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468 lines
14 KiB
C
468 lines
14 KiB
C
/* GStreamer AIFF muxer
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* Copyright (C) 2009 Robert Swain <robert.swain@gmail.com>
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*
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* Permission is hereby granted, free of charge, to any person obtaining a
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* copy of this software and associated documentation files (the "Software"),
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* to deal in the Software without restriction, including without limitation
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* the rights to use, copy, modify, merge, publish, distribute, sublicense,
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* and/or sell copies of the Software, and to permit persons to whom the
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* Software is furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
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* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
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* DEALINGS IN THE SOFTWARE.
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*
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* Alternatively, the contents of this file may be used under the
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* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
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* which case the following provisions apply instead of the ones
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* mentioned above:
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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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-aiffmux
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* @title: aiffmux
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*
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* Format an audio stream into the Audio Interchange File Format
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*
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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 <math.h>
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#include <gst/gst.h>
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#include <gst/base/gstbytewriter.h>
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#include "aiffelements.h"
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#include "aiffmux.h"
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GST_DEBUG_CATEGORY (aiffmux_debug);
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#define GST_CAT_DEFAULT aiffmux_debug
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = { S8, S16BE, S24BE, S32BE },"
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"channels = (int) [ 1, MAX ], " "rate = (int) [ 1, MAX ]")
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);
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static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-aiff")
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);
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#define gst_aiff_mux_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstAiffMux, gst_aiff_mux, GST_TYPE_ELEMENT,
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GST_DEBUG_CATEGORY_INIT (aiffmux_debug, "aiffmux", 0, "AIFF muxer"));
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (aiffmux, "aiffmux", GST_RANK_PRIMARY,
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GST_TYPE_AIFF_MUX, aiff_element_init (plugin));
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static GstStateChangeReturn
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gst_aiff_mux_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
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GstAiffMux *aiffmux = GST_AIFF_MUX (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_audio_info_init (&aiffmux->info);
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aiffmux->length = 0;
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aiffmux->sent_header = FALSE;
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aiffmux->overflow = FALSE;
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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if (ret != GST_STATE_CHANGE_SUCCESS)
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return ret;
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return ret;
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}
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static void
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gst_aiff_mux_class_init (GstAiffMuxClass * klass)
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{
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GstElementClass *gstelement_class;
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gstelement_class = (GstElementClass *) klass;
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gst_element_class_set_static_metadata (gstelement_class,
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"AIFF audio muxer", "Muxer/Audio", "Multiplex raw audio into AIFF",
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"Robert Swain <robert.swain@gmail.com>");
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gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
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gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_aiff_mux_change_state);
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}
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#define AIFF_FORM_HEADER_LEN 8 + 4
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#define AIFF_COMM_HEADER_LEN 8 + 18
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#define AIFF_SSND_HEADER_LEN 8 + 8
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#define AIFF_HEADER_LEN \
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(AIFF_FORM_HEADER_LEN + AIFF_COMM_HEADER_LEN + AIFF_SSND_HEADER_LEN)
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static void
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gst_aiff_mux_write_form_header (GstAiffMux * aiffmux, guint32 audio_data_size,
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GstByteWriter * writer)
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{
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guint64 cur_size;
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/* ckID == 'FORM' */
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gst_byte_writer_put_uint32_le_unchecked (writer,
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GST_MAKE_FOURCC ('F', 'O', 'R', 'M'));
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/* AIFF chunks must be even aligned */
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cur_size = AIFF_HEADER_LEN - 8 + audio_data_size;
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if ((cur_size & 1) && cur_size + 1 < G_MAXUINT32) {
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cur_size += 1;
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}
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gst_byte_writer_put_uint32_be_unchecked (writer, cur_size);
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/* formType == 'AIFF' */
