/* GStreamer

 * Copyright (C) 2004 Wim Taymans <wim@fluendo.com>

 * Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>

 * Copyright (C) 2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>

 * Copyright (C) 2011-2012 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

 *

 * 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.

 */

/*

 * Based on the speexdec element.

 */

/**

 * SECTION:element-opusdec

 * @title: opusdec

 * @see_also: opusenc, oggdemux

 *

 * This element decodes a OPUS stream to raw integer audio.

 *

 * ## Example pipelines

 * |[

 * gst-launch-1.0 -v filesrc location=opus.ogg ! oggdemux ! opusdec ! audioconvert ! audioresample ! alsasink

 * ]|

 * Decode an Ogg/Opus file. To create an Ogg/Opus file refer to the documentation of opusenc.

 *

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <math.h>

#include <string.h>

#include <stdio.h>

#include "gstopusheader.h"

#include "gstopuscommon.h"

#include "gstopusdec.h"

#include <gst/pbutils/pbutils.h>

GST_DEBUG_CATEGORY_STATIC (opusdec_debug);

#define GST_CAT_DEFAULT opusdec_debug

static GstStaticPadTemplate opus_dec_src_factory =

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) { 48000, 24000, 16000, 12000, 8000 }, "

        "channels = (int) [ 1, 8 ] ")

    );

static GstStaticPadTemplate opus_dec_sink_factory =

    GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("audio/x-opus, "

        "channel-mapping-family = (int) 0; "

        "audio/x-opus, "

        "channel-mapping-family = (int) [1, 255], "

        "channels = (int) [1, 255], "

        "stream-count = (int) [1, 255], " "coupled-count = (int) [0, 255]")

    );

G_DEFINE_TYPE (GstOpusDec, gst_opus_dec, GST_TYPE_AUDIO_DECODER);

#define DB_TO_LINEAR(x) pow (10., (x) / 20.)

#define DEFAULT_USE_INBAND_FEC FALSE

#define DEFAULT_APPLY_GAIN TRUE

enum

{

  PROP_0,

  PROP_USE_INBAND_FEC,

  PROP_APPLY_GAIN

};

static GstFlowReturn gst_opus_dec_parse_header (GstOpusDec * dec,

    GstBuffer * buf);

static gboolean gst_opus_dec_start (GstAudioDecoder * dec);

static gboolean gst_opus_dec_stop (GstAudioDecoder * dec);

static GstFlowReturn gst_opus_dec_handle_frame (GstAudioDecoder * dec,

    GstBuffer * buffer);

static gboolean gst_opus_dec_set_format (GstAudioDecoder * bdec,

    GstCaps * caps);

static void gst_opus_dec_get_property (GObject * object, guint prop_id,

    GValue * value, GParamSpec * pspec);

static void gst_opus_dec_set_property (GObject * object, guint prop_id,

    const GValue * value, GParamSpec * pspec);

static GstCaps *gst_opus_dec_getcaps (GstAudioDecoder * dec, GstCaps * filter);

static void

gst_opus_dec_class_init (GstOpusDecClass * klass)

{

  GObjectClass *gobject_class;

  GstAudioDecoderClass *adclass;

  GstElementClass *element_class;

  gobject_class = (GObjectClass *) klass;

  adclass = (GstAudioDecoderClass *) klass;

  element_class = (GstElementClass *) klass;

  gobject_class->set_property = gst_opus_dec_set_property;

  gobject_class->get_property = gst_opus_dec_get_property;

  adclass->start = GST_DEBUG_FUNCPTR (gst_opus_dec_start);

  adclass->stop = GST_DEBUG_FUNCPTR (gst_opus_dec_stop);

  adclass->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_dec_handle_frame);

  adclass->set_format = GST_DEBUG_FUNCPTR (gst_opus_dec_set_format);

  adclass->getcaps = GST_DEBUG_FUNCPTR (gst_opus_dec_getcaps);

  gst_element_class_add_static_pad_template (element_class,

      &opus_dec_src_factory);

