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712 lines
20 KiB
C
712 lines
20 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:gstaudio
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* @short_description: Support library for audio elements
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*
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* This library contains some helper functions for audio elements.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include "audio.h"
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#include "audio-enumtypes.h"
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#include <gst/gststructure.h>
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#define SINT (GST_AUDIO_FORMAT_FLAG_INTEGER | GST_AUDIO_FORMAT_FLAG_SIGNED)
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#define UINT (GST_AUDIO_FORMAT_FLAG_INTEGER)
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#define MAKE_FORMAT(str,desc,flags,end,width,depth,silent) \
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{ GST_AUDIO_FORMAT_ ##str, G_STRINGIFY(str), desc, flags, end, width, depth, silent }
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#define SILENT_0 { 0, 0, 0, 0, 0, 0, 0, 0 }
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#define SILENT_U8 { 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80 }
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#define SILENT_U16LE { 0x00, 0x80, 0x00, 0x80, 0x00, 0x80, 0x00, 0x80 }
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#define SILENT_U16BE { 0x80, 0x00, 0x80, 0x00, 0x80, 0x00, 0x80, 0x00 }
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#define SILENT_U24_32LE { 0x00, 0x00, 0x80, 0x00, 0x00, 0x00, 0x80, 0x00 }
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#define SILENT_U24_32BE { 0x00, 0x80, 0x00, 0x00, 0x00, 0x80, 0x00, 0x00 }
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#define SILENT_U32LE { 0x00, 0x00, 0x00, 0x80, 0x00, 0x00, 0x00, 0x80 }
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#define SILENT_U32BE { 0x80, 0x00, 0x00, 0x00, 0x80, 0x00, 0x00, 0x00 }
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#define SILENT_U24LE { 0x00, 0x00, 0x80, 0x00, 0x00, 0x80 }
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#define SILENT_U24BE { 0x80, 0x00, 0x00, 0x80, 0x00, 0x00 }
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#define SILENT_U20LE { 0x00, 0x00, 0x08, 0x00, 0x00, 0x08 }
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#define SILENT_U20BE { 0x08, 0x00, 0x00, 0x08, 0x00, 0x00 }
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#define SILENT_U18LE { 0x00, 0x00, 0x02, 0x00, 0x00, 0x02 }
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#define SILENT_U18BE { 0x02, 0x00, 0x00, 0x02, 0x00, 0x00 }
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static GstAudioFormatInfo formats[] = {
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{GST_AUDIO_FORMAT_UNKNOWN, "UNKNOWN", 0, 0, 0, 0},
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/* 8 bit */
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MAKE_FORMAT (S8, "8-bit signed PCM audio", SINT, 0, 8, 8, SILENT_0),
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MAKE_FORMAT (U8, "8-bit unsigned PCM audio", UINT, 0, 8, 8, SILENT_U8),
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/* 16 bit */
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MAKE_FORMAT (S16LE, "16-bit signed PCM audio", SINT, G_LITTLE_ENDIAN, 16, 16,
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SILENT_0),
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MAKE_FORMAT (S16BE, "16-bit signed PCM audio", SINT, G_BIG_ENDIAN, 16, 16,
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SILENT_0),
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MAKE_FORMAT (U16LE, "16-bit unsigned PCM audio", UINT, G_LITTLE_ENDIAN, 16,
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16, SILENT_U16LE),
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MAKE_FORMAT (U16BE, "16-bit unsigned PCM audio", UINT, G_BIG_ENDIAN, 16, 16,
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SILENT_U16BE),
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/* 24 bit in low 3 bytes of 32 bits */
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MAKE_FORMAT (S24_32LE, "24-bit signed PCM audio", SINT, G_LITTLE_ENDIAN, 32,
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24, SILENT_0),
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MAKE_FORMAT (S24_32BE, "24-bit signed PCM audio", SINT, G_BIG_ENDIAN, 32, 24,
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SILENT_0),
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MAKE_FORMAT (U24_32LE, "24-bit unsigned PCM audio", UINT, G_LITTLE_ENDIAN, 32,
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24, SILENT_U24_32LE),
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MAKE_FORMAT (U24_32BE, "24-bit unsigned PCM audio", UINT, G_BIG_ENDIAN, 32,
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24, SILENT_U24_32BE),
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/* 32 bit */
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MAKE_FORMAT (S32LE, "32-bit signed PCM audio", SINT, G_LITTLE_ENDIAN, 32, 32,
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SILENT_0),