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gst_byte_writer_put_uint32_le_unchecked (writer,
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GST_MAKE_FOURCC ('A', 'I', 'F', 'F'));
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}
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/*
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* BEGIN: Code borrowed from FFmpeg's libavutil/intfloat_readwrite.{c,h}
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* Copyright (c) 2005 Michael Niedermayer <michaelni@gmx.at>
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*/
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/* IEEE 80 bits extended float */
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typedef struct AVExtFloat
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{
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guint8 exponent[2];
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guint8 mantissa[8];
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} AVExtFloat;
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/* Courtesy http://www.devx.com/tips/Tip/42853 */
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static inline gint
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gst_aiff_mux_isinf (gdouble x)
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{
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volatile gdouble temp = x;
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if ((temp == x) && ((temp - x) != 0.0))
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return (x < 0.0 ? -1 : 1);
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else
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return 0;
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}
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static void
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gst_aiff_mux_write_ext (GstByteWriter * writer, double d)
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{
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struct AVExtFloat ext = { {0} };
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gint e, i;
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gdouble f;
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guint64 m;
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f = fabs (frexp (d, &e));
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if (f >= 0.5 && f < 1) {
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e += 16382;
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ext.exponent[0] = e >> 8;
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ext.exponent[1] = e;
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m = (guint64) ldexp (f, 64);
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for (i = 0; i < 8; i++)
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ext.mantissa[i] = m >> (56 - (i << 3));
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} else if (f != 0.0) {
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ext.exponent[0] = 0x7f;
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ext.exponent[1] = 0xff;
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if (!gst_aiff_mux_isinf (f))
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ext.mantissa[0] = ~0;
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}
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if (d < 0)
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ext.exponent[0] |= 0x80;
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gst_byte_writer_put_data_unchecked (writer, ext.exponent, 2);
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gst_byte_writer_put_data_unchecked (writer, ext.mantissa, 8);
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}
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/*
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* END: Code borrowed from FFmpeg's libavutil/intfloat_readwrite.{c,h}
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*/
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static void
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gst_aiff_mux_write_comm_header (GstAiffMux * aiffmux, guint32 audio_data_size,
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GstByteWriter * writer)
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{
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guint16 channels;
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guint16 width, depth;
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gdouble rate;
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channels = GST_AUDIO_INFO_CHANNELS (&aiffmux->info);
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width = GST_AUDIO_INFO_WIDTH (&aiffmux->info);
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depth = GST_AUDIO_INFO_DEPTH (&aiffmux->info);
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rate = GST_AUDIO_INFO_RATE (&aiffmux->info);
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gst_byte_writer_put_uint32_le_unchecked (writer,
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GST_MAKE_FOURCC ('C', 'O', 'M', 'M'));
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gst_byte_writer_put_uint32_be_unchecked (writer, 18);
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gst_byte_writer_put_uint16_be_unchecked (writer, channels);
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/* numSampleFrames value will be overwritten when known */
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gst_byte_writer_put_uint32_be_unchecked (writer,
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audio_data_size / (width / 8 * channels));
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gst_byte_writer_put_uint16_be_unchecked (writer, depth);
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gst_aiff_mux_write_ext (writer, rate);
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}
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static void
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gst_aiff_mux_write_ssnd_header (GstAiffMux * aiffmux, guint32 audio_data_size,
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GstByteWriter * writer)
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{
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gst_byte_writer_put_uint32_le_unchecked (writer,
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GST_MAKE_FOURCC ('S', 'S', 'N', 'D'));
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/* ckSize will be overwritten when known */
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gst_byte_writer_put_uint32_be_unchecked (writer,
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audio_data_size + AIFF_SSND_HEADER_LEN - 8);
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/* offset and blockSize are set to 0 as we don't support block-aligned sample data yet */
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gst_byte_writer_put_uint32_be_unchecked (writer, 0);
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gst_byte_writer_put_uint32_be_unchecked (writer, 0);
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}
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static GstFlowReturn
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gst_aiff_mux_push_header (GstAiffMux * aiffmux, guint32 audio_data_size)