  gst_element_class_add_static_pad_template (element_class,

      &opus_dec_sink_factory);

  gst_element_class_set_static_metadata (element_class, "Opus audio decoder",

      "Codec/Decoder/Audio", "decode opus streams to audio",

      "Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>");

  g_object_class_install_property (gobject_class, PROP_USE_INBAND_FEC,

      g_param_spec_boolean ("use-inband-fec", "Use in-band FEC",

          "Use forward error correction if available (needs PLC enabled)",

          DEFAULT_USE_INBAND_FEC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_APPLY_GAIN,

      g_param_spec_boolean ("apply-gain", "Apply gain",

          "Apply gain if any is specified in the header", DEFAULT_APPLY_GAIN,

          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  GST_DEBUG_CATEGORY_INIT (opusdec_debug, "opusdec", 0,

      "opus decoding element");

}

static void

gst_opus_dec_reset (GstOpusDec * dec)

{

  dec->packetno = 0;

  if (dec->state) {

    opus_multistream_decoder_destroy (dec->state);

    dec->state = NULL;

  }

  gst_buffer_replace (&dec->streamheader, NULL);

  gst_buffer_replace (&dec->vorbiscomment, NULL);

  gst_buffer_replace (&dec->last_buffer, NULL);

  dec->primed = FALSE;

  dec->pre_skip = 0;

  dec->r128_gain = 0;

  dec->sample_rate = 0;

  dec->n_channels = 0;

  dec->leftover_plc_duration = 0;

  dec->last_known_buffer_duration = GST_CLOCK_TIME_NONE;

}

static void

gst_opus_dec_init (GstOpusDec * dec)

{

  dec->use_inband_fec = FALSE;

  dec->apply_gain = DEFAULT_APPLY_GAIN;

  gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);

  gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST

      (dec), TRUE);

  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));

  gst_opus_dec_reset (dec);

}

static gboolean

gst_opus_dec_start (GstAudioDecoder * dec)

{

  GstOpusDec *odec = GST_OPUS_DEC (dec);

  gst_opus_dec_reset (odec);

  /* we know about concealment */

  gst_audio_decoder_set_plc_aware (dec, TRUE);

  if (odec->use_inband_fec) {

    /* opusdec outputs samples directly from an input buffer, except if

     * FEC is on, in which case it buffers one buffer in case one buffer

     * goes missing.

     */

    gst_audio_decoder_set_latency (dec, 120 * GST_MSECOND, 120 * GST_MSECOND);

  }

  return TRUE;

}

static gboolean

gst_opus_dec_stop (GstAudioDecoder * dec)

{

  GstOpusDec *odec = GST_OPUS_DEC (dec);

  gst_opus_dec_reset (odec);

  return TRUE;

}

static double

gst_opus_dec_get_r128_gain (gint16 r128_gain)

{

  return r128_gain / (double) (1 << 8);

}

static double

gst_opus_dec_get_r128_volume (gint16 r128_gain)

{

  return DB_TO_LINEAR (gst_opus_dec_get_r128_gain (r128_gain));

}

static gboolean

gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)

{

  GstCaps *caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));

  GstStructure *s;

  GstAudioInfo info;

  if (caps) {

    gint rate = dec->sample_rate, channels = dec->n_channels;

    GstCaps *constraint, *inter;

    constraint = gst_caps_from_string ("audio/x-raw");

    if (dec->n_channels <= 2) { /* including 0 */

      gst_caps_set_simple (constraint, "channels", GST_TYPE_INT_RANGE, 1, 2,

          NULL);

    } else {

      gst_caps_set_simple (constraint, "channels", G_TYPE_INT, dec->n_channels,

          NULL);

    }

    inter = gst_caps_intersect (caps, constraint);

    gst_caps_unref (constraint);

    if (gst_caps_is_empty (inter)) {

      GST_DEBUG_OBJECT (dec, "Empty intersection, failed to negotiate");

      gst_caps_unref (inter);

      gst_caps_unref (caps);

      return FALSE;