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MAKE_FORMAT (S32BE, "32-bit signed PCM audio", SINT, G_BIG_ENDIAN, 32, 32,
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SILENT_0),
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MAKE_FORMAT (U32LE, "32-bit unsigned PCM audio", UINT, G_LITTLE_ENDIAN, 32,
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32, SILENT_U32LE),
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MAKE_FORMAT (U32BE, "32-bit unsigned PCM audio", UINT, G_BIG_ENDIAN, 32, 32,
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SILENT_U32BE),
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/* 24 bit in 3 bytes */
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MAKE_FORMAT (S24LE, "24-bit signed PCM audio", SINT, G_LITTLE_ENDIAN, 24, 24,
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SILENT_0),
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MAKE_FORMAT (S24BE, "24-bit signed PCM audio", SINT, G_BIG_ENDIAN, 24, 24,
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SILENT_0),
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MAKE_FORMAT (U24LE, "24-bit unsigned PCM audio", UINT, G_LITTLE_ENDIAN, 24,
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24, SILENT_U24LE),
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MAKE_FORMAT (U24BE, "24-bit unsigned PCM audio", UINT, G_BIG_ENDIAN, 24, 24,
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SILENT_U24BE),
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/* 20 bit in 3 bytes */
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MAKE_FORMAT (S20LE, "20-bit signed PCM audio", SINT, G_LITTLE_ENDIAN, 24, 20,
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SILENT_0),
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MAKE_FORMAT (S20BE, "20-bit signed PCM audio", SINT, G_BIG_ENDIAN, 24, 20,
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SILENT_0),
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MAKE_FORMAT (U20LE, "20-bit unsigned PCM audio", UINT, G_LITTLE_ENDIAN, 24,
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20, SILENT_U20LE),
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MAKE_FORMAT (U20BE, "20-bit unsigned PCM audio", UINT, G_BIG_ENDIAN, 24, 20,
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SILENT_U20BE),
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/* 18 bit in 3 bytes */
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MAKE_FORMAT (S18LE, "18-bit signed PCM audio", SINT, G_LITTLE_ENDIAN, 24, 18,
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SILENT_0),
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MAKE_FORMAT (S18BE, "18-bit signed PCM audio", SINT, G_BIG_ENDIAN, 24, 18,
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SILENT_0),
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MAKE_FORMAT (U18LE, "18-bit unsigned PCM audio", UINT, G_LITTLE_ENDIAN, 24,
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18, SILENT_U18LE),
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MAKE_FORMAT (U18BE, "18-bit unsigned PCM audio", UINT, G_BIG_ENDIAN, 24, 18,
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SILENT_U18BE),
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/* float */
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MAKE_FORMAT (F32LE, "32-bit floating-point audio",
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GST_AUDIO_FORMAT_FLAG_FLOAT, G_LITTLE_ENDIAN, 32, 32,
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SILENT_0),
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MAKE_FORMAT (F32BE, "32-bit floating-point audio",
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GST_AUDIO_FORMAT_FLAG_FLOAT, G_BIG_ENDIAN, 32, 32,
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SILENT_0),
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MAKE_FORMAT (F64LE, "64-bit floating-point audio",
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GST_AUDIO_FORMAT_FLAG_FLOAT, G_LITTLE_ENDIAN, 64, 64,
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SILENT_0),
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MAKE_FORMAT (F64BE, "64-bit floating-point audio",
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GST_AUDIO_FORMAT_FLAG_FLOAT, G_BIG_ENDIAN, 64, 64,
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SILENT_0)
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};
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/**
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* gst_audio_format_build_integer:
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* @sign: signed or unsigned format
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* @endianness: G_LITTLE_ENDIAN or G_BIG_ENDIAN
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* @width: amount of bits used per sample
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* @depth: amount of used bits in @width
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*
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* Construct a #GstAudioFormat with given parameters.
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*
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* Returns: a #GstAudioFormat or GST_AUDIO_FORMAT_UNKNOWN when no audio format
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* exists with the given parameters.
