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