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{
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GstFlowReturn ret;
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GstBuffer *outbuf;
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GstByteWriter writer;
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GstSegment seg;
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/* seek to beginning of file */
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gst_segment_init (&seg, GST_FORMAT_BYTES);
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if (gst_pad_push_event (aiffmux->srcpad,
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gst_event_new_segment (&seg)) == FALSE) {
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GST_ELEMENT_WARNING (aiffmux, STREAM, MUX,
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("An output stream seeking error occurred when multiplexing."),
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("Failed to seek to beginning of stream to write header."));
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}
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GST_DEBUG_OBJECT (aiffmux, "writing header with datasize=%u",
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audio_data_size);
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gst_byte_writer_init_with_size (&writer, AIFF_HEADER_LEN, TRUE);
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gst_aiff_mux_write_form_header (aiffmux, audio_data_size, &writer);
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gst_aiff_mux_write_comm_header (aiffmux, audio_data_size, &writer);
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gst_aiff_mux_write_ssnd_header (aiffmux, audio_data_size, &writer);
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outbuf = gst_byte_writer_reset_and_get_buffer (&writer);
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ret = gst_pad_push (aiffmux->srcpad, outbuf);
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if (ret != GST_FLOW_OK) {
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GST_WARNING_OBJECT (aiffmux, "push header failed: flow = %s",
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gst_flow_get_name (ret));
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}
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return ret;
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}
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static GstFlowReturn
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gst_aiff_mux_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
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{
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GstAiffMux *aiffmux = GST_AIFF_MUX (parent);
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GstFlowReturn flow = GST_FLOW_OK;
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guint64 cur_size;
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gsize buf_size;
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if (!GST_AUDIO_INFO_CHANNELS (&aiffmux->info))
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goto not_negotiated;
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if (G_UNLIKELY (aiffmux->overflow))
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goto overflow;
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if (!aiffmux->sent_header) {
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/* use bogus size initially, we'll write the real
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* header when we get EOS and know the exact length */
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flow = gst_aiff_mux_push_header (aiffmux, 0x7FFF0000);
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if (flow != GST_FLOW_OK)
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goto flow_error;
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GST_DEBUG_OBJECT (aiffmux, "wrote dummy header");
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aiffmux->sent_header = TRUE;
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}
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/* AIFF has an audio data size limit of slightly under 4 GB.
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A value of audiosize + AIFF_HEADER_LEN - 8 is written, so
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I'll error out if writing data that makes this overflow. */
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cur_size = aiffmux->length + AIFF_HEADER_LEN - 8;
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buf_size = gst_buffer_get_size (buf);
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if (G_UNLIKELY (cur_size + buf_size >= G_MAXUINT32)) {
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GST_ERROR_OBJECT (aiffmux, "AIFF only supports about 4 GB worth of "
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"audio data, dropping any further data on the floor");
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GST_ELEMENT_WARNING (aiffmux, STREAM, MUX, ("AIFF has a 4GB size limit"),
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("AIFF only supports about 4 GB worth of audio data, "
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"dropping any further data on the floor"));
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aiffmux->overflow = TRUE;
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goto overflow;
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}
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GST_LOG_OBJECT (aiffmux,
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"pushing %" G_GSIZE_FORMAT " bytes raw audio, ts=%" GST_TIME_FORMAT,
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buf_size, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
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buf = gst_buffer_make_writable (buf);
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GST_BUFFER_OFFSET (buf) = AIFF_HEADER_LEN + aiffmux->length;
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GST_BUFFER_OFFSET_END (buf) = GST_BUFFER_OFFSET_NONE;
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aiffmux->length += buf_size;
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flow = gst_pad_push (aiffmux->srcpad, buf);
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return flow;
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not_negotiated:
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{
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GST_WARNING_OBJECT (aiffmux, "no input format negotiated");
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gst_buffer_unref (buf);
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return GST_FLOW_NOT_NEGOTIATED;
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}
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overflow:
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{
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GST_WARNING_OBJECT (aiffmux, "output file too large, dropping buffer");
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gst_buffer_unref (buf);
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return GST_FLOW_OK;
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}
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flow_error:
