gstreamer/ext/wavpack/gstwavpackdec.c
Vineeth TM 1d23930cd7 wavpackdec: fix taglist memory leak
When passing the taglist to gst_audio_decoder_merge_tags, the reference is increased
by audiodecoder and the caller should free the taglist being passed.

https://bugzilla.gnome.org/show_bug.cgi?id=753903
2015-08-21 10:22:14 +03:00

475 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-wavpackdec
*
* 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.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 filesrc location=test.wv ! wavpackparse ! wavpackdec ! audioconvert ! audioresample ! autoaudiosink
* ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers and
* tries to play it back using an automatically found audio sink.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#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"
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, "
"depth = (int) [ 1, 32 ], "
"channels = (int) [ 1, 8 ], "
"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, "
"format = (string) S8, "
"layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ]; "
"audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ]; "
"audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S32) ", "
"layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
);
static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec);
static gboolean gst_wavpack_dec_stop (GstAudioDecoder * dec);
static gboolean gst_wavpack_dec_set_format (GstAudioDecoder * dec,
GstCaps * caps);
static GstFlowReturn gst_wavpack_dec_handle_frame (GstAudioDecoder * dec,
GstBuffer * buffer);
static void gst_wavpack_dec_finalize (GObject * object);
static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
#define gst_wavpack_dec_parent_class parent_class
G_DEFINE_TYPE (GstWavpackDec, gst_wavpack_dec, GST_TYPE_AUDIO_DECODER);
static void
gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *element_class = (GstElementClass *) (klass);
GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (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_static_metadata (element_class, "Wavpack audio decoder",
"Codec/Decoder/Audio",
"Decodes Wavpack audio data",
"Arwed v. Merkatz <v.merkatz@gmx.net>, "
"Sebastian Dröge <slomo@circular-chaos.org>");
gobject_class->finalize = gst_wavpack_dec_finalize;
base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_dec_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_dec_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_dec_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_dec_handle_frame);
}
static void
gst_wavpack_dec_reset (GstWavpackDec * dec)
{
dec->wv_id.buffer = NULL;
dec->wv_id.position = dec->wv_id.length = 0;
dec->channels = 0;
dec->channel_mask = 0;
dec->sample_rate = 0;
dec->depth = 0;
}
static void
gst_wavpack_dec_init (GstWavpackDec * dec)
{
dec->context = NULL;
dec->stream_reader = gst_wavpack_stream_reader_new ();
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
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 gboolean
gst_wavpack_dec_start (GstAudioDecoder * dec)
{
GST_DEBUG_OBJECT (dec, "start");
/* never mind a few errors */
gst_audio_decoder_set_max_errors (dec, 16);
/* don't bother us with flushing */
gst_audio_decoder_set_drainable (dec, FALSE);
/* aim for some perfect timestamping */
gst_audio_decoder_set_tolerance (dec, 10 * GST_MSECOND);
return TRUE;
}
static gboolean
gst_wavpack_dec_stop (GstAudioDecoder * dec)
{
GstWavpackDec *wpdec = GST_WAVPACK_DEC (dec);
GST_DEBUG_OBJECT (dec, "stop");
if (wpdec->context) {
WavpackCloseFile (wpdec->context);
wpdec->context = NULL;
}
gst_wavpack_dec_reset (wpdec);
return TRUE;
}
static void
gst_wavpack_dec_negotiate (GstWavpackDec * dec)
{
GstAudioInfo info;
GstAudioFormat fmt;
GstAudioChannelPosition pos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
/* arrange for 1, 2 or 4-byte width == depth output */
dec->width = dec->depth;
switch (dec->depth) {
case 8:
fmt = GST_AUDIO_FORMAT_S8;
break;
case 16:
fmt = _GST_AUDIO_FORMAT_NE (S16);
break;
case 24:
case 32:
fmt = _GST_AUDIO_FORMAT_NE (S32);
dec->width = 32;
break;
default:
fmt = GST_AUDIO_FORMAT_UNKNOWN;
g_assert_not_reached ();
break;
}
g_assert (dec->channel_mask != 0);
if (!gst_wavpack_get_channel_positions (dec->channels,
dec->channel_mask, pos))
GST_WARNING_OBJECT (dec, "Failed to set channel layout");
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, fmt, dec->sample_rate, dec->channels, pos);
gst_audio_channel_positions_to_valid_order (info.position, info.channels);
gst_audio_get_channel_reorder_map (info.channels,
info.position, pos, dec->channel_reorder_map);
/* should always succeed */
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
}
static gboolean
gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