    }

    inter = gst_caps_truncate (inter);

    s = gst_caps_get_structure (inter, 0);

    rate = dec->sample_rate > 0 ? dec->sample_rate : 48000;

    gst_structure_fixate_field_nearest_int (s, "rate", dec->sample_rate);

    gst_structure_get_int (s, "rate", &rate;;

    channels = dec->n_channels > 0 ? dec->n_channels : 2;

    gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);

    gst_structure_get_int (s, "channels", &channels);

    gst_caps_unref (inter);

    dec->sample_rate = rate;

    dec->n_channels = channels;

    gst_caps_unref (caps);

  }

  if (dec->n_channels == 0) {

    GST_DEBUG_OBJECT (dec, "Using a default of 2 channels");

    dec->n_channels = 2;

    pos = NULL;

  }

  if (dec->sample_rate == 0) {

    GST_DEBUG_OBJECT (dec, "Using a default of 48kHz sample rate");

    dec->sample_rate = 48000;

  }

  GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,

      dec->sample_rate);

  /* pass valid order to audio info */

  if (pos) {

    memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);

    gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);

  }

  /* set up source format */

  gst_audio_info_init (&info;;

  gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,

      dec->sample_rate, dec->n_channels, pos ? dec->opus_pos : NULL);

  gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info;;

  /* but we still need the opus order for later reordering */

  if (pos) {

    memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);

  } else {

    dec->opus_pos[0] = GST_AUDIO_CHANNEL_POSITION_INVALID;

  }

  dec->info = info;

  return TRUE;

}

static GstFlowReturn

gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)

{

  GstAudioChannelPosition pos[64];

  const GstAudioChannelPosition *posn = NULL;

  if (!gst_opus_header_is_id_header (buf)) {

    GST_ELEMENT_ERROR (dec, STREAM, FORMAT, (NULL),

        ("Header is not an Opus ID header"));

    return GST_FLOW_ERROR;

  }

  if (!gst_codec_utils_opus_parse_header (buf,

          &dec->sample_rate,

          &dec->n_channels,

          &dec->channel_mapping_family,

          &dec->n_streams,

          &dec->n_stereo_streams,

          dec->channel_mapping, &dec->pre_skip, &dec->r128_gain)) {

    GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),

        ("Failed to parse Opus ID header"));

    return GST_FLOW_ERROR;

  }

  dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);

  GST_INFO_OBJECT (dec,

      "Found pre-skip of %u samples, R128 gain %d (volume %f)",

      dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);

  if (dec->channel_mapping_family == 1) {

    GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");

    switch (dec->n_channels) {

      case 1:

      case 2:

        /* nothing */

        break;

      case 3:

      case 4:

      case 5:

      case 6:

      case 7:

      case 8:

        posn = gst_opus_channel_positions[dec->n_channels - 1];

        break;

      default:{

        gint i;

        GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,

            (NULL), ("Using NONE channel layout for more than 8 channels"));

        for (i = 0; i < dec->n_channels; i++)

          pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;

        posn = pos;

      }

    }

  } else {

    GST_INFO_OBJECT (dec, "Channel mapping family %d",

        dec->channel_mapping_family);

  }

  if (!gst_opus_dec_negotiate (dec, posn))

    return GST_FLOW_NOT_NEGOTIATED;

  return GST_FLOW_OK;

}

static GstFlowReturn

gst_opus_dec_parse_comments (GstOpusDec * dec, GstBuffer * buf)

{

  return GST_FLOW_OK;

}

/* adapted from ext/ogg/gstoggstream.c */

static gint64

packet_duration_opus (const unsigned char *data, size_t bytes)

{

  static const guint64 durations[32] = {

    480, 960, 1920, 2880,       /* Silk NB */

    480, 960, 1920, 2880,       /* Silk MB */

    480, 960, 1920, 2880,       /* Silk WB */

    480, 960,                   /* Hybrid SWB */

    480, 960,                   /* Hybrid FB */

    120, 240, 480, 960,         /* CELT NB */

    120, 240, 480, 960,         /* CELT NB */

    120, 240, 480, 960,         /* CELT NB */

    120, 240, 480, 960,         /* CELT NB */

  };

  gint64 duration;

  gint64 frame_duration;

  gint nframes = 0;

  guint8 toc;

  if (bytes < 1)

    return 0;

