/* returns TRUE on success, with byte_offset set to the offset of the
 * wavpack chunk containing the sample requested. start_sample will be
 * set to the first sample in the chunk starting at byte_offset.
 * Scanning from the last known header offset to the wanted position
 * when seeking forward isn't very clever, but seems fast enough in
 * practice and has the nice side effect of populating our index
 * table */
static gboolean
gst_wavpack_parse_scan_to_find_sample (GstWavpackParse * parse,
    gint64 sample, gint64 * byte_offset, gint64 * start_sample)
{
  GstWavpackParseIndexEntry *entry;
  GstFlowReturn ret;
  gint64 off = 0;

  /* first, check if we have to scan at all */
  entry = gst_wavpack_parse_index_get_entry_from_sample (parse, sample);
  if (entry) {
    *byte_offset = entry->byte_offset;
    *start_sample = entry->sample_offset;
    GST_LOG_OBJECT (parse, "Found index entry: sample %" G_GINT64_FORMAT
        " @ offset %" G_GINT64_FORMAT, entry->sample_offset,
        entry->byte_offset);
    return TRUE;
  }

  GST_LOG_OBJECT (parse, "No matching entry in index, scanning file ...");

  /* if we have an index, we can start scanning from the last known offset
   * in there, after all we know our wanted sample is not in the index */
  if (parse->entries) {
    GstWavpackParseIndexEntry *entry;

    entry = gst_wavpack_parse_index_get_last_entry (parse);
    off = entry->byte_offset;
  }

  /* now scan forward until we find the chunk we're looking for or hit EOS */
  do {
    WavpackHeader header;
    GstBuffer *buf;

    buf = gst_wavpack_parse_pull_buffer (parse, off, sizeof (WavpackHeader),
        &ret);

    if (buf == NULL)
      break;

    gst_wavpack_read_header (&header, GST_BUFFER_DATA (buf));
    gst_buffer_unref (buf);

    if (header.flags & INITIAL_BLOCK)
      gst_wavpack_parse_index_append_entry (parse, off, header.block_index,
          header.block_samples);
    else
      continue;

    if (header.block_index <= sample &&
        sample < (header.block_index + header.block_samples)) {
      *byte_offset = off;
      *start_sample = header.block_index;
      return TRUE;
    }

    off += header.ckSize + 8;
  } while (1);

  GST_DEBUG_OBJECT (parse, "scan failed: %s (off=0x%08" G_GINT64_MODIFIER "x)",
      gst_flow_get_name (ret), off);

  return FALSE;
}

static GstCaps *
gst_rtp_h264_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
    GstCaps * filter)
{
  GstCaps *template_caps;
  GstCaps *allowed_caps;
  GstCaps *caps, *icaps;
  gboolean append_unrestricted;
  guint i;

  allowed_caps =
      gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), NULL);

  if (allowed_caps == NULL)
    return NULL;

  template_caps =
      gst_static_pad_template_get_caps (&gst_rtp_h264_pay_sink_template);

  if (gst_caps_is_any (allowed_caps)) {
    caps = gst_caps_ref (template_caps);
    goto done;
  }

  if (gst_caps_is_empty (allowed_caps)) {
    caps = gst_caps_ref (allowed_caps);
    goto done;
  }

  caps = gst_caps_new_empty ();

  append_unrestricted = FALSE;
  for (i = 0; i < gst_caps_get_size (allowed_caps); i++) {
    GstStructure *s = gst_caps_get_structure (allowed_caps, i);
    GstStructure *new_s = gst_structure_new_empty ("video/x-h264");
    const gchar *profile_level_id;

    profile_level_id = gst_structure_get_string (s, "profile-level-id");

    if (profile_level_id && strlen (profile_level_id) == 6) {
      const gchar *profile;
      const gchar *level;
      long int spsint;
      guint8 sps[3];

      spsint = strtol (profile_level_id, NULL, 16);
      sps[0] = spsint >> 16;
      sps[1] = spsint >> 8;
      sps[2] = spsint;

      profile = gst_codec_utils_h264_get_profile (sps, 3);
      level = gst_codec_utils_h264_get_level (sps, 3);

      if (profile && level) {
        GST_LOG_OBJECT (payload, "In caps, have profile %s and level %s",
            profile, level);

        if (!strcmp (profile, "constrained-baseline"))
          gst_structure_set (new_s, "profile", G_TYPE_STRING, profile, NULL);
        else {
          GValue val = { 0, };
          GValue profiles = { 0, };

          g_value_init (&profiles, GST_TYPE_LIST);
          g_value_init (&val, G_TYPE_STRING);

          g_value_set_static_string (&val, profile);
          gst_value_list_append_value (&profiles, &val);

          g_value_set_static_string (&val, "constrained-baseline");
          gst_value_list_append_value (&profiles, &val);

          gst_structure_take_value (new_s, "profile", &profiles);
        }

        if (!strcmp (level, "1"))
          gst_structure_set (new_s, "level", G_TYPE_STRING, level, NULL);
        else {
          GValue levels = { 0, };
          GValue val = { 0, };
          int j;

          g_value_init (&levels, GST_TYPE_LIST);
          g_value_init (&val, G_TYPE_STRING);

          for (j = 0; j < G_N_ELEMENTS (all_levels); j++) {
            g_value_set_static_string (&val, all_levels[j]);
            gst_value_list_prepend_value (&levels, &val);
            if (!strcmp (level, all_levels[j]))
              break;
          }
          gst_structure_take_value (new_s, "level", &levels);
        }
      } else {
        /* Invalid profile-level-id means baseline */

        gst_structure_set (new_s,
            "profile", G_TYPE_STRING, "constrained-baseline", NULL);
      }
    } else {
//.........这里部分代码省略.........

