gstreamer/ext/wavpack/gstwavpackdec.c
Sebastian Dröge 4586d0398e ext/wavpack/gstwavpackdec.c: Revert to use gst_pad_alloc_buffer() here. We can and should use it.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Revert to use gst_pad_alloc_buffer() here. We can and should use it.
Thanks to Jan and Mike for noticing my mistake.
2007-03-22 11:08:03 +00:00

498 lines
14 KiB
C

/* GStreamer Wavpack plugin
* Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
* Copyright (c) 2006 Edward Hervey <bilboed@gmail.com>
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstwavpackdec.c: raw Wavpack bitstream decoder
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-wavpackdec
*
* <refsect2>
* WavpackDec decodes framed (for example by the WavpackParse element)
* Wavpack streams and decodes them to raw audio.
* <ulink url="http://www.wavpack.com/">Wavpack</ulink> is an open-source
* audio codec that features both lossless and lossy encoding.
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch filesrc location=test.wv ! wavpackparse ! wavpackdec ! audioconvert ! audioresample ! autoaudiosink
* </programlisting>
* This pipeline decodes the Wavpack file test.wv into raw audio buffers and
* tries to play it back using an automatically found audio sink.
* </para>
* </refsect2>
*/
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <math.h>
#include <string.h>
#include <wavpack/wavpack.h>
#include "gstwavpackdec.h"
#include "gstwavpackcommon.h"
#include "gstwavpackstreamreader.h"
#define WAVPACK_DEC_MAX_ERRORS 16
GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
#define GST_CAT_DEFAULT gst_wavpack_dec_debug
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) { 8, 16, 24, 32 }, "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) { 8, 16, 32 }, "
"depth = (int) [ 8, 32 ], "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ], "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true")
);
static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
static void gst_wavpack_dec_finalize (GObject * object);
static GstStateChangeReturn gst_wavpack_dec_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT);
static void
gst_wavpack_dec_base_init (gpointer klass)
{
static const GstElementDetails plugin_details =
GST_ELEMENT_DETAILS ("Wavpack audio decoder",
"Codec/Decoder/Audio",
"Decodes Wavpack audio data",
"Arwed v. Merkatz <v.merkatz@gmx.net>, "
"Sebastian Dröge <slomo@circular-chaos.org>");
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_set_details (element_class, &plugin_details);
}
static void
gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_wavpack_dec_finalize);
}
static void
gst_wavpack_dec_reset (GstWavpackDec * dec)
{
dec->wv_id.buffer = NULL;
dec->wv_id.position = dec->wv_id.length = 0;
dec->error_count = 0;
dec->channels = 0;
dec->sample_rate = 0;
dec->width = 0;
dec->depth = 0;
gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
}
static void
gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass)
{
dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
gst_pad_set_chain_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_chain));
gst_pad_set_event_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event));
gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_pad_use_fixed_caps (dec->srcpad);
gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
dec->context = NULL;
dec->stream_reader = gst_wavpack_stream_reader_new ();
gst_wavpack_dec_reset (dec);
}
static void
gst_wavpack_dec_finalize (GObject * object)
{
GstWavpackDec *dec = GST_WAVPACK_DEC (object);
g_free (dec->stream_reader);
dec->stream_reader = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_wavpack_dec_format_samples (GstWavpackDec * dec, guint8 * out_buffer,
int32_t * samples, guint num_samples)
{
switch (dec->width) {
case 8:{
gint8 *dst = (gint8 *) out_buffer;
gint8 *end = dst + (num_samples * dec->channels);
while (dst < end) {
*dst++ = (gint8) * samples++;
}
break;
}
case 16:{
gint16 *dst = (gint16 *) out_buffer;
gint16 *end = dst + (num_samples * dec->channels);
while (dst < end) {
*dst++ = (gint16) * samples++;
}
break;
}
case 24:
case 32:{
gint32 *dst = (gint32 *) out_buffer;
gint32 *end = dst + (num_samples * dec->channels);
while (dst < end) {
*dst++ = *samples++;
}
break;
}
default:
g_return_if_reached ();
break;
}
}
static gboolean
gst_wavpack_dec_clip_outgoing_buffer (GstWavpackDec * dec, GstBuffer * buf)
{
gint64 start, stop, cstart, cstop, diff;
if (dec->segment.format != GST_FORMAT_TIME)
return TRUE;
start = GST_BUFFER_TIMESTAMP (buf);
stop = start + GST_BUFFER_DURATION (buf);
if (gst_segment_clip (&dec->segment, GST_FORMAT_TIME,
start, stop, &cstart, &cstop)) {
diff = cstart - start;
if (diff > 0) {
GST_BUFFER_TIMESTAMP (buf) = cstart;
GST_BUFFER_DURATION (buf) -= diff;
diff = (dec->width / 8) * dec->channels
* GST_CLOCK_TIME_TO_FRAMES (diff, dec->sample_rate);
GST_BUFFER_DATA (buf) += diff;
GST_BUFFER_SIZE (buf) -= diff;
}
diff = cstop - stop;
if (diff > 0) {
GST_BUFFER_DURATION (buf) -= diff;
diff = (dec->width / 8) * dec->channels
* GST_CLOCK_TIME_TO_FRAMES (diff, dec->sample_rate);
GST_BUFFER_SIZE (buf) -= diff;
}
} else {
GST_DEBUG_OBJECT (dec, "buffer is outside configured segment");
return FALSE;
}
return TRUE;
}
static void
gst_wavpack_dec_post_tags (GstWavpackDec * dec, WavpackHeader * wph)
{
GstTagList *list;
GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES;
gint64 duration, size;
list = gst_tag_list_new ();
