C++ (Cpp) gst_audio_format_fill_silence Examples

Example #1

File: gstaudioaggregator.c Project: iperry/gst-plugins-bad

static GstBuffer *
gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
    guint num_frames)
{
  GstBuffer *outbuf = gst_buffer_new_allocate (NULL, num_frames *
      GST_AUDIO_INFO_BPF (&aagg->info), NULL);
  GstMapInfo outmap;

  gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
  gst_audio_format_fill_silence (aagg->info.finfo, outmap.data, outmap.size);
  gst_buffer_unmap (outbuf, &outmap);

  return outbuf;
}

Example #2

File: gstaudiosink.c Project: ConfusedReality/pkg_multimedia_gst-plugins-base

static gboolean
gst_audio_sink_ring_buffer_acquire (GstAudioRingBuffer * buf,
                                    GstAudioRingBufferSpec * spec)
{
    GstAudioSink *sink;
    GstAudioSinkClass *csink;
    gboolean result = FALSE;

    sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
    csink = GST_AUDIO_SINK_GET_CLASS (sink);

    if (csink->prepare)
        result = csink->prepare (sink, spec);
    if (!result)
        goto could_not_prepare;

    /* set latency to one more segment as we need some headroom */
    spec->seglatency = spec->segtotal + 1;

    buf->size = spec->segtotal * spec->segsize;

    buf->memory = g_malloc (buf->size);

    if (buf->spec.type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW) {
        gst_audio_format_fill_silence (buf->spec.info.finfo, buf->memory,
                                       buf->size);
    } else {
        /* FIXME, non-raw formats get 0 as the empty sample */
        memset (buf->memory, 0, buf->size);
    }


    return TRUE;

    /* ERRORS */
could_not_prepare:
    {
        GST_DEBUG_OBJECT (sink, "could not prepare device");
        return FALSE;
    }
}

Example #3

File: gstaudioaggregator.c Project: auni53/gst-plugins-bad

static GstBuffer *
gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
    guint num_frames)
{
  GstAllocator *allocator;
  GstAllocationParams params;
  GstBuffer *outbuf;
  GstMapInfo outmap;

  gst_aggregator_get_allocator (GST_AGGREGATOR (aagg), &allocator, &params);

  outbuf = gst_buffer_new_allocate (allocator, num_frames *
      GST_AUDIO_INFO_BPF (&aagg->info), &params);

  if (allocator)
    gst_object_unref (allocator);

  gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
  gst_audio_format_fill_silence (aagg->info.finfo, outmap.data, outmap.size);
  gst_buffer_unmap (outbuf, &outmap);

  return outbuf;
}

Example #4

File: gstinteraudiosrc.c Project: Distrotech/gst-plugins-bad

static GstFlowReturn
gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
    GstBuffer ** buf)
{
  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
  GstCaps *caps;
  GstBuffer *buffer;
  guint n, bpf;
  guint64 period_time;
  guint64 period_samples;

  GST_DEBUG_OBJECT (interaudiosrc, "create");

  buffer = NULL;
  caps = NULL;

  g_mutex_lock (&interaudiosrc->surface->mutex);
  if (interaudiosrc->surface->audio_info.finfo) {
    if (!gst_audio_info_is_equal (&interaudiosrc->surface->audio_info,
            &interaudiosrc->info)) {
      caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info);
      interaudiosrc->timestamp_offset +=
          gst_util_uint64_scale (interaudiosrc->n_samples, GST_SECOND,
          interaudiosrc->info.rate);
      interaudiosrc->n_samples = 0;
    }
  }

  bpf = interaudiosrc->surface->audio_info.bpf;
  period_time = interaudiosrc->surface->audio_period_time;
  period_samples =
      gst_util_uint64_scale (period_time, interaudiosrc->info.rate, GST_SECOND);

  if (bpf > 0)
    n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / bpf;
  else
    n = 0;

  if (n > period_samples)
    n = period_samples;
  if (n > 0) {
    buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter,
        n * bpf);
  } else {
    buffer = gst_buffer_new ();
    GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_GAP);
  }
  g_mutex_unlock (&interaudiosrc->surface->mutex);

  if (caps) {
    gboolean ret = gst_base_src_set_caps (src, caps);
    gst_caps_unref (caps);
    if (!ret) {
      GST_ERROR_OBJECT (src, "Failed to set caps %" GST_PTR_FORMAT, caps);
      if (buffer)
        gst_buffer_unref (buffer);
      return GST_FLOW_NOT_NEGOTIATED;
    }
  }

  buffer = gst_buffer_make_writable (buffer);

  bpf = interaudiosrc->info.bpf;
  if (n < period_samples) {
    GstMapInfo map;
    GstMemory *mem;

    GST_DEBUG_OBJECT (interaudiosrc,
        "creating %" G_GUINT64_FORMAT " samples of silence",
        period_samples - n);
    mem = gst_allocator_alloc (NULL, (period_samples - n) * bpf, NULL);
    if (gst_memory_map (mem, &map, GST_MAP_WRITE)) {
      gst_audio_format_fill_silence (interaudiosrc->info.finfo, map.data,
          map.size);
      gst_memory_unmap (mem, &map);
    }
    gst_buffer_prepend_memory (buffer, mem);
  }
  n = period_samples;

  GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
  GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n;
  GST_BUFFER_TIMESTAMP (buffer) = interaudiosrc->timestamp_offset +
      gst_util_uint64_scale (interaudiosrc->n_samples, GST_SECOND,
      interaudiosrc->info.rate);
  GST_DEBUG_OBJECT (interaudiosrc, "create ts %" GST_TIME_FORMAT,
      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
  GST_BUFFER_DURATION (buffer) = interaudiosrc->timestamp_offset +
      gst_util_uint64_scale (interaudiosrc->n_samples + n, GST_SECOND,
      interaudiosrc->info.rate) - GST_BUFFER_TIMESTAMP (buffer);
  GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
  if (interaudiosrc->n_samples == 0) {
    GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
  }
  interaudiosrc->n_samples += n;

  *buf = buffer;

  return GST_FLOW_OK;
}