static gboolean
gst_soup_http_client_sink_start (GstBaseSink * sink)
{
  GstSoupHttpClientSink *souphttpsink = GST_SOUP_HTTP_CLIENT_SINK (sink);

  if (souphttpsink->prop_session) {
    souphttpsink->session = souphttpsink->prop_session;
  } else {
    GSource *source;
    GError *error = NULL;

    souphttpsink->context = g_main_context_new ();

    /* set up idle source to signal when the main loop is running and
     * it's safe for ::stop() to call g_main_loop_quit() */
    source = g_idle_source_new ();
    g_source_set_callback (source, thread_ready_idle_cb, sink, NULL);
    g_source_attach (source, souphttpsink->context);
    g_source_unref (source);

    souphttpsink->loop = g_main_loop_new (souphttpsink->context, TRUE);

    g_mutex_lock (&souphttpsink->mutex);

    souphttpsink->thread = g_thread_try_new ("souphttpclientsink-thread",
        thread_func, souphttpsink, &error);

    if (error != NULL) {
      GST_DEBUG_OBJECT (souphttpsink, "failed to start thread, %s",
          error->message);
      g_error_free (error);
      g_mutex_unlock (&souphttpsink->mutex);
      return FALSE;
    }

    GST_LOG_OBJECT (souphttpsink, "waiting for main loop thread to start up");
    g_cond_wait (&souphttpsink->cond, &souphttpsink->mutex);
    g_mutex_unlock (&souphttpsink->mutex);
    GST_LOG_OBJECT (souphttpsink, "main loop thread running");

    if (souphttpsink->proxy == NULL) {
      souphttpsink->session =
          soup_session_async_new_with_options (SOUP_SESSION_ASYNC_CONTEXT,
          souphttpsink->context, SOUP_SESSION_USER_AGENT,
          souphttpsink->user_agent, SOUP_SESSION_TIMEOUT, souphttpsink->timeout,
          NULL);
    } else {
      souphttpsink->session =
          soup_session_async_new_with_options (SOUP_SESSION_ASYNC_CONTEXT,
          souphttpsink->context, SOUP_SESSION_USER_AGENT,
          souphttpsink->user_agent, SOUP_SESSION_TIMEOUT, souphttpsink->timeout,
          SOUP_SESSION_PROXY_URI, souphttpsink->proxy, NULL);
    }

    g_signal_connect (souphttpsink->session, "authenticate",
        G_CALLBACK (authenticate), souphttpsink);
  }

  /* Set up logging */
  gst_soup_util_log_setup (souphttpsink->session, souphttpsink->log_level,
      GST_ELEMENT (souphttpsink));

  return TRUE;
}

static void
gst_gnome_vfs_src_received_headers_callback (gconstpointer in,
    gsize in_size, gpointer out, gsize out_size, gpointer callback_data)
{
  GList *i;
  gint icy_metaint;
  GstGnomeVFSSrc *src = GST_GNOME_VFS_SRC (callback_data);
  GnomeVFSModuleCallbackReceivedHeadersIn *in_args =
      (GnomeVFSModuleCallbackReceivedHeadersIn *) in;

  /* This is only used for internet radio stuff right now */
  if (!src->iradio_mode)
    return;

  GST_DEBUG_OBJECT (src, "receiving internet radio metadata\n");

  /* FIXME: Could we use "Accept-Ranges: bytes"
   * http://www.w3.org/Protocols/rfc2616/rfc2616-sec14.html#sec14.5
   * to enable pull-mode?
   */

  for (i = in_args->headers; i; i = i->next) {
    char *data = (char *) i->data;
    char *key = data;
    char *value = strchr (data, ':');

    if (!value)
      continue;

    value++;
    g_strstrip (value);
    if (!strlen (value))
      continue;

    GST_LOG_OBJECT (src, "data %s", data);

    /* Icecast stuff */
    if (strncmp (data, "icy-metaint:", 12) == 0) {      /* ugh */
      if (sscanf (data + 12, "%d", &icy_metaint) == 1) {
        if (icy_metaint > 0) {
          GstCaps *icy_caps;

          icy_caps = gst_caps_new_simple ("application/x-icy",
              "metadata-interval", G_TYPE_INT, icy_metaint, NULL);
          gst_pad_set_caps (GST_BASE_SRC_PAD (src), icy_caps);
          gst_caps_unref (icy_caps);
        }
      }
      continue;
    }

    if (!strncmp (data, "icy-", 4))
      key = data + 4;
    else
      continue;

    GST_DEBUG_OBJECT (src, "key: %s", key);
    if (!strncmp (key, "name", 4)) {
      g_free (src->iradio_name);
      src->iradio_name = gst_gnome_vfs_src_unicodify (value);
      if (src->iradio_name)
        g_object_notify (G_OBJECT (src), "iradio-name");
    } else if (!strncmp (key, "genre", 5)) {
      g_free (src->iradio_genre);
      src->iradio_genre = gst_gnome_vfs_src_unicodify (value);
      if (src->iradio_genre)
        g_object_notify (G_OBJECT (src), "iradio-genre");
    } else if (!strncmp (key, "url", 3)) {
      g_free (src->iradio_url);
      src->iradio_url = gst_gnome_vfs_src_unicodify (value);
      if (src->iradio_url)
        g_object_notify (G_OBJECT (src), "iradio-url");
    }
  }
}

static gboolean
gst_raw_parse_handle_seek_pull (GstRawParse * rp, GstEvent * event)
{
  gdouble rate;
  GstFormat format;
  GstSeekFlags flags;
  GstSeekType start_type, stop_type;
  gint64 start, stop;
  gint64 last_stop;
  gboolean ret = FALSE;
  gboolean flush;
  GstSegment seeksegment;

  if (event) {
    gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
        &stop_type, &stop);

