gstreamer/ext/sndfile/gstsfdec.c
Stefan Sauer 5f89bee749 sndfile: rewrite sndfile for 1.0
Add a sfdec for a start. Instead of a source plugin, this is a demuxer/decoder combination. This makes it work with auto-plugging.
2013-12-20 20:00:54 +01:00

448 lines
12 KiB
C

/* GStreamer libsndfile plugin
* Copyright (C) 2013 Stefan Sauer <ensonic@users.sf.net>
*
* 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 self library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <gst/gst-i18n-plugin.h>
#include "gstsfdec.h"
#include <gst/audio/audio.h>
enum
{
PROP_0,
PROP_LOCATION
};
#define FORMATS \
"{ "GST_AUDIO_NE (F32)", "GST_AUDIO_NE (S32)", "GST_AUDIO_NE (S16)" }"
static GstStaticPadTemplate sf_dec_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " FORMATS ", "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]"));
GST_DEBUG_CATEGORY_STATIC (gst_sf_dec_debug);
#define GST_CAT_DEFAULT gst_sf_dec_debug
#define DEFAULT_BUFFER_FRAMES (256)
static GstStateChangeReturn gst_sf_dec_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_sf_dec_sink_activate (GstPad * pad, GstObject * parent);
static gboolean gst_sf_dec_sink_activate_mode (GstPad * sinkpad,
GstObject * parent, GstPadMode mode, gboolean active);
static void gst_sf_dec_loop (GstPad * pad);
static gboolean gst_sf_dec_start (GstSFDec * bsrc);
static gboolean gst_sf_dec_stop (GstSFDec * bsrc);
#define _do_init \
GST_DEBUG_CATEGORY_INIT (gst_sf_dec_debug, "sfdec", 0, "sfdec element");
#define gst_sf_dec_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstSFDec, gst_sf_dec, GST_TYPE_ELEMENT, _do_init);
/* sf virtual io */
static sf_count_t
gst_sf_vio_get_filelen (void *user_data)
{
GstElement *element = GST_ELEMENT (user_data);
gint64 dur;
if (gst_element_query_duration (element, GST_FORMAT_BYTES, &dur)) {
return (sf_count_t) dur;
}
GST_WARNING_OBJECT (element, "query_duration failed");
return -1;
}
static sf_count_t
gst_sf_vio_tell (void *user_data)
{
GstSFDec *self = GST_SF_DEC (user_data);
return self->pos;
}
static sf_count_t
gst_sf_vio_seek (sf_count_t offset, int whence, void *user_data)
{
GstSFDec *self = GST_SF_DEC (user_data);
switch (whence) {
case SEEK_CUR:
self->pos += offset;
break;
case SEEK_SET:
self->pos = offset;
break;
case SEEK_END:
self->pos = gst_sf_vio_get_filelen (user_data) - offset;
break;
}
return (sf_count_t) self->pos;
}
static sf_count_t
gst_sf_vio_read (void *ptr, sf_count_t count, void *user_data)
{
GstSFDec *self = GST_SF_DEC (user_data);
GstBuffer *buffer = gst_buffer_new_wrapped_full (0, ptr, count, 0, count,
ptr, NULL);
if (gst_pad_pull_range (self->sinkpad, self->pos, count, &buffer) ==
GST_FLOW_OK) {
GST_DEBUG_OBJECT (self, "read %d bytes @ pos %" G_GUINT64_FORMAT,
(gint) count, self->pos);
self->pos += count;
return count;
}
GST_WARNING_OBJECT (self, "read failed");
return 0;
}
static sf_count_t
gst_sf_vio_write (const void *ptr, sf_count_t count, void *user_data)
{
GstSFDec *self = GST_SF_DEC (user_data);
GstBuffer *buffer = gst_buffer_new_wrapped (g_memdup (ptr, count), count);
if (gst_pad_push (self->srcpad, buffer) == GST_FLOW_OK) {
return count;
}
GST_WARNING_OBJECT (self, "write failed");
return 0;
}
SF_VIRTUAL_IO gst_sf_vio = {
&gst_sf_vio_get_filelen,
&gst_sf_vio_seek,
&gst_sf_vio_read,
&gst_sf_vio_write,
&gst_sf_vio_tell,
};
static void
gst_sf_dec_class_init (GstSFDecClass * klass)
{
GstElementClass *gstelement_class;
gstelement_class = GST_ELEMENT_CLASS (klass);