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*/
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GstAudioFormat
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gst_audio_format_build_integer (gboolean sign, gint endianness,
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gint width, gint depth)
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{
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gint i, e;
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for (i = 0; i < G_N_ELEMENTS (formats); i++) {
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GstAudioFormatInfo *finfo = &formats[i];
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/* must be int */
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if (!GST_AUDIO_FORMAT_INFO_IS_INTEGER (finfo))
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continue;
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/* width and depth must match */
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if (width != GST_AUDIO_FORMAT_INFO_WIDTH (finfo))
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continue;
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if (depth != GST_AUDIO_FORMAT_INFO_DEPTH (finfo))
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continue;
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/* if there is endianness, it must match */
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e = GST_AUDIO_FORMAT_INFO_ENDIANNESS (finfo);
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if (e && e != endianness)
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continue;
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/* check sign */
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if (sign && !GST_AUDIO_FORMAT_INFO_IS_SIGNED (finfo))
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continue;
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return GST_AUDIO_FORMAT_INFO_FORMAT (finfo);
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}
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return GST_AUDIO_FORMAT_UNKNOWN;
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}
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/**
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* gst_audio_format_from_string:
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* @format: a format string
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*
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* Convert the @format string to its #GstAudioFormat.
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*
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* Returns: the #GstAudioFormat for @format or GST_AUDIO_FORMAT_UNKNOWN when the
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* string is not a known format.
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*/
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GstAudioFormat
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gst_audio_format_from_string (const gchar * format)
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{
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guint i;
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for (i = 0; i < G_N_ELEMENTS (formats); i++) {
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if (strcmp (GST_AUDIO_FORMAT_INFO_NAME (&formats[i]), format) == 0)
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return GST_AUDIO_FORMAT_INFO_FORMAT (&formats[i]);
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}
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return GST_AUDIO_FORMAT_UNKNOWN;
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}
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const gchar *
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gst_audio_format_to_string (GstAudioFormat format)
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{
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g_return_val_if_fail (format != GST_AUDIO_FORMAT_UNKNOWN, NULL);
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if (format >= G_N_ELEMENTS (formats))
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return NULL;
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return GST_AUDIO_FORMAT_INFO_NAME (&formats[format]);
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}
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/**
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* gst_audio_format_get_info:
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* @format: a #GstAudioFormat
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*
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* Get the #GstAudioFormatInfo for @format
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*
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* Returns: The #GstAudioFormatInfo for @format.
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*/
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const GstAudioFormatInfo *
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gst_audio_format_get_info (GstAudioFormat format)
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{
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g_return_val_if_fail (format != GST_AUDIO_FORMAT_UNKNOWN, NULL);
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g_return_val_if_fail (format < G_N_ELEMENTS (formats), NULL);
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return &formats[format];
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}
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/**
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* gst_audio_format_fill_silence:
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* @info: a #GstAudioFormatInfo
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* @dest: a destination to fill
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* @length: the length to fill
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*
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* Fill @length bytes in @dest with silence samples for @info.
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*/
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void
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gst_audio_format_fill_silence (const GstAudioFormatInfo * info,
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gpointer dest, gsize length)
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{
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guint8 *dptr = dest;
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g_return_if_fail (info != NULL);
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g_return_if_fail (dest != NULL);
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if (info->flags & GST_AUDIO_FORMAT_FLAG_FLOAT ||
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info->flags & GST_AUDIO_FORMAT_FLAG_SIGNED) {
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/* float or signed always 0 */
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memset (dest, 0, length);
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} else {
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gint i, j, bps = info->width >> 3;
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switch (bps) {
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case 1:
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memset (dest, info->silence[0], length);
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break;
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default:
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for (i = 0; i < length; i += bps) {
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for (j = 0; j < bps; j++)
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*dptr++ = info->silence[j];
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}
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break;
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}
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}
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}
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/**
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* gst_audio_info_init:
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* @info: a #GstAudioInfo
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*
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* Initialize @info with default values.
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*/
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void
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gst_audio_info_init (GstAudioInfo * info)
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{
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g_return_if_fail (info != NULL);
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memset (info, 0, sizeof (GstAudioInfo));
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}
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/**
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* gst_audio_info_set_format:
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* @info: a #GstAudioInfo
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* @format: the format
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* @rate: the samplerate
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* @channels: the number of channels
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*
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* Set the default info for the audio info of @format and @rate and @channels.