    /* convert input offsets to time */
    ret = gst_raw_parse_convert (rp, format, start, GST_FORMAT_TIME, &start);
    ret &= gst_raw_parse_convert (rp, format, stop, GST_FORMAT_TIME, &stop);
    if (!ret)
      goto convert_failed;

    GST_DEBUG_OBJECT (rp, "converted start - stop to time");

    format = GST_FORMAT_TIME;

    gst_event_unref (event);
  } else {
    format = GST_FORMAT_TIME;
    flags = 0;
  }

  flush = ((flags & GST_SEEK_FLAG_FLUSH) != 0);

  /* start flushing up and downstream so that the loop function pauses and we
   * can acquire the STREAM_LOCK. */
  if (flush) {
    GST_LOG_OBJECT (rp, "flushing");
    gst_pad_push_event (rp->sinkpad, gst_event_new_flush_start ());
    gst_pad_push_event (rp->srcpad, gst_event_new_flush_start ());
  } else {
    GST_LOG_OBJECT (rp, "pause task");
    gst_pad_pause_task (rp->sinkpad);
  }

  GST_PAD_STREAM_LOCK (rp->sinkpad);

  memcpy (&seeksegment, &rp->segment, sizeof (GstSegment));

  if (event) {
    /* configure the seek values */
    gst_segment_do_seek (&seeksegment, rate, format, flags,
        start_type, start, stop_type, stop, NULL);
  }

  /* get the desired position */
  last_stop = seeksegment.position;

  GST_LOG_OBJECT (rp, "seeking to %" GST_TIME_FORMAT,
      GST_TIME_ARGS (last_stop));

  /* convert the desired position to bytes */
  ret =
      gst_raw_parse_convert (rp, format, last_stop, GST_FORMAT_BYTES,
      &last_stop);

  /* prepare for streaming */
  if (flush) {
    GST_LOG_OBJECT (rp, "stop flush");
    gst_pad_push_event (rp->sinkpad, gst_event_new_flush_stop (TRUE));
    gst_pad_push_event (rp->srcpad, gst_event_new_flush_stop (TRUE));
  }

  if (ret) {
    /* seek done */

    /* Seek on a frame boundary */
    last_stop -= last_stop % rp->framesize;

    rp->offset = last_stop;
    rp->n_frames = last_stop / rp->framesize;

    GST_LOG_OBJECT (rp, "seeking to bytes %" G_GINT64_FORMAT, last_stop);

    memcpy (&rp->segment, &seeksegment, sizeof (GstSegment));

    if (rp->segment.flags & GST_SEEK_FLAG_SEGMENT) {
      gst_element_post_message (GST_ELEMENT_CAST (rp),
          gst_message_new_segment_start (GST_OBJECT_CAST (rp),
              rp->segment.format, rp->segment.position));
    }

    /* for deriving a stop position for the playback segment from the seek
     * segment, we must take the duration when the stop is not set */
    if ((stop = rp->segment.stop) == -1)
      stop = rp->segment.duration;

    GST_DEBUG_OBJECT (rp, "preparing newsegment from %" G_GINT64_FORMAT
        " to %" G_GINT64_FORMAT, rp->segment.start, stop);
//.........这里部分代码省略.........

static GstFlowReturn
gst_tcp_server_src_create (GstPushSrc * psrc, GstBuffer ** outbuf)
{
  GstTCPServerSrc *src;
  GstFlowReturn ret = GST_FLOW_OK;
  gssize rret, avail;
  gsize read;
  GError *err = NULL;
  GstMapInfo map;

  src = GST_TCP_SERVER_SRC (psrc);

  if (!GST_OBJECT_FLAG_IS_SET (src, GST_TCP_SERVER_SRC_OPEN))
    goto wrong_state;

  if (!src->client_socket) {
    /* wait on server socket for connections */
    src->client_socket =
        g_socket_accept (src->server_socket, src->cancellable, &err);
    if (!src->client_socket)
      goto accept_error;
    /* now read from the socket. */
  }

  /* if we have a client, wait for read */
  GST_LOG_OBJECT (src, "asked for a buffer");

  /* read the buffer header */
  avail = g_socket_get_available_bytes (src->client_socket);
  if (avail < 0) {
    goto get_available_error;
  } else if (avail == 0) {
    GIOCondition condition;

    if (!g_socket_condition_wait (src->client_socket,
            G_IO_IN | G_IO_PRI | G_IO_ERR | G_IO_HUP, src->cancellable, &err))
      goto select_error;

    condition =
        g_socket_condition_check (src->client_socket,
        G_IO_IN | G_IO_PRI | G_IO_ERR | G_IO_HUP);

    if ((condition & G_IO_ERR)) {
      GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
          ("Socket in error state"));
      *outbuf = NULL;
      ret = GST_FLOW_ERROR;
      goto done;
    } else if ((condition & G_IO_HUP)) {
      GST_DEBUG_OBJECT (src, "Connection closed");
      *outbuf = NULL;
      ret = GST_FLOW_EOS;
      goto done;
    }
    avail = g_socket_get_available_bytes (src->client_socket);
    if (avail < 0)
      goto get_available_error;
  }

  if (avail > 0) {
    read = MIN (avail, MAX_READ_SIZE);
    *outbuf = gst_buffer_new_and_alloc (read);
    gst_buffer_map (*outbuf, &map, GST_MAP_READWRITE);
    rret =
        g_socket_receive (src->client_socket, (gchar *) map.data, read,
        src->cancellable, &err);
  } else {
    /* Connection closed */
    rret = 0;
    *outbuf = NULL;
    read = 0;
  }

  if (rret == 0) {
    GST_DEBUG_OBJECT (src, "Connection closed");
    ret = GST_FLOW_EOS;
    if (*outbuf) {
      gst_buffer_unmap (*outbuf, &map);
      gst_buffer_unref (*outbuf);
    }
    *outbuf = NULL;
  } else if (rret < 0) {
    if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_CANCELLED)) {
      ret = GST_FLOW_FLUSHING;
      GST_DEBUG_OBJECT (src, "Cancelled reading from socket");
    } else {
      ret = GST_FLOW_ERROR;
      GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
          ("Failed to read from socket: %s", err->message));
    }
    gst_buffer_unmap (*outbuf, &map);
    gst_buffer_unref (*outbuf);
    *outbuf = NULL;
  } else {
    ret = GST_FLOW_OK;
    gst_buffer_unmap (*outbuf, &map);
    gst_buffer_resize (*outbuf, 0, rret);