  /* headers */

  if (bytes >= 8 && !memcmp (data, "Opus", 4))

    return 0;

  toc = data[0];

  frame_duration = durations[toc >> 3];

  switch (toc & 3) {

    case 0:

      nframes = 1;

      break;

    case 1:

      nframes = 2;

      break;

    case 2:

      nframes = 2;

      break;

    case 3:

      if (bytes < 2) {

        GST_WARNING ("Code 3 Opus packet has less than 2 bytes");

        return 0;

      }

      nframes = data[1] & 63;

      break;

  }

  duration = nframes * frame_duration;

  if (duration > 5760) {

    GST_WARNING ("Opus packet duration > 120 ms, invalid");

    return 0;

  }

  GST_LOG ("Opus packet: frame size %.1f ms, %d frames, duration %.1f ms",

      frame_duration / 48.f, nframes, duration / 48.f);

  return duration / 48.f * 1000000;

}

static GstFlowReturn

opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)

{

  GstFlowReturn res = GST_FLOW_OK;

  gsize size;

  guint8 *data;

  GstBuffer *outbuf, *bufd;

  gint16 *out_data;

  int n, err;

  int samples;

  unsigned int packet_size;

  GstBuffer *buf;

  GstMapInfo map, omap;

  GstAudioClippingMeta *cmeta = NULL;

  if (dec->state == NULL) {

    /* If we did not get any headers, default to 2 channels */

    if (dec->n_channels == 0) {

      GST_INFO_OBJECT (dec, "No header, assuming single stream");

      dec->n_channels = 2;

      dec->sample_rate = 48000;

      /* default stereo mapping */

      dec->channel_mapping_family = 0;

      dec->channel_mapping[0] = 0;

      dec->channel_mapping[1] = 1;

      dec->n_streams = 1;

      dec->n_stereo_streams = 1;

      if (!gst_opus_dec_negotiate (dec, NULL))

        return GST_FLOW_NOT_NEGOTIATED;

    }

    if (dec->n_channels == 2 && dec->n_streams == 1

        && dec->n_stereo_streams == 0) {

      /* if we are automatically decoding 2 channels, but only have

         a single encoded one, direct both channels to it */

      dec->channel_mapping[1] = 0;

    }

    GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",

        dec->n_channels, dec->sample_rate);

#ifndef GST_DISABLE_GST_DEBUG

    gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,

        "Mapping table", dec->n_channels, dec->channel_mapping);

#endif

    GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,

        dec->n_stereo_streams);

    dec->state =

        opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,

        dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err;;

    if (!dec->state || err != OPUS_OK)

      goto creation_failed;

  }

  if (buffer) {

    GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT,

        gst_buffer_get_size (buffer));

  } else {

    GST_DEBUG_OBJECT (dec, "Received missing buffer");

  }

  /* if using in-band FEC, we introdude one extra frame's delay as we need

     to potentially wait for next buffer to decode a missing buffer */

  if (dec->use_inband_fec && !dec->primed) {

    GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");

    gst_buffer_replace (&dec->last_buffer, buffer);

    dec->primed = TRUE;

    goto done;

  }

  /* That's the buffer we'll be sending to the opus decoder. */

  buf = (dec->use_inband_fec

      && gst_buffer_get_size (dec->last_buffer) >

      0) ? dec->last_buffer : buffer;

  /* That's the buffer we get duration from */

  bufd = dec->use_inband_fec ? dec->last_buffer : buffer;

  if (buf && gst_buffer_get_size (buf) > 0) {

    gst_buffer_map (buf, &map, GST_MAP_READ);

    data = map.data;

    size = map.size;

    GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size);

  } else {

    /* concealment data, pass NULL as the bits parameters */

    GST_DEBUG_OBJECT (dec, "Using NULL buffer");

    data = NULL;

    size = 0;