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*/
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void
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gst_audio_info_set_format (GstAudioInfo * info, GstAudioFormat format,
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gint rate, gint channels)
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{
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const GstAudioFormatInfo *finfo;
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g_return_if_fail (info != NULL);
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g_return_if_fail (format != GST_AUDIO_FORMAT_UNKNOWN);
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finfo = &formats[format];
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info->flags = 0;
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info->finfo = finfo;
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info->rate = rate;
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info->channels = channels;
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info->bpf = (finfo->width * channels) / 8;
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}
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/**
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* gst_audio_info_from_caps:
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* @info: a #GstAudioInfo
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* @caps: a #GstCaps
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*
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* Parse @caps and update @info.
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*
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* Returns: TRUE if @caps could be parsed
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*/
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gboolean
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gst_audio_info_from_caps (GstAudioInfo * info, const GstCaps * caps)
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{
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GstStructure *str;
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const gchar *s;
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GstAudioFormat format;
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gint rate, channels;
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const GValue *pos_val_arr, *pos_val_entry;
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gint i;
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g_return_val_if_fail (info != NULL, FALSE);
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g_return_val_if_fail (caps != NULL, FALSE);
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g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
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GST_DEBUG ("parsing caps %" GST_PTR_FORMAT, caps);
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str = gst_caps_get_structure (caps, 0);
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if (!gst_structure_has_name (str, "audio/x-raw"))
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goto wrong_name;
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if (!(s = gst_structure_get_string (str, "format")))
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goto no_format;
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format = gst_audio_format_from_string (s);
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if (format == GST_AUDIO_FORMAT_UNKNOWN)
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goto unknown_format;
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if (!gst_structure_get_int (str, "rate", &rate))
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goto no_rate;
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if (!gst_structure_get_int (str, "channels", &channels))
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goto no_channels;
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gst_audio_info_set_format (info, format, rate, channels);
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pos_val_arr = gst_structure_get_value (str, "channel-positions");
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if (pos_val_arr) {
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guint max_pos = MIN (channels, 64);
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if (channels != gst_value_array_get_size (pos_val_arr))
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goto incoherent_channels;
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for (i = 0; i < max_pos; i++) {
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pos_val_entry = gst_value_array_get_value (pos_val_arr, i);
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info->position[i] = g_value_get_enum (pos_val_entry);
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}
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} else {
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info->flags |= GST_AUDIO_FLAG_DEFAULT_POSITIONS;
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/* FIXME, set default positions */
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}
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return TRUE;
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/* ERROR */
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wrong_name:
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{
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GST_ERROR ("wrong name, expected audio/x-raw");
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return FALSE;
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}
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no_format:
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{
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GST_ERROR ("no format given");
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return FALSE;
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}
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unknown_format:
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{
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GST_ERROR ("unknown format given");
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return FALSE;
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}
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no_rate:
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{
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GST_ERROR ("no rate property given");
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return FALSE;
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}
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no_channels:
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{
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GST_ERROR ("no channels property given");
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return FALSE;
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}
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incoherent_channels:
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{
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GST_ERROR ("There should be %d channels positions, but %d are present",
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channels, gst_value_array_get_size (pos_val_arr));
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return FALSE;
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}
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}
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/**
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* gst_audio_info_to_caps:
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* @info: a #GstAudioInfo
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*
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* Convert the values of @info into a #GstCaps.
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*
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* Returns: (transfer full): the new #GstCaps containing the
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* info of @info.