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_sf_dec_change_state);
gst_element_class_set_static_metadata (gstelement_class, "Sndfile decoder",
"Demuxer/Audio",
"Read audio streams using libsndfile",
"Stefan Sauer <ensonic@user.sf.net>");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sf_dec_src_factory));
gst_element_class_add_pad_template (gstelement_class,
gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
gst_sf_create_audio_template_caps ()));
}
static void
gst_sf_dec_init (GstSFDec * self)
{
self->sinkpad = gst_pad_new_from_template (gst_element_class_get_pad_template
(GST_ELEMENT_GET_CLASS (self), "sink"), "sink");
gst_pad_set_activate_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_sf_dec_sink_activate));
gst_pad_set_activatemode_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_sf_dec_sink_activate_mode));
gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);
self->srcpad = gst_pad_new_from_static_template (&sf_dec_src_factory, "src");
/* TODO(ensonic): event + query functions */
gst_element_add_pad (GST_ELEMENT (self), self->srcpad);
}
static GstStateChangeReturn
gst_sf_dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstSFDec *self = GST_SF_DEC (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_sf_dec_start (self);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_sf_dec_stop (self);
break;
default:
break;
}
return ret;
}
static gboolean
gst_sf_dec_start (GstSFDec * self)
{
self->file = NULL;
return TRUE;
}
static gboolean
gst_sf_dec_stop (GstSFDec * self)
{
int err = 0;
g_return_val_if_fail (self->file != NULL, FALSE);
GST_INFO_OBJECT (self, "Closing sndfile stream");
if ((err = sf_close (self->file)))
goto close_failed;
self->file = NULL;
self->offset = 0;
self->channels = 0;
self->rate = 0;
return TRUE;
close_failed:
{
GST_ELEMENT_ERROR (self, RESOURCE, CLOSE,
("Could not close sndfile stream."),
("soundfile error: %s", sf_error_number (err)));
return FALSE;
}
}
static gboolean
gst_sf_dec_sink_activate (GstPad * sinkpad, GstObject * parent)
{
GstQuery *query;
gboolean pull_mode;
query = gst_query_new_scheduling ();
if (!gst_pad_peer_query (sinkpad, query)) {
gst_query_unref (query);
goto activate_push;
}
pull_mode = gst_query_has_scheduling_mode_with_flags (query,
GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
gst_query_unref (query);
if (!pull_mode)
goto activate_push;
GST_DEBUG_OBJECT (sinkpad, "activating pull");
return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
activate_push:
{
GST_DEBUG_OBJECT (sinkpad, "activating push");
return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
}
}
static gboolean
gst_sf_dec_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
GstPadMode mode, gboolean active)
{
gboolean res;
switch (mode) {
case GST_PAD_MODE_PUSH:
res = FALSE; /* no push support */
break;
case GST_PAD_MODE_PULL:
if (active) {
/* if we have a scheduler we can start the task */
res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_sf_dec_loop,
sinkpad, NULL);
} else {
res = gst_pad_stop_task (sinkpad);
}
break;
default:
res = FALSE;
break;
}
return res;
}
static gboolean
gst_sf_dec_open_file (GstSFDec * self)
{
SF_INFO info;
GstCaps *caps;
GstStructure *s;
GstSegment seg;
gint width;
const gchar *format;
gchar *stream_id;
GST_DEBUG_OBJECT (self, "opening the stream");
info.format = 0;
self->file = sf_open_virtual (&gst_sf_vio, SFM_READ, &info, self);
if (!self->file)
goto open_failed;
stream_id =