  }

  if (gst_buffer_get_size (bufd) == 0) {

    GstClockTime const opus_plc_alignment = 2500 * GST_USECOND;

    GstClockTime aligned_missing_duration;

    GstClockTime missing_duration = GST_BUFFER_DURATION (bufd);

    if (!GST_CLOCK_TIME_IS_VALID (missing_duration) || missing_duration == 0) {

      if (GST_CLOCK_TIME_IS_VALID (dec->last_known_buffer_duration)) {

        missing_duration = dec->last_known_buffer_duration;

        GST_WARNING_OBJECT (dec,

            "Missing duration, using last duration %" GST_TIME_FORMAT,

            GST_TIME_ARGS (missing_duration));

      } else {

        GST_WARNING_OBJECT (dec,

            "Missing buffer, but unknown duration, and no previously known duration, assuming 20 ms");

        missing_duration = 20 * GST_MSECOND;

      }

    }

    GST_DEBUG_OBJECT (dec,

        "missing buffer, doing PLC duration %" GST_TIME_FORMAT

        " plus leftover %" GST_TIME_FORMAT, GST_TIME_ARGS (missing_duration),

        GST_TIME_ARGS (dec->leftover_plc_duration));

    /* add the leftover PLC duration to that of the buffer */

    missing_duration += dec->leftover_plc_duration;

    /* align the combined buffer and leftover PLC duration to multiples

     * of 2.5ms, rounding to nearest, and store excess duration for later */

    aligned_missing_duration =

        ((missing_duration +

            opus_plc_alignment / 2) / opus_plc_alignment) * opus_plc_alignment;

    dec->leftover_plc_duration = missing_duration - aligned_missing_duration;

    /* Opus' PLC cannot operate with less than 2.5ms; skip PLC

     * and accumulate the missing duration in the leftover_plc_duration

     * for the next PLC attempt */

    if (aligned_missing_duration < opus_plc_alignment) {

      GST_DEBUG_OBJECT (dec,

          "current duration %" GST_TIME_FORMAT

          " of missing data not enough for PLC (minimum needed: %"

          GST_TIME_FORMAT ") - skipping", GST_TIME_ARGS (missing_duration),

          GST_TIME_ARGS (opus_plc_alignment));

      goto done;

    }

    /* convert the duration (in nanoseconds) to sample count */

    samples =

        gst_util_uint64_scale_int (aligned_missing_duration, dec->sample_rate,

        GST_SECOND);

    GST_DEBUG_OBJECT (dec,

        "calculated PLC frame length: %" GST_TIME_FORMAT

        " num frame samples: %d new leftover: %" GST_TIME_FORMAT,

        GST_TIME_ARGS (aligned_missing_duration), samples,

        GST_TIME_ARGS (dec->leftover_plc_duration));

  } else {

    /* use maximum size (120 ms) as the number of returned samples is

       not constant over the stream. */

    samples = 120 * dec->sample_rate / 1000;

  }

  packet_size = samples * dec->n_channels * 2;

  outbuf =

      gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),

      packet_size);

  if (!outbuf) {

    goto buffer_failed;

  }

  if (size > 0)

    dec->last_known_buffer_duration = packet_duration_opus (data, size);

  gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);

  out_data = (gint16 *) omap.data;

  do {

    if (dec->use_inband_fec) {

      if (gst_buffer_get_size (dec->last_buffer) > 0) {

        /* normal delayed decode */

        GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");

        n = opus_multistream_decode (dec->state, data, size, out_data, samples,

            0);

      } else {

        /* FEC reconstruction decode */

        GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");

        n = opus_multistream_decode (dec->state, data, size, out_data, samples,

            1);

      }

    } else {

      /* normal decode */

      GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");

      n = opus_multistream_decode (dec->state, data, size, out_data, samples,

          0);