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*/
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GstCaps *
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gst_audio_info_to_caps (GstAudioInfo * info)
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{
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GstCaps *caps;
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const gchar *format;
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g_return_val_if_fail (info != NULL, NULL);
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g_return_val_if_fail (info->finfo != NULL, NULL);
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g_return_val_if_fail (info->finfo->format != GST_AUDIO_FORMAT_UNKNOWN, NULL);
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format = gst_audio_format_to_string (info->finfo->format);
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g_return_val_if_fail (format != NULL, NULL);
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caps = gst_caps_new_simple ("audio/x-raw",
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"format", G_TYPE_STRING, format,
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"rate", G_TYPE_INT, info->rate,
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"channels", G_TYPE_INT, info->channels, NULL);
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if (info->channels > 2) {
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GValue pos_val_arr = { 0 }
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, pos_val_entry = {
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0};
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gint i, max_pos;
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GstStructure *str;
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/* build gvaluearray from positions */
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g_value_init (&pos_val_arr, GST_TYPE_ARRAY);
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g_value_init (&pos_val_entry, GST_TYPE_AUDIO_CHANNEL_POSITION);
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max_pos = MAX (info->channels, 64);
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for (i = 0; i < max_pos; i++) {
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g_value_set_enum (&pos_val_entry, info->position[i]);
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gst_value_array_append_value (&pos_val_arr, &pos_val_entry);
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}
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g_value_unset (&pos_val_entry);
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/* add to structure */
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str = gst_caps_get_structure (caps, 0);
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gst_structure_set_value (str, "channel-positions", &pos_val_arr);
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g_value_unset (&pos_val_arr);
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}
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return caps;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_format_convert:
|
|
* @info: a #GstAudioInfo
|
|
* @src_format: #GstFormat of the @src_value
|
|
* @src_value: value to convert
|
|
* @dest_format: #GstFormat of the @dest_value
|
|
* @dest_value: pointer to destination value
|
|
*
|
|
* Converts among various #GstFormat types. This function handles
|
|
* GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For
|
|
* raw audio, GST_FORMAT_DEFAULT corresponds to audio frames. This
|
|
* function can be used to handle pad queries of the type GST_QUERY_CONVERT.
|
|
*
|
|
* Returns: TRUE if the conversion was successful.
|
|
*/
|
|
gboolean
|
|
gst_audio_info_convert (GstAudioInfo * info,
|
|
GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)
|
|
{
|
|
gboolean res = TRUE;
|
|
gint bpf, rate;
|
|
|
|
GST_DEBUG ("converting value %" G_GINT64_FORMAT " from %s (%d) to %s (%d)",
|
|
src_val, gst_format_get_name (src_fmt), src_fmt,
|
|
gst_format_get_name (dest_fmt), dest_fmt);
|
|
|
|
if (src_fmt == dest_fmt || src_val == -1) {
|
|
*dest_val = src_val;
|
|
goto done;
|
|
}
|
|
|
|
/* get important info */
|
|
bpf = GST_AUDIO_INFO_BPF (info);
|
|
rate = GST_AUDIO_INFO_RATE (info);
|
|
|
|
if (bpf == 0 || rate == 0) {
|
|
GST_DEBUG ("no rate or bpf configured");
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
|
|
switch (src_fmt) {
|
|
case GST_FORMAT_BYTES:
|
|
switch (dest_fmt) {
|
|
case GST_FORMAT_TIME:
|
|
*dest_val = GST_FRAMES_TO_CLOCK_TIME (src_val / bpf, rate);
|
|
break;
|
|
case GST_FORMAT_DEFAULT:
|
|
*dest_val = src_val / bpf;
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
break;
|
|
case GST_FORMAT_DEFAULT:
|
|
switch (dest_fmt) {
|
|
case GST_FORMAT_TIME:
|
|
*dest_val = GST_FRAMES_TO_CLOCK_TIME (src_val, rate);
|
|
break;
|
|
case GST_FORMAT_BYTES:
|
|
*dest_val = src_val * bpf;
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
break;
|
|
case GST_FORMAT_TIME:
|
|
switch (dest_fmt) {
|
|
case GST_FORMAT_DEFAULT:
|
|
*dest_val = GST_CLOCK_TIME_TO_FRAMES (src_val, rate);
|
|
break;
|
|
case GST_FORMAT_BYTES:
|
|
*dest_val = GST_CLOCK_TIME_TO_FRAMES (src_val, rate);
|
|
*dest_val *= bpf;
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
done:
|
|
GST_DEBUG ("ret=%d result %" G_GINT64_FORMAT, res, *dest_val);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
* @bpf: size of one audio frame in bytes. This is the size of one sample
|
|
* * channels.
|
|
*
|
|
* 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 bpf)
|
|
{
|
|
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 / bpf, 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 / bpf;
|
|
}
|
|
|
|
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 * bpf;
|
|
size -= diff * bpf;
|
|
}
|
|
|
|
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 * bpf;
|
|
}
|
|
} 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 * bpf;
|
|
size -= diff * bpf;
|
|
}
|
|
|
|
diff = stop - cstop;
|
|
if (diff > 0) {
|
|
offset_end = cstop;
|
|
|
|
if (change_duration)
|
|
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
|
|
|
|
size -= diff * bpf;
|
|
}
|
|
} 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;
|
|
}
|