gst_pad_create_stream_id (self->srcpad, GST_ELEMENT_CAST (self), NULL);
gst_pad_push_event (self->srcpad, gst_event_new_stream_start (stream_id));
g_free (stream_id);
self->channels = info.channels;
self->rate = info.samplerate;
/* TODO(ensonic): do something with info.seekable? */
/* TODO(ensonic): calculate duration info.frames */
/* negotiate srcpad caps */
if ((caps = gst_pad_get_allowed_caps (self->srcpad)) == NULL) {
caps = gst_pad_get_pad_template_caps (self->srcpad);
}
caps = gst_caps_make_writable (caps);
GST_DEBUG_OBJECT (self, "allowed caps %" GST_PTR_FORMAT, caps);
s = gst_caps_get_structure (caps, 0);
gst_structure_set (s,
"channels", G_TYPE_INT, self->channels,
"rate", G_TYPE_INT, self->rate, NULL);
if (!gst_structure_fixate_field_string (s, "format", GST_AUDIO_NE (S16)))
GST_WARNING_OBJECT (self, "Failed to fixate format to S16NE");
caps = gst_caps_fixate (caps);
GST_DEBUG_OBJECT (self, "fixated caps %" GST_PTR_FORMAT, caps);
/* configure to output the negotiated format */
s = gst_caps_get_structure (caps, 0);
format = gst_structure_get_string (s, "format");
if (g_str_equal (format, GST_AUDIO_NE (S32))) {
self->reader = (GstSFReader) sf_readf_int;
width = 32;
} else if (g_str_equal (format, GST_AUDIO_NE (S16))) {
self->reader = (GstSFReader) sf_readf_short;
width = 16;
} else {
self->reader = (GstSFReader) sf_readf_float;
width = 32;
}
self->bytes_per_frame = width * self->channels / 8;
gst_pad_set_caps (self->srcpad, caps);
gst_caps_unref (caps);
gst_segment_init (&seg, GST_FORMAT_TIME);
/* TODO: calculate seg.stop = song_length; */
gst_pad_push_event (self->srcpad, gst_event_new_segment (&seg));
return TRUE;
open_failed:
{
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ,
(_("Could not open sndfile stream for reading.")),
("soundfile error: %s", sf_strerror (NULL)));
return FALSE;
}
}
static void
gst_sf_dec_loop (GstPad * pad)
{
GstSFDec *self = GST_SF_DEC (GST_PAD_PARENT (pad));
GstBuffer *buf;
GstMapInfo map;
GstFlowReturn flow;
sf_count_t bytes_read;
guint num_frames = 1024; /* arbitrary */
if (G_UNLIKELY (!self->file)) {
/* not started yet */
if (!gst_sf_dec_open_file (self))
goto pause;
}
buf = gst_buffer_new_and_alloc (self->bytes_per_frame * num_frames);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
bytes_read = self->reader (self->file, map.data, num_frames);
GST_DEBUG_OBJECT (self, "read %d / %d bytes = %d frames of audio",
(gint) bytes_read, (gint) map.size, num_frames);
gst_buffer_unmap (buf, &map);
if (G_UNLIKELY (bytes_read < 0))
goto could_not_read;
if (G_UNLIKELY (bytes_read == 0))
goto eos;
num_frames = bytes_read / self->bytes_per_frame;
GST_BUFFER_OFFSET (buf) = self->offset;
GST_BUFFER_TIMESTAMP (buf) = gst_util_uint64_scale_int (self->offset,
GST_SECOND, self->rate);
self->offset += num_frames;
GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (self->offset,
GST_SECOND, self->rate) - GST_BUFFER_TIMESTAMP (buf);
flow = gst_pad_push (self->srcpad, buf);
if (flow != GST_FLOW_OK) {
GST_LOG_OBJECT (self, "pad push flow: %s", gst_flow_get_name (flow));
goto pause;
}
return;
/* ERROR */
could_not_read:
{
GST_ELEMENT_ERROR (self, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
gst_buffer_unref (buf);
goto pause;
}
eos:
{
GST_DEBUG_OBJECT (self, "EOS");
gst_buffer_unref (buf);
gst_pad_push_event (self->srcpad, gst_event_new_eos ());
goto pause;
}
pause:
{
GST_INFO_OBJECT (self, "Pausing");
gst_pad_pause_task (self->sinkpad);
}
}