    }

    if (n == OPUS_BUFFER_TOO_SMALL) {

      /* if too small, add 2.5 milliseconds and try again, up to the

       * Opus max size of 120 milliseconds */

      if (samples >= 120 * dec->sample_rate / 1000)

        break;

      samples += 25 * dec->sample_rate / 10000;

      packet_size = samples * dec->n_channels * 2;

      gst_buffer_unmap (outbuf, &omap;;

      gst_buffer_unref (outbuf);

      outbuf =

          gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),

          packet_size);

      if (!outbuf) {

        goto buffer_failed;

      }

      gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);

      out_data = (gint16 *) omap.data;

    }

  } while (n == OPUS_BUFFER_TOO_SMALL);

  gst_buffer_unmap (outbuf, &omap;;

  if (data != NULL)

    gst_buffer_unmap (buf, &map);

  if (n < 0) {

    GstFlowReturn ret = GST_FLOW_ERROR;

    gst_buffer_unref (outbuf);

    GST_AUDIO_DECODER_ERROR (dec, 1, STREAM, DECODE, (NULL),

        ("Decoding error (%d): %s", n, opus_strerror (n)), ret);

    return ret;

  }

  GST_DEBUG_OBJECT (dec, "decoded %d samples", n);

  gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);

  GST_BUFFER_DURATION (outbuf) = samples * GST_SECOND / dec->sample_rate;

  samples = n;

  cmeta = gst_buffer_get_audio_clipping_meta (buf);

  g_assert (!cmeta || cmeta->format == GST_FORMAT_DEFAULT);

  /* Skip any samples that need skipping */

  if (cmeta && cmeta->start) {

    guint pre_skip = cmeta->start;

    guint scaled_pre_skip = pre_skip * dec->sample_rate / 48000;

    guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;

    guint scaled_skip = skip * 48000 / dec->sample_rate;

    gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);

    GST_INFO_OBJECT (dec,

        "Skipping %u samples at the beginning (%u at 48000 Hz)",

        skip, scaled_skip);

  }

  if (cmeta && cmeta->end) {

    guint post_skip = cmeta->end;

    guint scaled_post_skip = post_skip * dec->sample_rate / 48000;

    guint skip = scaled_post_skip > n ? n : scaled_post_skip;

    guint scaled_skip = skip * 48000 / dec->sample_rate;

    guint outsize = gst_buffer_get_size (outbuf);

    guint skip_bytes = skip * 2 * dec->n_channels;

    if (outsize > skip_bytes)

      outsize -= skip_bytes;

    else

      outsize = 0;

    gst_buffer_resize (outbuf, 0, outsize);

    GST_INFO_OBJECT (dec,

        "Skipping %u samples at the end (%u at 48000 Hz)", skip, scaled_skip);

  }

  if (gst_buffer_get_size (outbuf) == 0) {

    gst_buffer_unref (outbuf);

    outbuf = NULL;

  } else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {

    gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,

        dec->n_channels, dec->opus_pos, dec->info.position);

  }

  /* Apply gain */

  /* Would be better off leaving this to a volume element, as this is

     a naive conversion that does too many int/float conversions.

     However, we don't have control over the pipeline...

     So make it optional if the user program wants to use a volume,

     but do it by default so the correct volume goes out by default */

  if (dec->apply_gain && outbuf && dec->r128_gain) {

    gsize rsize;

    unsigned int i, nsamples;

    double volume = dec->r128_gain_volume;

    gint16 *samples;

    gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);

    samples = (gint16 *) omap.data;

    rsize = omap.size;

    GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);

    nsamples = rsize / 2;

    for (i = 0; i < nsamples; ++i) {

      int sample = (int) (samples[i] * volume + 0.5);

      samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;

    }

    gst_buffer_unmap (outbuf, &omap;;

  }

  if (dec->use_inband_fec) {

    gst_buffer_replace (&dec->last_buffer, buffer);

  }

  res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);

  if (res != GST_FLOW_OK)

    GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));

done:

  return res;

creation_failed:

  GST_ELEMENT_ERROR (dec, LIBRARY, INIT, ("Failed to create Opus decoder"),

      ("Failed to create Opus decoder (%d): %s", err, opus_strerror (err)));

  return GST_FLOW_ERROR;

buffer_failed:

  GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),

      ("Failed to create %u byte buffer", packet_size));

  return GST_FLOW_ERROR;

}

static gboolean

gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)

{

  GstOpusDec *dec = GST_OPUS_DEC (bdec);

  gboolean ret = TRUE;

  GstStructure *s;

  const GValue *streamheader;

  GstCaps *old_caps;

  GST_DEBUG_OBJECT (dec, "set_format: %" GST_PTR_FORMAT, caps);

  if ((old_caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (bdec)))) {

    if (gst_caps_is_equal (caps, old_caps)) {

      gst_caps_unref (old_caps);

      GST_DEBUG_OBJECT (dec, "caps didn't change");

      goto done;

    }

    GST_DEBUG_OBJECT (dec, "caps have changed, resetting decoder");

    gst_opus_dec_reset (dec);

    gst_caps_unref (old_caps);

  }

  s = gst_caps_get_structure (caps, 0);

  if ((streamheader = gst_structure_get_value (s, "streamheader")) &&

      G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&

      gst_value_array_get_size (streamheader) >= 2) {

    const GValue *header, *vorbiscomment;

    GstBuffer *buf;

    GstFlowReturn res = GST_FLOW_OK;

    header = gst_value_array_get_value (streamheader, 0);

    if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {

      buf = gst_value_get_buffer (header);

      res = gst_opus_dec_parse_header (dec, buf);

      if (res != GST_FLOW_OK) {

        ret = FALSE;

        goto done;

      }

      gst_buffer_replace (&dec->streamheader, buf);

    }

    vorbiscomment = gst_value_array_get_value (streamheader, 1);

    if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {

      buf = gst_value_get_buffer (vorbiscomment);

      res = gst_opus_dec_parse_comments (dec, buf);

      if (res != GST_FLOW_OK) {

        ret = FALSE;

        goto done;

      }

      gst_buffer_replace (&dec->vorbiscomment, buf);

    }

  } else {

    const GstAudioChannelPosition *posn = NULL;

    if (!gst_codec_utils_opus_parse_caps (caps, &dec->sample_rate,

            &dec->n_channels, &dec->channel_mapping_family, &dec->n_streams,

            &dec->n_stereo_streams, dec->channel_mapping)) {

      ret = FALSE;

      goto done;

    }

    if (dec->channel_mapping_family == 1 && dec->n_channels <= 8)

      posn = gst_opus_channel_positions[dec->n_channels - 1];

    if (!gst_opus_dec_negotiate (dec, posn))

      return FALSE;

  }

done:

  return ret;

}

static gboolean

memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)

{

  gsize size1, size2;

  gboolean res;

  GstMapInfo map;

  size1 = gst_buffer_get_size (buf1);

  size2 = gst_buffer_get_size (buf2);

  if (size1 != size2)

    return FALSE;

  gst_buffer_map (buf1, &map, GST_MAP_READ);

  res = gst_buffer_memcmp (buf2, 0, map.data, map.size) == 0;

  gst_buffer_unmap (buf1, &map);

  return res;

}

static GstFlowReturn

gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)

{

  GstFlowReturn res;

  GstOpusDec *dec;

  /* no fancy draining */

  if (G_UNLIKELY (!buf))

    return GST_FLOW_OK;

  dec = GST_OPUS_DEC (adec);

  GST_LOG_OBJECT (dec,

      "Got buffer ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT,

      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),

      GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));

  /* If we have the streamheader and vorbiscomment from the caps already

   * ignore them here */

  if (dec->streamheader && dec->vorbiscomment) {

    if (memcmp_buffers (dec->streamheader, buf)) {

      GST_DEBUG_OBJECT (dec, "found streamheader");

      gst_audio_decoder_finish_frame (adec, NULL, 1);

      res = GST_FLOW_OK;

    } else if (memcmp_buffers (dec->vorbiscomment, buf)) {

      GST_DEBUG_OBJECT (dec, "found vorbiscomments");

      gst_audio_decoder_finish_frame (adec, NULL, 1);

      res = GST_FLOW_OK;