gstreamer/gst-libs/gst/audio/audio.c

/* GStreamer
 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
 *
 * 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:gstaudio
 * @short_description: Support library for audio elements
 *
 * This library contains some helper functions for audio elements.
 */

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

#include "audio.h"
#include "audio-enumtypes.h"

#include <gst/gststructure.h>

/**
 * gst_audio_frame_byte_size:
 * @pad: the #GstPad to get the caps from
 *
 * Calculate byte size of an audio frame.
 *
 * Returns: the byte size, or 0 if there was an error
 */
int
gst_audio_frame_byte_size (GstPad * pad)
{
  /* FIXME: this should be moved closer to the gstreamer core
   * and be implemented for every mime type IMO
   */

  int width = 0;
  int channels = 0;
  GstCaps *caps;
  GstStructure *structure;

  /* get caps of pad */
  caps = gst_pad_get_current_caps (pad);

  if (caps == NULL)
    goto no_caps;

  structure = gst_caps_get_structure (caps, 0);

  gst_structure_get_int (structure, "width", &width);
  gst_structure_get_int (structure, "channels", &channels);
  gst_caps_unref (caps);

  return (width / 8) * channels;

  /* ERRORS */
no_caps:
  {
    /* ERROR: could not get caps of pad */
    g_warning ("gstaudio: could not get caps of pad %s:%s\n",
        GST_DEBUG_PAD_NAME (pad));
    return 0;
  }
}

/**
 * gst_audio_frame_length:
 * @pad: the #GstPad to get the caps from
 * @buf: the #GstBuffer
 *
 * Calculate length of buffer in frames.
 *
 * Returns: 0 if there's an error, or the number of frames if everything's ok
 */
long
gst_audio_frame_length (GstPad * pad, GstBuffer * buf)
{
  /* FIXME: this should be moved closer to the gstreamer core
   * and be implemented for every mime type IMO
   */
  int frame_byte_size = 0;

  frame_byte_size = gst_audio_frame_byte_size (pad);
  if (frame_byte_size == 0)
    /* error */
    return 0;
  /* FIXME: this function assumes the buffer size to be a whole multiple
   *        of the frame byte size
   */
  return gst_buffer_get_size (buf) / frame_byte_size;
}

/**
 * gst_audio_duration_from_pad_buffer:
 * @pad: the #GstPad to get the caps from
 * @buf: the #GstBuffer
 *
 * Calculate length in nanoseconds of audio buffer @buf based on capabilities of
 * @pad.
 *
 * Returns: the length.
 */
GstClockTime
gst_audio_duration_from_pad_buffer (GstPad * pad, GstBuffer * buf)
{
  long bytes = 0;
  int width = 0;
  int channels = 0;
  int rate = 0;
  GstCaps *caps;
  GstStructure *structure;

  g_assert (GST_IS_BUFFER (buf));

  /* get caps of pad */
  caps = gst_pad_get_current_caps (pad);
  if (caps == NULL)
    goto no_caps;

  structure = gst_caps_get_structure (caps, 0);
  bytes = gst_buffer_get_size (buf);
  gst_structure_get_int (structure, "width", &width);
  gst_structure_get_int (structure, "channels", &channels);
  gst_structure_get_int (structure, "rate", &rate);
  gst_caps_unref (caps);

  g_assert (bytes != 0);
  g_assert (width != 0);
  g_assert (channels != 0);
  g_assert (rate != 0);

  return (bytes * 8 * GST_SECOND) / (rate * channels * width);

  /* ERRORS */
no_caps:
  {
    /* ERROR: could not get caps of pad */
    g_warning ("gstaudio: could not get caps of pad %s:%s\n",
        GST_DEBUG_PAD_NAME (pad));
    return GST_CLOCK_TIME_NONE;
  }
}

/**
 * gst_audio_is_buffer_framed:
 * @pad: the #GstPad to get the caps from
 * @buf: the #GstBuffer
 *
 * Check if the buffer size is a whole multiple of the frame size.
 *
 * Returns: %TRUE if buffer size is multiple.
 */
gboolean
gst_audio_is_buffer_framed (GstPad * pad, GstBuffer * buf)
{
  if (gst_buffer_get_size (buf) % gst_audio_frame_byte_size (pad) == 0)
    return TRUE;
  else
    return FALSE;
}