/* pretty much nothing to do here,
* we'll parse it all from the stream and setup then */
return TRUE;
}
static void
gst_wavpack_dec_post_tags (GstWavpackDec * dec)
{
GstTagList *list;
GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES;
gint64 duration, size;
/* try to estimate the average bitrate */
if (gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
format_bytes, &size) &&
gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
format_time, &duration) && size > 0 && duration > 0) {
guint64 bitrate;
list = gst_tag_list_new_empty ();
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_audio_decoder_merge_tags (GST_AUDIO_DECODER (dec), list,
GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (list);
}
}
static GstFlowReturn
gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstWavpackDec *dec;
GstBuffer *outbuf = NULL;
GstFlowReturn ret = GST_FLOW_OK;
WavpackHeader wph;
int32_t decoded, unpacked_size;
gboolean format_changed;
gint width, depth, i, j, max;
gint32 *dec_data = NULL;
guint8 *out_data;
GstMapInfo map, omap;
dec = GST_WAVPACK_DEC (bdec);
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
gst_buffer_map (buf, &map, GST_MAP_READ);
/* check input, we only accept framed input with complete chunks */
if (map.size < sizeof (WavpackHeader))
goto input_not_framed;
if (!gst_wavpack_read_header (&wph, map.data))
goto invalid_header;
if (map.size < wph.ckSize + 4 * 1 + 4)
goto input_not_framed;
if (!(wph.flags & INITIAL_BLOCK))
goto input_not_framed;
dec->wv_id.buffer = map.data;
dec->wv_id.length = map.size;
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);
/* expect this to work */
if (!dec->context) {
GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg);
goto context_failed;
}
}
g_assert (dec->context != NULL);
format_changed =
(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
(dec->channels != WavpackGetNumChannels (dec->context)) ||
(dec->depth != WavpackGetBytesPerSample (dec->context) * 8) ||
(dec->channel_mask != WavpackGetChannelMask (dec->context));
if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (dec)) ||
format_changed) {
gint channel_mask;
dec->sample_rate = WavpackGetSampleRate (dec->context);
dec->channels = WavpackGetNumChannels (dec->context);
dec->depth = WavpackGetBytesPerSample (dec->context) * 8;
channel_mask = WavpackGetChannelMask (dec->context);
if (channel_mask == 0)
channel_mask = gst_wavpack_get_default_channel_mask (dec->channels);
dec->channel_mask = channel_mask;
gst_wavpack_dec_negotiate (dec);
/* 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);
}
/* alloc output buffer */
dec_data = g_malloc (4 * wph.block_samples * dec->channels);
/* decode */
decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
outbuf = gst_buffer_new_and_alloc (unpacked_size);
/* legacy; pass along offset, whatever that might entail */
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET (buf);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = omap.data;
width = dec->width;
depth = dec->depth;
max = dec->channels * wph.block_samples;
if (width == 8) {
gint8 *outbuffer = (gint8 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint8) (dec_data[i + reorder_map[j]]);
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint16) (dec_data[i + reorder_map[j]]);
}
} else if (dec->width == 32) {
gint32 *outbuffer = (gint32 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
if (width != depth) {
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ =
(gint32) (dec_data[i + reorder_map[j]] << (width - depth));
}
} else {
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint32) (dec_data[i + reorder_map[j]]);
}
}
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unmap (buf, &map);
buf = NULL;
g_free (dec_data);
ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
out:
if (buf)
gst_buffer_unmap (buf, &map);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
}
return ret;
/* ERRORS */
input_not_framed:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
ret = GST_FLOW_ERROR;
goto out;
}
invalid_header:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
ret = GST_FLOW_ERROR;
goto out;
}
context_failed:
{
GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL),
("error creating Wavpack context"), ret);
goto out;
}
decode_error:
{
const gchar *reason = "unknown";
if (dec->context) {
reason = WavpackGetErrorMessage (dec->context);
} else {
reason = "couldn't create decoder context";
}
GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
("decoding error: %s", reason), ret);
g_free (dec_data);
if (ret == GST_FLOW_OK)
gst_audio_decoder_finish_frame (bdec, NULL, 1);
goto out;
}
}
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, "wavpackdec", 0,
"Wavpack decoder");
return TRUE;
}