    GST_LOG_OBJECT (src,
        "Returning buffer from _get of size %" G_GSIZE_FORMAT ", ts %"
//.........这里部分代码省略.........

static GstFlowReturn
gst_spectrum_transform_ip (GstBaseTransform * trans, GstBuffer * buffer)
{
  GstSpectrum *spectrum = GST_SPECTRUM (trans);
  guint rate = GST_AUDIO_FILTER_RATE (spectrum);
  guint channels = GST_AUDIO_FILTER_CHANNELS (spectrum);
  guint bps = GST_AUDIO_FILTER_BPS (spectrum);
  guint bpf = GST_AUDIO_FILTER_BPF (spectrum);
  guint output_channels = spectrum->multi_channel ? channels : 1;
  guint c;
  gfloat max_value = (1UL << ((bps << 3) - 1)) - 1;
  guint bands = spectrum->bands;
  guint nfft = 2 * bands - 2;
  guint input_pos;
  gfloat *input;
  GstMapInfo map;
  const guint8 *data;
  gsize size;
  guint fft_todo, msg_todo, block_size;
  gboolean have_full_interval;
  GstSpectrumChannel *cd;
  GstSpectrumInputData input_data;

  g_mutex_lock (&spectrum->lock);
  gst_buffer_map (buffer, &map, GST_MAP_READ);
  data = map.data;
  size = map.size;

  GST_LOG_OBJECT (spectrum, "input size: %" G_GSIZE_FORMAT " bytes", size);

  if (GST_BUFFER_IS_DISCONT (buffer)) {
    GST_DEBUG_OBJECT (spectrum, "Discontinuity detected -- flushing");
    gst_spectrum_flush (spectrum);
  }

  /* If we don't have a FFT context yet (or it was reset due to parameter
   * changes) get one and allocate memory for everything
   */
  if (spectrum->channel_data == NULL) {
    GST_DEBUG_OBJECT (spectrum, "allocating for bands %u", bands);

    gst_spectrum_alloc_channel_data (spectrum);

    /* number of sample frames we process before posting a message
     * interval is in ns */
    spectrum->frames_per_interval =
        gst_util_uint64_scale (spectrum->interval, rate, GST_SECOND);
    spectrum->frames_todo = spectrum->frames_per_interval;
    /* rounding error for frames_per_interval in ns,
     * aggregated it in accumulated_error */
    spectrum->error_per_interval = (spectrum->interval * rate) % GST_SECOND;
    if (spectrum->frames_per_interval == 0)
      spectrum->frames_per_interval = 1;

    GST_INFO_OBJECT (spectrum, "interval %" GST_TIME_FORMAT ", fpi %"
        G_GUINT64_FORMAT ", error %" GST_TIME_FORMAT,
        GST_TIME_ARGS (spectrum->interval), spectrum->frames_per_interval,
        GST_TIME_ARGS (spectrum->error_per_interval));

    spectrum->input_pos = 0;

    gst_spectrum_flush (spectrum);
  }

  if (spectrum->num_frames == 0)
    spectrum->message_ts = GST_BUFFER_TIMESTAMP (buffer);

  input_pos = spectrum->input_pos;
  input_data = spectrum->input_data;

  while (size >= bpf) {
    /* run input_data for a chunk of data */
    fft_todo = nfft - (spectrum->num_frames % nfft);
    msg_todo = spectrum->frames_todo - spectrum->num_frames;
    GST_LOG_OBJECT (spectrum,
        "message frames todo: %u, fft frames todo: %u, input frames %"
        G_GSIZE_FORMAT, msg_todo, fft_todo, (size / bpf));
    block_size = msg_todo;
    if (block_size > (size / bpf))
      block_size = (size / bpf);
    if (block_size > fft_todo)
      block_size = fft_todo;

    for (c = 0; c < output_channels; c++) {
      cd = &spectrum->channel_data[c];
      input = cd->input;
      /* Move the current frames into our ringbuffers */
      input_data (data + c * bps, input, block_size, channels, max_value,
          input_pos, nfft);
    }
    data += block_size * bpf;
    size -= block_size * bpf;
    input_pos = (input_pos + block_size) % nfft;
    spectrum->num_frames += block_size;

    have_full_interval = (spectrum->num_frames == spectrum->frames_todo);

    GST_LOG_OBJECT (spectrum,
        "size: %" G_GSIZE_FORMAT ", do-fft = %d, do-message = %d", size,
        (spectrum->num_frames % nfft == 0), have_full_interval);
//.........这里部分代码省略.........

static GstFlowReturn
vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet,
                           GstClockTime timestamp, GstClockTime duration)
{
#ifdef USE_TREMOLO
    vorbis_sample_t *pcm;
#else
    vorbis_sample_t **pcm;
#endif
    guint sample_count;
    GstBuffer *out = NULL;
    GstFlowReturn result;
    gint size;

    if (G_UNLIKELY (!vd->initialized)) {
        result = vorbis_dec_handle_header_caps (vd);
        if (result != GST_FLOW_OK)
            goto not_initialized;
    }

    /* normal data packet */
    /* FIXME, we can skip decoding if the packet is outside of the
     * segment, this is however not very trivial as we need a previous
     * packet to decode the current one so we must be careful not to
     * throw away too much. For now we decode everything and clip right
     * before pushing data. */