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, "Wavpack", NULL);
/* try to estimate the average bitrate */
if (gst_pad_query_peer_duration (dec->sinkpad, &format_bytes, &size) &&
gst_pad_query_peer_duration (dec->sinkpad, &format_time, &duration) &&
size > 0 && duration > 0) {
guint64 bitrate;
bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration);
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
(guint) bitrate, NULL);
}
gst_element_post_message (GST_ELEMENT (dec),
gst_message_new_tag (GST_OBJECT (dec), list));
}
static GstFlowReturn
gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
{
GstWavpackDec *dec;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
WavpackHeader wph;
int32_t *unpack_buf = NULL;
int32_t decoded, unpacked_size;
gboolean format_changed;
dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
/* check input, we only accept framed input with complete chunks */
if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
goto input_not_framed;
if (!gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf)))
goto invalid_header;
if (GST_BUFFER_SIZE (buf) != wph.ckSize + 4 * 1 + 4)
goto input_not_framed;
dec->wv_id.buffer = GST_BUFFER_DATA (buf);
dec->wv_id.length = GST_BUFFER_SIZE (buf);
dec->wv_id.position = 0;
/* create a new wavpack context if there is none yet but if there
* was already one (i.e. caps were set on the srcpad) check whether
* the new one has the same caps */
if (!dec->context) {
gchar error_msg[80];
dec->context = WavpackOpenFileInputEx (dec->stream_reader,
&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
if (!dec->context) {
GST_WARNING ("Couldn't decode buffer: %s", error_msg);
dec->error_count++;
if (dec->error_count <= WAVPACK_DEC_MAX_ERRORS) {
goto out; /* just return OK for now */
} else {
goto decode_error;
}
}
}
g_assert (dec->context != NULL);
dec->error_count = 0;
format_changed =
(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
(dec->channels != WavpackGetNumChannels (dec->context)) ||
(dec->depth != WavpackGetBitsPerSample (dec->context));
if (!GST_PAD_CAPS (dec->srcpad) || format_changed) {
GstCaps *caps;
dec->sample_rate = WavpackGetSampleRate (dec->context);
dec->channels = WavpackGetNumChannels (dec->context);
dec->depth = WavpackGetBitsPerSample (dec->context);
dec->width =
(GST_ROUND_UP_8 (dec->depth) == 24) ? 32 : GST_ROUND_UP_8 (dec->depth);
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->channels,
"depth", G_TYPE_INT, dec->depth,
"width", G_TYPE_INT, dec->width,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
/* should always succeed */
gst_pad_set_caps (dec->srcpad, caps);
gst_caps_unref (caps);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
gst_wavpack_dec_post_tags (dec, &wph);
}
/* decode */
unpack_buf = g_new (int32_t, wph.block_samples * dec->channels);
decoded = WavpackUnpackSamples (dec->context, unpack_buf, wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
/* alloc output buffer */
unpacked_size = wph.block_samples * (dec->width / 8) * dec->channels;
ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf),
unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf);
if (ret != GST_FLOW_OK)
goto out;
/* put samples into the output buffer */
gst_wavpack_dec_format_samples (dec, GST_BUFFER_DATA (outbuf),
unpack_buf, wph.block_samples);
gst_buffer_stamp (outbuf, buf);
if (gst_wavpack_dec_clip_outgoing_buffer (dec, outbuf)) {
GST_LOG_OBJECT (dec, "pushing buffer with time %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
ret = gst_pad_push (dec->srcpad, outbuf);
} else {
gst_buffer_unref (outbuf);
}
out:
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
}
g_free (unpack_buf);
gst_buffer_unref (buf);
return ret;
/* ERRORS */
input_not_framed:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
invalid_header:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
decode_error:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Failed to decode wavpack stream"));
g_free (unpack_buf);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static gboolean
gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
{
GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
GST_LOG_OBJECT (dec, "Received %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:{
GstFormat fmt;
gboolean is_update;
gint64 start, end, base;
gdouble rate;
gst_event_parse_new_segment (event, &is_update, &rate, &fmt, &start,
&end, &base);
if (fmt == GST_FORMAT_TIME) {
GST_DEBUG ("Got NEWSEGMENT event in GST_FORMAT_TIME, passing on (%"
GST_TIME_FORMAT " - %" GST_TIME_FORMAT ")", GST_TIME_ARGS (start),
GST_TIME_ARGS (end));
gst_segment_set_newsegment (&dec->segment, is_update, rate, fmt,
start, end, base);
} else {
gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
}
break;
}
default:
break;
}
gst_object_unref (dec);
return gst_pad_event_default (pad, event);
}
static GstStateChangeReturn
gst_wavpack_dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstWavpackDec *dec = GST_WAVPACK_DEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (dec->context) {
WavpackCloseFile (dec->context);
dec->context = NULL;
}
gst_wavpack_dec_reset (dec);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
gboolean
gst_wavpack_dec_plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "wavpackdec",
GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC))
return FALSE;
GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpack_dec", 0,
"Wavpack decoder");
return TRUE;
}