gstreamer/ext/opus/gstopusparse.c

/* 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>
 *
 * 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., 59 Temple Place - Suite 330,
 * Boston, MA 02111-1307, USA.
 */

/**
 * SECTION:element-opusparse
 * @see_also: opusenc, opusdec
 *
 * This element parses OPUS packets.
 *
 * <refsect2>
 * <title>Example pipelines</title>
 * |[
 * gst-launch -v filesrc location=opusdata ! opusparse ! opusdec ! audioconvert ! audioresample ! alsasink
 * ]| Decode and plays an unmuxed Opus file.
 * </refsect2>
 */

#ifdef HAVE_CONFIG_H
#  include "config.h"
#endif

#include <opus/opus.h>
#include "gstopusheader.h"
#include "gstopusparse.h"

GST_DEBUG_CATEGORY_STATIC (opusparse_debug);
#define GST_CAT_DEFAULT opusparse_debug

#define MAX_PAYLOAD_BYTES 1500

static GstStaticPadTemplate opus_parse_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
    GST_PAD_SRC,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/x-opus, framed = (boolean) true")
    );

static GstStaticPadTemplate opus_parse_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/x-opus")
    );

G_DEFINE_TYPE (GstOpusParse, gst_opus_parse, GST_TYPE_BASE_PARSE);

static gboolean gst_opus_parse_start (GstBaseParse * parse);
static gboolean gst_opus_parse_stop (GstBaseParse * parse);
static gboolean gst_opus_parse_check_valid_frame (GstBaseParse * base,
    GstBaseParseFrame * frame, guint * frame_size, gint * skip);
static GstFlowReturn gst_opus_parse_parse_frame (GstBaseParse * base,
    GstBaseParseFrame * frame);

static void
gst_opus_parse_class_init (GstOpusParseClass * klass)
{
  GstBaseParseClass *bpclass;
  GstElementClass *element_class;

  bpclass = (GstBaseParseClass *) klass;
  element_class = (GstElementClass *) klass;

  bpclass->start = GST_DEBUG_FUNCPTR (gst_opus_parse_start);
  bpclass->stop = GST_DEBUG_FUNCPTR (gst_opus_parse_stop);
  bpclass->check_valid_frame =
      GST_DEBUG_FUNCPTR (gst_opus_parse_check_valid_frame);
  bpclass->parse_frame = GST_DEBUG_FUNCPTR (gst_opus_parse_parse_frame);

  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&opus_parse_src_factory));
  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&opus_parse_sink_factory));
  gst_element_class_set_details_simple (element_class, "Opus audio parser",
      "Codec/Parser/Audio",
      "parses opus audio streams",
      "Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>");

  GST_DEBUG_CATEGORY_INIT (opusparse_debug, "opusparse", 0,
      "opus parsing element");
}

static void
gst_opus_parse_init (GstOpusParse * parse)
{
  parse->header_sent = FALSE;
}

static gboolean
gst_opus_parse_start (GstBaseParse * base)
{
  GstOpusParse *parse = GST_OPUS_PARSE (base);

  parse->header_sent = FALSE;
  parse->next_ts = 0;

  return TRUE;
}

static gboolean
gst_opus_parse_stop (GstBaseParse * base)
{
  GstOpusParse *parse = GST_OPUS_PARSE (base);

  g_slist_foreach (parse->headers, (GFunc) gst_buffer_unref, NULL);
  parse->headers = NULL;

  parse->header_sent = FALSE;

  return TRUE;
}

static gboolean
gst_opus_parse_check_valid_frame (GstBaseParse * base,
    GstBaseParseFrame * frame, guint * frame_size, gint * skip)
{
  GstOpusParse *parse;
  guint8 *data;
  gsize size;
  guint32 packet_size;
  int ret = FALSE;
  int channels, bandwidth;

  parse = GST_OPUS_PARSE (base);

  data = GST_BUFFER_DATA (frame->buffer);
  size = GST_BUFFER_SIZE (frame->buffer);
  GST_DEBUG_OBJECT (parse, "Checking for frame, %u bytes in buffer", size);
  if (size < 4) {
    GST_DEBUG_OBJECT (parse, "Too small");
    goto beach;
  }
  packet_size = GST_READ_UINT32_BE (data);
  GST_DEBUG_OBJECT (parse, "Packet size: %u bytes", packet_size);
  if (packet_size > MAX_PAYLOAD_BYTES) {
    GST_DEBUG_OBJECT (parse, "Too large");
    goto beach;
  }
  if (packet_size > size - 4) {
    GST_DEBUG_OBJECT (parse, "Truncated");
    goto beach;
  }

  channels = opus_packet_get_nb_channels (data + 8);
  bandwidth = opus_packet_get_bandwidth (data + 8);
  if (channels < 0 || bandwidth < 0) {
    GST_DEBUG_OBJECT (parse, "It looked like a packet, but it is not");
    goto beach;
  }

  GST_DEBUG_OBJECT (parse, "Got Opus packet, %d bytes");

  if (!parse->header_sent) {
    GstCaps *caps;

    g_slist_foreach (parse->headers, (GFunc) gst_buffer_unref, NULL);
    parse->headers = NULL;

    gst_opus_header_create_caps (&caps, &parse->headers, channels, 0, NULL);

    gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);

    parse->header_sent = TRUE;
  }

  *skip = 8;
  *frame_size = packet_size;
  ret = TRUE;

beach:
  return ret;
}

/* Adapted copy of the one in gstoggstream.c... */
static guint64
packet_duration_opus (const guint8 * data, size_t len)
{
  static const guint64 durations[32] = {
    10000, 20000, 40000, 60000, /* Silk NB */
    10000, 20000, 40000, 60000, /* Silk MB */
    10000, 20000, 40000, 60000, /* Silk WB */
    10000, 20000,               /* Hybrid SWB */
    10000, 20000,               /* Hybrid FB */
    2500, 5000, 10000, 20000,   /* CELT NB */
    2500, 5000, 10000, 20000,   /* CELT NB */
    2500, 5000, 10000, 20000,   /* CELT NB */
    2500, 5000, 10000, 20000,   /* CELT NB */
  };

  gint64 duration;
  gint64 frame_duration;
  gint nframes;
  guint8 toc;

  if (len < 1)
    return 0;

  toc = data[0];

  frame_duration = durations[toc >> 3] * 1000;
  switch (toc & 3) {
    case 0:
      nframes = 1;
      break;
    case 1:
      nframes = 2;
      break;
    case 2:
      nframes = 2;
      break;
    case 3:
      if (len < 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 > 120 * GST_MSECOND) {
    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 / 1000000.f, nframes, duration / 1000000.f);
  return duration;
}

static GstFlowReturn
gst_opus_parse_parse_frame (GstBaseParse * base, GstBaseParseFrame * frame)
{
  guint64 duration;
  GstOpusParse *parse;

  parse = GST_OPUS_PARSE (base);

  GST_BUFFER_TIMESTAMP (frame->buffer) = parse->next_ts;

  duration =
      packet_duration_opus (GST_BUFFER_DATA (frame->buffer),
      GST_BUFFER_SIZE (frame->buffer));
  parse->next_ts += duration;

  GST_BUFFER_DURATION (frame->buffer) = duration;
  GST_BUFFER_OFFSET_END (frame->buffer) =
      gst_util_uint64_scale (parse->next_ts, 48000, GST_SECOND);
  GST_BUFFER_OFFSET (frame->buffer) = parse->next_ts;

  return GST_FLOW_OK;
}