/**
 * gst_audio_buffer_clip:
 * @buffer: The buffer to clip.
 * @segment: Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which the buffer should be clipped.
 * @rate: sample rate.
 * @frame_size: size of one audio frame in bytes.
 *
 * Clip the the buffer to the given %GstSegment.
 *
 * After calling this function the caller does not own a reference to 
 * @buffer anymore.
 *
 * Returns: %NULL if the buffer is completely outside the configured segment,
 * otherwise the clipped buffer is returned.
 *
 * If the buffer has no timestamp, it is assumed to be inside the segment and
 * is not clipped 
 *
 * Since: 0.10.14
 */
GstBuffer *
gst_audio_buffer_clip (GstBuffer * buffer, GstSegment * segment, gint rate,
    gint frame_size)
{
  GstBuffer *ret;
  GstClockTime timestamp = GST_CLOCK_TIME_NONE, duration = GST_CLOCK_TIME_NONE;
  guint64 offset = GST_BUFFER_OFFSET_NONE, offset_end = GST_BUFFER_OFFSET_NONE;
  gsize trim, size;
  gboolean change_duration = TRUE, change_offset = TRUE, change_offset_end =
      TRUE;

  g_return_val_if_fail (segment->format == GST_FORMAT_TIME ||
      segment->format == GST_FORMAT_DEFAULT, buffer);
  g_return_val_if_fail (GST_IS_BUFFER (buffer), NULL);

  if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
    /* No timestamp - assume the buffer is completely in the segment */
    return buffer;

  /* Get copies of the buffer metadata to change later. 
   * Calculate the missing values for the calculations,
   * they won't be changed later though. */

  trim = 0;
  size = gst_buffer_get_size (buffer);

  timestamp = GST_BUFFER_TIMESTAMP (buffer);
  GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
  if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
    duration = GST_BUFFER_DURATION (buffer);
  } else {
    change_duration = FALSE;
    duration = gst_util_uint64_scale (size / frame_size, GST_SECOND, rate);
  }

  if (GST_BUFFER_OFFSET_IS_VALID (buffer)) {
    offset = GST_BUFFER_OFFSET (buffer);
  } else {
    change_offset = FALSE;
    offset = 0;
  }

  if (GST_BUFFER_OFFSET_END_IS_VALID (buffer)) {
    offset_end = GST_BUFFER_OFFSET_END (buffer);
  } else {
    change_offset_end = FALSE;
    offset_end = offset + size / frame_size;
  }

  if (segment->format == GST_FORMAT_TIME) {
    /* Handle clipping for GST_FORMAT_TIME */

    guint64 start, stop, cstart, cstop, diff;

    start = timestamp;
    stop = timestamp + duration;

    if (gst_segment_clip (segment, GST_FORMAT_TIME,
            start, stop, &cstart, &cstop)) {

      diff = cstart - start;
      if (diff > 0) {
        timestamp = cstart;

        if (change_duration)
          duration -= diff;

        diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
        if (change_offset)
          offset += diff;
        trim += diff * frame_size;
        size -= diff * frame_size;
      }

      diff = stop - cstop;
      if (diff > 0) {
        /* duration is always valid if stop is valid */
        duration -= diff;

        diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
        if (change_offset_end)
          offset_end -= diff;
        size -= diff * frame_size;
      }
    } else {
      gst_buffer_unref (buffer);
      return NULL;
    }
  } else {
    /* Handle clipping for GST_FORMAT_DEFAULT */
    guint64 start, stop, cstart, cstop, diff;

    g_return_val_if_fail (GST_BUFFER_OFFSET_IS_VALID (buffer), buffer);

    start = offset;
    stop = offset_end;

    if (gst_segment_clip (segment, GST_FORMAT_DEFAULT,
            start, stop, &cstart, &cstop)) {

      diff = cstart - start;
      if (diff > 0) {
        offset = cstart;

        timestamp = gst_util_uint64_scale (cstart, GST_SECOND, rate);

        if (change_duration)
          duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);

        trim += diff * frame_size;
        size -= diff * frame_size;
      }

      diff = stop - cstop;
      if (diff > 0) {
        offset_end = cstop;

        if (change_duration)
          duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);

        size -= diff * frame_size;
      }
    } else {
      gst_buffer_unref (buffer);
      return NULL;
    }
  }

  /* Get a writable buffer and apply all changes */
  GST_DEBUG ("trim %" G_GSIZE_FORMAT " size %" G_GSIZE_FORMAT, trim, size);
  ret = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, trim, size);
  gst_buffer_unref (buffer);

  GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
  GST_BUFFER_TIMESTAMP (ret) = timestamp;

  if (change_duration)
    GST_BUFFER_DURATION (ret) = duration;
  if (change_offset)
    GST_BUFFER_OFFSET (ret) = offset;
  if (change_offset_end)
    GST_BUFFER_OFFSET_END (ret) = offset_end;

  return ret;
}