gstreamer/gst/aiff/aiffmux.c

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/* 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
*
* 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 "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,
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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")
);
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#define gst_aiff_mux_parent_class parent_class
G_DEFINE_TYPE (GstAiffMux, gst_aiff_mux, GST_TYPE_ELEMENT);
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) {
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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;
}
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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;
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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_pad_template (gstelement_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&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)
{
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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);
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gst_byte_writer_put_uint16_be_unchecked (writer, channels);
/* numSampleFrames value will be overwritten when known */
gst_byte_writer_put_uint32_be_unchecked (writer,
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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;
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GstByteWriter writer;
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GstSegment seg;
/* seek to beginning of file */
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gst_segment_init (&seg, GST_FORMAT_BYTES);
if (gst_pad_push_event (aiffmux->srcpad,
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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);
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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);
gst_aiff_mux_write_comm_header (aiffmux, audio_data_size, &writer);
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);
if (ret != GST_FLOW_OK) {
GST_WARNING_OBJECT (aiffmux, "push header failed: flow = %s",
gst_flow_get_name (ret));
}
return ret;
}
static GstFlowReturn
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gst_aiff_mux_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
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GstAiffMux *aiffmux = GST_AIFF_MUX (parent);
GstFlowReturn flow = GST_FLOW_OK;
guint64 cur_size;
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gsize buf_size;
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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);
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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;
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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;
}
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GST_LOG_OBJECT (aiffmux,
"pushing %" G_GSIZE_FORMAT " bytes raw audio, ts=%" GST_TIME_FORMAT,
buf_size, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
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buf = gst_buffer_make_writable (buf);
GST_BUFFER_OFFSET (buf) = AIFF_HEADER_LEN + aiffmux->length;
GST_BUFFER_OFFSET_END (buf) = GST_BUFFER_OFFSET_NONE;
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aiffmux->length += buf_size;
flow = gst_pad_push (aiffmux->srcpad, buf);
return flow;
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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;
}
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flow_error:
{
GST_DEBUG_OBJECT (aiffmux, "got flow error %s", gst_flow_get_name (flow));
gst_buffer_unref (buf);
return flow;
}
}
static gboolean
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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;
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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 */
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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;
}
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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:
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res = gst_pad_event_default (pad, parent, event);
break;
}
return res;
}
static void
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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);
}