#ifdef USE_TREMOLO
    if (G_UNLIKELY (vorbis_dsp_synthesis (&vd->vd, packet, 1)))
        goto could_not_read;
#else
    if (G_UNLIKELY (vorbis_synthesis (&vd->vb, packet)))
        goto could_not_read;

    if (G_UNLIKELY (vorbis_synthesis_blockin (&vd->vd, &vd->vb) < 0))
        goto not_accepted;
#endif

    /* assume all goes well here */
    result = GST_FLOW_OK;

    /* count samples ready for reading */
#ifdef USE_TREMOLO
    if ((sample_count = vorbis_dsp_pcmout (&vd->vd, NULL, 0)) == 0)
#else
    if ((sample_count = vorbis_synthesis_pcmout (&vd->vd, NULL)) == 0)
        goto done;
#endif

        size = sample_count * vd->vi.channels * vd->width;
    GST_LOG_OBJECT (vd, "%d samples ready for reading, size %d", sample_count,
                    size);

    /* alloc buffer for it */
    result =
        gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (vd),
                                           GST_BUFFER_OFFSET_NONE, size,
                                           GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (vd)), &out);
    if (G_UNLIKELY (result != GST_FLOW_OK))
        goto done;

    /* get samples ready for reading now, should be sample_count */
#ifdef USE_TREMOLO
    pcm = GST_BUFFER_DATA (out);
    if (G_UNLIKELY (vorbis_dsp_pcmout (&vd->vd, pcm, sample_count) !=
                    sample_count))
#else
    if (G_UNLIKELY (vorbis_synthesis_pcmout (&vd->vd, &pcm) != sample_count))
#endif
        goto wrong_samples;

#ifndef USE_TREMOLO
    /* copy samples in buffer */
    vd->copy_samples ((vorbis_sample_t *) GST_BUFFER_DATA (out), pcm,
                      sample_count, vd->vi.channels, vd->width);
#endif

    GST_LOG_OBJECT (vd, "setting output size to %d", size);
    GST_BUFFER_SIZE (out) = size;

done:
    /* whether or not data produced, consume one frame and advance time */
    result = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (vd), out, 1);

#ifdef USE_TREMOLO
    vorbis_dsp_read (&vd->vd, sample_count);
#else
    vorbis_synthesis_read (&vd->vd, sample_count);
#endif

    return result;

    /* ERRORS */
not_initialized:
    {
        GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
                           (NULL), ("no header sent yet"));
        return GST_FLOW_NOT_NEGOTIATED;
    }
could_not_read:
//.........这里部分代码省略.........

static GstFlowReturn
vorbis_dec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{
    ogg_packet *packet;
    ogg_packet_wrapper packet_wrapper;
    GstFlowReturn result = GST_FLOW_OK;
    GstVorbisDec *vd = GST_VORBIS_DEC (dec);

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

    /* make ogg_packet out of the buffer */
    gst_ogg_packet_wrapper_from_buffer (&packet_wrapper, buffer);
    packet = gst_ogg_packet_from_wrapper (&packet_wrapper);
    /* set some more stuff */
    packet->granulepos = -1;
    packet->packetno = 0;         /* we don't care */
    /* EOS does not matter, it is used in vorbis to implement clipping the last
     * block of samples based on the granulepos. We clip based on segments. */
    packet->e_o_s = 0;

    GST_LOG_OBJECT (vd, "decode buffer of size %ld", packet->bytes);

    /* error out on empty header packets, but just skip empty data packets */
    if (G_UNLIKELY (packet->bytes == 0)) {
        if (vd->initialized)
            goto empty_buffer;
        else
            goto empty_header;
    }

    /* switch depending on packet type */
    if ((gst_ogg_packet_data (packet))[0] & 1) {
        if (vd->initialized) {
            GST_WARNING_OBJECT (vd, "Already initialized, so ignoring header packet");
            goto done;
        }
        result = vorbis_handle_header_packet (vd, packet);
        /* consumer header packet/frame */
        gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (vd), NULL, 1);
    } else {
        GstClockTime timestamp, duration;

        timestamp = GST_BUFFER_TIMESTAMP (buffer);
        duration = GST_BUFFER_DURATION (buffer);

        result = vorbis_handle_data_packet (vd, packet, timestamp, duration);
    }

done:
    return result;

empty_buffer:
    {
        /* don't error out here, just ignore the buffer, it's invalid for vorbis
         * but not fatal. */
        GST_WARNING_OBJECT (vd, "empty buffer received, ignoring");
        result = GST_FLOW_OK;
        goto done;
    }

    /* ERRORS */
empty_header:
    {
        GST_ELEMENT_ERROR (vd, STREAM, DECODE, (NULL), ("empty header received"));
        result = GST_FLOW_ERROR;
        goto done;
    }
}

static GstFlowReturn
fs_funnel_chain (GstPad * pad, GstBuffer * buffer)
{
  GstFlowReturn res;
  FsFunnel *funnel = FS_FUNNEL (gst_pad_get_parent (pad));
  FsFunnelPadPrivate *priv = gst_pad_get_element_private (pad);
  GstEvent *event = NULL;
  GstClockTime newts;
  GstCaps *padcaps;

  GST_DEBUG_OBJECT (funnel, "received buffer %p", buffer);

  GST_OBJECT_LOCK (funnel);
  if (priv->segment.format == GST_FORMAT_UNDEFINED) {
    GST_WARNING_OBJECT (funnel, "Got buffer without segment,"
        " setting segment [0,inf[");
     gst_segment_set_newsegment_full (&priv->segment, FALSE, 1.0, 1.0,
         GST_FORMAT_TIME, 0, -1, 0);
  }