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{
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GST_DEBUG_OBJECT (aiffmux, "got flow error %s", gst_flow_get_name (flow));
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gst_buffer_unref (buf);
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return flow;
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}
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}
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static gboolean
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gst_aiff_mux_set_caps (GstAiffMux * aiffmux, GstCaps * caps)
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{
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GstCaps *outcaps;
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GstAudioInfo info;
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if (aiffmux->sent_header) {
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GST_WARNING_OBJECT (aiffmux, "cannot change format mid-stream");
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return FALSE;
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}
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GST_DEBUG_OBJECT (aiffmux, "got caps: %" GST_PTR_FORMAT, caps);
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if (!gst_audio_info_from_caps (&info, caps)) {
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GST_WARNING_OBJECT (aiffmux, "caps incomplete");
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return FALSE;
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}
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aiffmux->info = info;
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GST_LOG_OBJECT (aiffmux,
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"accepted caps: chans=%d depth=%d rate=%d",
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GST_AUDIO_INFO_CHANNELS (&info), GST_AUDIO_INFO_DEPTH (&info),
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GST_AUDIO_INFO_RATE (&info));
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outcaps = gst_static_pad_template_get_caps (&src_factory);
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gst_pad_push_event (aiffmux->srcpad, gst_event_new_caps (outcaps));
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gst_caps_unref (outcaps);
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return TRUE;
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}
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static gboolean
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gst_aiff_mux_event (GstPad * pad, GstObject * parent, GstEvent * event)
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{
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gboolean res = TRUE;
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GstAiffMux *aiffmux;
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aiffmux = GST_AIFF_MUX (parent);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:{
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guint64 cur_size;
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GST_DEBUG_OBJECT (aiffmux, "got EOS");
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cur_size = aiffmux->length + AIFF_HEADER_LEN - 8;
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/* ID3 chunk must be even aligned */
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if ((aiffmux->length & 1) && cur_size + 1 < G_MAXUINT32) {
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GstFlowReturn ret;
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guint8 *data = g_new0 (guint8, 1);
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GstBuffer *buffer = gst_buffer_new_wrapped (data, 1);
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GST_BUFFER_OFFSET (buffer) = AIFF_HEADER_LEN + aiffmux->length;
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GST_BUFFER_OFFSET_END (buffer) = GST_BUFFER_OFFSET_NONE;
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ret = gst_pad_push (aiffmux->srcpad, buffer);
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if (ret != GST_FLOW_OK) {
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GST_WARNING_OBJECT (aiffmux, "failed to push padding byte: %s",
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gst_flow_get_name (ret));
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}
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}
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/* write header with correct length values */
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gst_aiff_mux_push_header (aiffmux, aiffmux->length);
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/* and forward the EOS event */
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res = gst_pad_event_default (pad, parent, event);
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break;
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}
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case GST_EVENT_CAPS:
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{
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GstCaps *caps;
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gst_event_parse_caps (event, &caps);
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res = gst_aiff_mux_set_caps (aiffmux, caps);
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gst_event_unref (event);
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break;
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}
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case GST_EVENT_SEGMENT:
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/* Just drop it, it's probably in TIME format
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* anyway. We'll send our own newsegment event */
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gst_event_unref (event);
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break;
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default:
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res = gst_pad_event_default (pad, parent, event);
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break;
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}
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return res;
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}
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static void
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gst_aiff_mux_init (GstAiffMux * aiffmux)
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{
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aiffmux->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
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gst_pad_set_chain_function (aiffmux->sinkpad,
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GST_DEBUG_FUNCPTR (gst_aiff_mux_chain));
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gst_pad_set_event_function (aiffmux->sinkpad,
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GST_DEBUG_FUNCPTR (gst_aiff_mux_event));
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gst_element_add_pad (GST_ELEMENT (aiffmux), aiffmux->sinkpad);
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aiffmux->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
|
|
gst_pad_use_fixed_caps (aiffmux->srcpad);
|
|
gst_element_add_pad (GST_ELEMENT (aiffmux), aiffmux->srcpad);
|
|
}
|