opus: remove Opus encoder/decoder, moved to -base

https://bugzilla.gnome.org/show_bug.cgi?id=756282

1 files changed, 0 insertions, 819 deletions

diff --git a/ext/opus/gstopusdec.c b/ext/opus/gstopusdec.c
deleted file mode 100644
index 1470ea321..000000000
--- a/ext/opus/gstopusdec.c
+++ /dev/null
@@ -1,819 +0,0 @@
-/* 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
- * @see_also: opusenc, oggdemux
- *
- * This element decodes a OPUS stream to raw integer audio.
- *
- * <refsect2>
- * <title>Example pipelines</title>
- * |[
- * 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.
- * </refsect2>
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <math.h>
-#include <string.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 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);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&opus_dec_src_factory));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&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;
-}
-
-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 void
-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, channels;
-
-    caps = gst_caps_truncate (caps);
-    caps = gst_caps_make_writable (caps);
-    s = gst_caps_get_structure (caps, 0);
-
-    if (gst_structure_has_field (s, "rate"))
-      gst_structure_fixate_field_nearest_int (s, "rate", dec->sample_rate);
-    else
-      gst_structure_set (s, "rate", G_TYPE_INT, dec->sample_rate, NULL);
-    gst_structure_get_int (s, "rate", &rate);
-    dec->sample_rate = rate;
-
-    if (gst_structure_has_field (s, "channels"))
-      gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
-    else
-      gst_structure_set (s, "channels", G_TYPE_INT, dec->n_channels, NULL);
-    gst_structure_get_int (s, "channels", &channels);
-    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;
-}
-
-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_ERROR_OBJECT (dec, "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_ERROR_OBJECT (dec, "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);
-  }
-
-  gst_opus_dec_negotiate (dec, posn);
-
-  return GST_FLOW_OK;
-}
-
-
-static GstFlowReturn
-gst_opus_dec_parse_comments (GstOpusDec * dec, GstBuffer * buf)
-{
-  return GST_FLOW_OK;
-}
-
-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;
-
-      gst_opus_dec_negotiate (dec, NULL);
-    }
-
-    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);
-
-    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;
-  }
-
-  gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
-  out_data = (gint16 *) omap.data;
-
-  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);
-  }
-  gst_buffer_unmap (outbuf, &omap);
-  if (data != NULL)
-    gst_buffer_unmap (buf, &map);
-
-  if (n < 0) {
-    GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
-    gst_buffer_unref (outbuf);
-    return GST_FLOW_ERROR;
-  }
-  GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
-  gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);
-
-  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_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err);
-  return GST_FLOW_ERROR;
-
-buffer_failed:
-  GST_ERROR_OBJECT (dec, "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];
-
-    gst_opus_dec_negotiate (dec, posn);
-  }
-
-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;
-    } else {
-      res = opus_dec_chain_parse_data (dec, buf);
-    }
-  } else {
-    /* Otherwise fall back to packet counting and assume that the
-     * first two packets might be the headers, checking magic. */
-    switch (dec->packetno) {
-      case 0:
-        if (gst_opus_header_is_header (buf, "OpusHead", 8)) {
-          GST_DEBUG_OBJECT (dec, "found streamheader");
-          res = gst_opus_dec_parse_header (dec, buf);
-          gst_audio_decoder_finish_frame (adec, NULL, 1);
-        } else {
-          res = opus_dec_chain_parse_data (dec, buf);
-        }
-        break;
-      case 1:
-        if (gst_opus_header_is_header (buf, "OpusTags", 8)) {
-          GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
-          res = gst_opus_dec_parse_comments (dec, buf);
-          gst_audio_decoder_finish_frame (adec, NULL, 1);
-        } else {
-          res = opus_dec_chain_parse_data (dec, buf);
-        }
-        break;
-      default:
-      {
-        res = opus_dec_chain_parse_data (dec, buf);
-        break;
-      }
-    }
-  }
-
-  dec->packetno++;
-
-  return res;
-}
-
-static void
-gst_opus_dec_get_property (GObject * object, guint prop_id, GValue * value,
-    GParamSpec * pspec)
-{
-  GstOpusDec *dec = GST_OPUS_DEC (object);
-
-  switch (prop_id) {
-    case PROP_USE_INBAND_FEC:
-      g_value_set_boolean (value, dec->use_inband_fec);
-      break;
-    case PROP_APPLY_GAIN:
-      g_value_set_boolean (value, dec->apply_gain);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_opus_dec_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  GstOpusDec *dec = GST_OPUS_DEC (object);
-
-  switch (prop_id) {
-    case PROP_USE_INBAND_FEC:
-      dec->use_inband_fec = g_value_get_boolean (value);
-      break;
-    case PROP_APPLY_GAIN:
-      dec->apply_gain = g_value_get_boolean (value);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}