  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buffer)))
    gst_segment_set_last_stop (&priv->segment, priv->segment.format,
        GST_BUFFER_TIMESTAMP (buffer));

  newts = gst_segment_to_running_time (&priv->segment,
      priv->segment.format, GST_BUFFER_TIMESTAMP (buffer));
  if (newts != GST_BUFFER_TIMESTAMP (buffer)) {
    buffer = gst_buffer_make_metadata_writable (buffer);
    GST_BUFFER_TIMESTAMP (buffer) = newts;
  }

  if (!funnel->has_segment)
  {
    event = gst_event_new_new_segment_full (FALSE, 1.0, 1.0, GST_FORMAT_TIME,
        0, -1, 0);
    funnel->has_segment = TRUE;
  }
  GST_OBJECT_UNLOCK (funnel);

  if (event) {
    if (!gst_pad_push_event (funnel->srcpad, event))
      GST_WARNING_OBJECT (funnel, "Could not push out newsegment event");
  }


  GST_OBJECT_LOCK (pad);
  padcaps = GST_PAD_CAPS (funnel->srcpad);
  GST_OBJECT_UNLOCK (pad);

  if (GST_BUFFER_CAPS (buffer) && GST_BUFFER_CAPS (buffer) != padcaps) {
    if (!gst_pad_set_caps (funnel->srcpad, GST_BUFFER_CAPS (buffer))) {
      res = GST_FLOW_NOT_NEGOTIATED;
      goto out;
    }
  }

  res = gst_pad_push (funnel->srcpad, buffer);

  GST_LOG_OBJECT (funnel, "handled buffer %s", gst_flow_get_name (res));

 out:
  gst_object_unref (funnel);

  return res;
}

static GstFlowReturn gst_openh264dec_handle_frame(GstVideoDecoder *decoder, GstVideoCodecFrame *frame)
{
    GstOpenh264Dec *openh264dec = GST_OPENH264DEC(decoder);
    GstMapInfo map_info;
    GstVideoCodecState *state;
    SBufferInfo dst_buf_info;
    DECODING_STATE ret;
    guint8 *yuvdata[3];
    GstFlowReturn flow_status;
    GstVideoFrame video_frame;
    guint actual_width, actual_height;
    guint i;
    guint8 *p;
    guint row_stride, component_width, component_height, src_width, row;

    if (frame) {
        if (!gst_buffer_map(frame->input_buffer, &map_info, GST_MAP_READ)) {
            GST_ERROR_OBJECT(openh264dec, "Cannot map input buffer!");
            return GST_FLOW_ERROR;
        }

        GST_LOG_OBJECT(openh264dec, "handle frame, %d", map_info.size > 4 ? map_info.data[4] & 0x1f : -1);

        memset (&dst_buf_info, 0, sizeof (SBufferInfo));
        ret = openh264dec->priv->decoder->DecodeFrame2(map_info.data, map_info.size, yuvdata, &dst_buf_info);

        if (ret == dsNoParamSets) {
            GST_DEBUG_OBJECT(openh264dec, "Requesting a key unit");
            gst_pad_push_event(GST_VIDEO_DECODER_SINK_PAD(decoder),
                gst_video_event_new_upstream_force_key_unit(GST_CLOCK_TIME_NONE, FALSE, 0));
        }

        if (ret != dsErrorFree && ret != dsNoParamSets) {
            GST_DEBUG_OBJECT(openh264dec, "Requesting a key unit");
            gst_pad_push_event(GST_VIDEO_DECODER_SINK_PAD(decoder),
                               gst_video_event_new_upstream_force_key_unit(GST_CLOCK_TIME_NONE, FALSE, 0));
            GST_LOG_OBJECT(openh264dec, "error decoding nal, return code: %d", ret);
        }

        gst_buffer_unmap(frame->input_buffer, &map_info);
        gst_video_codec_frame_unref (frame);
        frame = NULL;
    } else {
        memset (&dst_buf_info, 0, sizeof (SBufferInfo));
        ret = openh264dec->priv->decoder->DecodeFrame2(NULL, 0, yuvdata, &dst_buf_info);
        if (ret != dsErrorFree)
            return GST_FLOW_EOS;
    }

    /* FIXME: openh264 has no way for us to get a connection
     * between the input and output frames, we just have to
     * guess based on the input. Fortunately openh264 can
     * only do baseline profile. */
    frame = gst_video_decoder_get_oldest_frame (decoder);
    if (!frame) {
      /* Can only happen in finish() */
      return GST_FLOW_EOS;
    }

    /* No output available yet */
    if (dst_buf_info.iBufferStatus != 1) {
        return (frame ? GST_FLOW_OK : GST_FLOW_EOS);
    }

    actual_width  = dst_buf_info.UsrData.sSystemBuffer.iWidth;
    actual_height = dst_buf_info.UsrData.sSystemBuffer.iHeight;

    if (!gst_pad_has_current_caps (GST_VIDEO_DECODER_SRC_PAD (openh264dec)) || actual_width != openh264dec->priv->width || actual_height != openh264dec->priv->height) {
        state = gst_video_decoder_set_output_state(decoder,
            GST_VIDEO_FORMAT_I420,
            actual_width,
            actual_height,
            openh264dec->priv->input_state);
        openh264dec->priv->width = actual_width;
        openh264dec->priv->height = actual_height;

        if (!gst_video_decoder_negotiate(decoder)) {
            GST_ERROR_OBJECT(openh264dec, "Failed to negotiate with downstream elements");
            return GST_FLOW_NOT_NEGOTIATED;
        }
    } else {
        state = gst_video_decoder_get_output_state(decoder);
    }

    flow_status = gst_video_decoder_allocate_output_frame(decoder, frame);
    if (flow_status != GST_FLOW_OK) {
        gst_video_codec_state_unref (state);
        return flow_status;
    }

    if (!gst_video_frame_map(&video_frame, &state->info, frame->output_buffer, GST_MAP_WRITE)) {
        GST_ERROR_OBJECT(openh264dec, "Cannot map output buffer!");
        gst_video_codec_state_unref (state);
        return GST_FLOW_ERROR;
    }

    for (i = 0; i < 3; i++) {
        p = GST_VIDEO_FRAME_COMP_DATA(&video_frame, i);
        row_stride = GST_VIDEO_FRAME_COMP_STRIDE(&video_frame, i);
        component_width = GST_VIDEO_FRAME_COMP_WIDTH(&video_frame, i);
//.........这里部分代码省略.........

