-    avcodec_close (output_codec_context);
-  avcodec_close (codec_ctx);
-  avformat_close_input (&format_ctx);
-  av_free (io_ctx);
-}
-/**
- * Main method for the opus-preview plugin.
- *
- * @param ec extraction context
- */
-void
-EXTRACTOR_previewopus_extract_method (struct EXTRACTOR_ExtractContext *ec)
-{
-  ssize_t iret;
-  void *data;
-  if (-1 == (iret = ec->read (ec->cls,
-                              &data,
-                              16 * 1024)))
-    return;
-  if (0 != ec->seek (ec->cls, 0, SEEK_SET))
-    return;
-  extract_audio (ec);
-}
-/**
- * Log callback.  Does nothing.
- *
- * @param ptr NULL
- * @param level log level
- * @param format format string
- * @param ap arguments for format
- */
-static void
-previewopus_av_log_callback (void*ptr,
-                             int level,
-                             const char *format,
-                             va_list ap)
-{
-#if DEBUG
-  vfprintf (stderr, format, ap);
-#endif
-}
-/**
- * Initialize av-libs
- */
-void __attribute__ ((constructor))
-previewopus_lib_init (void)
-{
-  av_log_set_callback (&previewopus_av_log_callback);
-}
-/**
- * Destructor for the library, cleans up.
- */
-void __attribute__ ((destructor))
-previewopus_ltdl_fini ()
-{
-}
-/* end of previewopus_extractor.c */