static gboolean
rsn_audiomunge_sink_event (GstPad * pad, GstEvent * event)
{
  gboolean ret = FALSE;
  RsnAudioMunge *munge = RSN_AUDIOMUNGE (gst_pad_get_parent (pad));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_FLUSH_STOP:
      rsn_audiomunge_reset (munge);
      ret = gst_pad_push_event (munge->srcpad, event);
      break;
    case GST_EVENT_NEWSEGMENT:
    {
      GstSegment *segment;
      gboolean update;
      GstFormat format;
      gdouble rate, arate;
      gint64 start, stop, time;

      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
          &start, &stop, &time);

      /* we need TIME format */
      if (format != GST_FORMAT_TIME)
        goto newseg_wrong_format;

      /* now configure the values */
      segment = &munge->sink_segment;

      gst_segment_set_newsegment_full (segment, update,
          rate, arate, format, start, stop, time);

      /*
       * FIXME:
       * If this is a segment update and accum >= threshold,
       * or we're in a still frame and there's been no audio received,
       * then we need to generate some audio data.
       *
       * If caused by a segment start update (time advancing in a gap) adjust
       * the new-segment and send the buffer.
       *
       * Otherwise, send the buffer before the newsegment, so that it appears
       * in the closing segment.
       */
      if (!update) {
        GST_DEBUG_OBJECT (munge,
            "Sending newsegment: update %d start %" GST_TIME_FORMAT " stop %"
            GST_TIME_FORMAT " accum now %" GST_TIME_FORMAT, update,
            GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
            GST_TIME_ARGS (segment->accum));

        ret = gst_pad_push_event (munge->srcpad, event);
      }

      if (!munge->have_audio) {
        if ((update && segment->accum >= AUDIO_FILL_THRESHOLD)
            || munge->in_still) {
          GST_DEBUG_OBJECT (munge,
              "Sending audio fill with ts %" GST_TIME_FORMAT ": accum = %"
              GST_TIME_FORMAT " still-state=%d", GST_TIME_ARGS (segment->start),
              GST_TIME_ARGS (segment->accum), munge->in_still);

          /* Just generate a 200ms silence buffer for now. FIXME: Fill the gap */
          if (rsn_audiomunge_make_audio (munge, segment->start,
                  GST_SECOND / 5) == GST_FLOW_OK)
            munge->have_audio = TRUE;
        } else {
          GST_LOG_OBJECT (munge, "Not sending audio fill buffer: "
              "Not segment update, or segment accum below thresh: accum = %"
              GST_TIME_FORMAT, GST_TIME_ARGS (segment->accum));
        }
      }

      if (update) {
        GST_DEBUG_OBJECT (munge,
            "Sending newsegment: update %d start %" GST_TIME_FORMAT " stop %"
            GST_TIME_FORMAT " accum now %" GST_TIME_FORMAT, update,
            GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
            GST_TIME_ARGS (segment->accum));

        ret = gst_pad_push_event (munge->srcpad, event);
      }

      break;
    }
    case GST_EVENT_CUSTOM_DOWNSTREAM:
    {
      gboolean in_still;

      if (gst_video_event_parse_still_frame (event, &in_still)) {
        /* Remember the still-frame state, so we can generate a pre-roll
         * buffer when a new-segment arrives */
        munge->in_still = in_still;
        GST_INFO_OBJECT (munge, "AUDIO MUNGE: still-state now %d",
            munge->in_still);
      }

      ret = gst_pad_push_event (munge->srcpad, event);
      break;
    }
//.........这里部分代码省略.........

static GstFlowReturn
gst_wavpack_parse_push_buffer (GstWavpackParse * wvparse, GstBuffer * buf,
    WavpackHeader * header)
{
  GstFlowReturn ret;

  wvparse->current_offset += header->ckSize + 8;
  wvparse->segment.last_stop = header->block_index;

  if (wvparse->need_newsegment) {
    if (gst_wavpack_parse_send_newsegment (wvparse, FALSE))
      wvparse->need_newsegment = FALSE;
  }

  /* send any queued events */
  if (wvparse->queued_events) {
    GList *l;

    for (l = wvparse->queued_events; l != NULL; l = l->next) {
      gst_pad_push_event (wvparse->srcpad, GST_EVENT (l->data));
    }
    g_list_free (wvparse->queued_events);
    wvparse->queued_events = NULL;
  }

  if (wvparse->pending_buffer == NULL) {
    wvparse->pending_buffer = buf;
    wvparse->pending_offset = header->block_index;
  } else if (wvparse->pending_offset == header->block_index) {
    wvparse->pending_buffer = gst_buffer_join (wvparse->pending_buffer, buf);
  } else {
    GST_ERROR ("Got incomplete block, dropping");
    gst_buffer_unref (wvparse->pending_buffer);
    wvparse->pending_buffer = buf;
    wvparse->pending_offset = header->block_index;
  }

  if (!(header->flags & FINAL_BLOCK))
    return GST_FLOW_OK;

  buf = wvparse->pending_buffer;
  wvparse->pending_buffer = NULL;

  GST_BUFFER_TIMESTAMP (buf) = gst_util_uint64_scale_int (header->block_index,
      GST_SECOND, wvparse->samplerate);
  GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (header->block_samples,
      GST_SECOND, wvparse->samplerate);
  GST_BUFFER_OFFSET (buf) = header->block_index;
  GST_BUFFER_OFFSET_END (buf) = header->block_index + header->block_samples;

  if (wvparse->discont || wvparse->next_block_index != header->block_index) {
    GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
    wvparse->discont = FALSE;
  }

  wvparse->next_block_index = header->block_index + header->block_samples;

  gst_buffer_set_caps (buf, GST_PAD_CAPS (wvparse->srcpad));

  GST_LOG_OBJECT (wvparse, "Pushing buffer with time %" GST_TIME_FORMAT,
      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));

  ret = gst_pad_push (wvparse->srcpad, buf);

  wvparse->segment.last_stop = wvparse->next_block_index;

  return ret;
}

static gboolean
gst_wavpack_parse_create_src_pad (GstWavpackParse * wvparse, GstBuffer * buf,
    WavpackHeader * header)
{
  GstWavpackMetadata meta;
  GstCaps *caps = NULL;
  guchar *bufptr;

  g_assert (wvparse->srcpad == NULL);

  bufptr = GST_BUFFER_DATA (buf) + sizeof (WavpackHeader);

  while (gst_wavpack_read_metadata (&meta, GST_BUFFER_DATA (buf), &bufptr)) {
    switch (meta.id) {
      case ID_WVC_BITSTREAM:{
        caps = gst_caps_new_simple ("audio/x-wavpack-correction",
            "framed", G_TYPE_BOOLEAN, TRUE, NULL);
        wvparse->srcpad =
            gst_pad_new_from_template (gst_element_class_get_pad_template
            (GST_ELEMENT_GET_CLASS (wvparse), "wvcsrc"), "wvcsrc");
        break;
      }
      case ID_WV_BITSTREAM:
      case ID_WVX_BITSTREAM:{
        WavpackStreamReader *stream_reader = gst_wavpack_stream_reader_new ();

        WavpackContext *wpc;

        gchar error_msg[80];

        read_id rid;

        gint channel_mask;

        rid.buffer = GST_BUFFER_DATA (buf);
        rid.length = GST_BUFFER_SIZE (buf);
        rid.position = 0;

        wpc =
            WavpackOpenFileInputEx (stream_reader, &rid, NULL, error_msg, 0, 0);

        if (!wpc)
          return FALSE;

        wvparse->samplerate = WavpackGetSampleRate (wpc);
        wvparse->channels = WavpackGetNumChannels (wpc);
        wvparse->total_samples =
            (header->total_samples ==
            0xffffffff) ? G_GINT64_CONSTANT (-1) : header->total_samples;

        caps = gst_caps_new_simple ("audio/x-wavpack",
            "width", G_TYPE_INT, WavpackGetBitsPerSample (wpc),
            "channels", G_TYPE_INT, wvparse->channels,
            "rate", G_TYPE_INT, wvparse->samplerate,
            "framed", G_TYPE_BOOLEAN, TRUE, NULL);
#ifdef WAVPACK_OLD_API
        channel_mask = wpc->config.channel_mask;
#else
        channel_mask = WavpackGetChannelMask (wpc);
#endif
        if (channel_mask == 0)
          channel_mask =
              gst_wavpack_get_default_channel_mask (wvparse->channels);

        if (channel_mask != 0) {
          if (!gst_wavpack_set_channel_layout (caps, channel_mask)) {
            GST_WARNING_OBJECT (wvparse, "Failed to set channel layout");
            gst_caps_unref (caps);
            caps = NULL;
            WavpackCloseFile (wpc);
            g_free (stream_reader);
            break;
          }
        }

        wvparse->srcpad =
            gst_pad_new_from_template (gst_element_class_get_pad_template
            (GST_ELEMENT_GET_CLASS (wvparse), "src"), "src");
        WavpackCloseFile (wpc);
        g_free (stream_reader);
        break;
      }
      default:{
        GST_LOG_OBJECT (wvparse, "unhandled ID: 0x%02x", meta.id);
        break;
      }
    }
    if (caps != NULL)
      break;
  }

  if (caps == NULL || wvparse->srcpad == NULL)
    return FALSE;

  GST_DEBUG_OBJECT (wvparse, "Added src pad with caps %" GST_PTR_FORMAT, caps);

  gst_pad_set_query_function (wvparse->srcpad,
      GST_DEBUG_FUNCPTR (gst_wavpack_parse_src_query));
  gst_pad_set_query_type_function (wvparse->srcpad,
      GST_DEBUG_FUNCPTR (gst_wavpack_parse_get_src_query_types));
//.........这里部分代码省略.........

GstFlowReturn
gst_vdp_video_yuv_transform (GstBaseTransform * trans, GstBuffer * inbuf,
    GstBuffer * outbuf)
{
  GstVdpVideoYUV *video_yuv = GST_VDP_VIDEO_YUV (trans);
  GstVdpDevice *device;
  VdpVideoSurface surface;

  device = GST_VDP_VIDEO_BUFFER (inbuf)->device;
  surface = GST_VDP_VIDEO_BUFFER (inbuf)->surface;

  switch (video_yuv->format) {
    case GST_MAKE_FOURCC ('Y', 'V', '1', '2'):
    {
      VdpStatus status;
      guint8 *data[3];
      guint32 stride[3];

      data[0] = GST_BUFFER_DATA (outbuf) +
          gst_video_format_get_component_offset (GST_VIDEO_FORMAT_YV12,
          0, video_yuv->width, video_yuv->height);
      data[1] = GST_BUFFER_DATA (outbuf) +
          gst_video_format_get_component_offset (GST_VIDEO_FORMAT_YV12,
          2, video_yuv->width, video_yuv->height);
      data[2] = GST_BUFFER_DATA (outbuf) +
          gst_video_format_get_component_offset (GST_VIDEO_FORMAT_YV12,
          1, v