FaSTTUBe/GST-Tensordecoder-ov_ep
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wavpackdec: port to 0.11

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This commit is contained in:
Mark Nauwelaerts 2012-03-05 13:36:39 +01:00
parent e0c7004667
commit 2f7d8f55a4
3 changed files with 109 additions and 114 deletions
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1
configure.ac
View File

@ -316,7 +316,6 @@ GST_PLUGINS_NONPORTED="deinterlace interleave flx \
videobox \ videobox \
cairo cairo_gobject dv1394 \ cairo cairo_gobject dv1394 \
oss oss4 \ oss oss4 \
wavpack \
osx_video osx_audio " osx_video osx_audio "
AC_SUBST(GST_PLUGINS_NONPORTED) AC_SUBST(GST_PLUGINS_NONPORTED)

219
ext/wavpack/gstwavpackdec.c
View File

@ -44,7 +44,6 @@
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/audio/audio.h> #include <gst/audio/audio.h>
#include <gst/audio/multichannel.h>
#include <math.h> #include <math.h>
#include <string.h> #include <string.h>
@ -62,7 +61,7 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_SINK,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, " GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) [ 1, 32 ], " "depth = (int) [ 1, 32 ], "
"channels = (int) [ 1, 8 ], " "channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) true") "rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
); );
@ -70,21 +69,20 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_SRC,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, " GST_STATIC_CAPS ("audio/x-raw, "
"width = (int) 8, depth = (int) 8, " "format = (string) S8, "
"layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], " "channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "rate = (int) [ 6000, 192000 ]; "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true; " "audio/x-raw, "
"audio/x-raw-int, " "format = (string) " GST_AUDIO_NE (S16) ", "
"width = (int) 16, depth = (int) 16, " "layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], " "channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "rate = (int) [ 6000, 192000 ]; "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true; " "audio/x-raw, "
"audio/x-raw-int, " "format = (string) " GST_AUDIO_NE (S32) ", "
"width = (int) 32, depth = (int) 32, " "layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], " "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
"rate = (int) [ 6000, 192000 ], "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true; ")
); );
static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec); static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec);
@ -97,13 +95,15 @@ static GstFlowReturn gst_wavpack_dec_handle_frame (GstAudioDecoder * dec,
static void gst_wavpack_dec_finalize (GObject * object); static void gst_wavpack_dec_finalize (GObject * object);
static void gst_wavpack_dec_post_tags (GstWavpackDec * dec); static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstAudioDecoder, #define gst_wavpack_dec_parent_class parent_class
GST_TYPE_AUDIO_DECODER); G_DEFINE_TYPE (GstWavpackDec, gst_wavpack_dec, GST_TYPE_AUDIO_DECODER);
static void static void
gst_wavpack_dec_base_init (gpointer klass) gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
{ {
GstElementClass *element_class = GST_ELEMENT_CLASS (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_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory)); gst_static_pad_template_get (&src_factory));
@ -114,13 +114,6 @@ gst_wavpack_dec_base_init (gpointer klass)
"Decodes Wavpack audio data", "Decodes Wavpack audio data",
"Arwed v. Merkatz <v.merkatz@gmx.net>, " "Arwed v. Merkatz <v.merkatz@gmx.net>, "
"Sebastian Dröge <slomo@circular-chaos.org>"); "Sebastian Dröge <slomo@circular-chaos.org>");
}
static void
gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass);
gobject_class->finalize = gst_wavpack_dec_finalize; gobject_class->finalize = gst_wavpack_dec_finalize;
@ -143,7 +136,7 @@ gst_wavpack_dec_reset (GstWavpackDec * dec)
} }
static void static void
gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass) gst_wavpack_dec_init (GstWavpackDec * dec)
{ {
dec->context = NULL; dec->context = NULL;
dec->stream_reader = gst_wavpack_stream_reader_new (); dec->stream_reader = gst_wavpack_stream_reader_new ();
@ -197,69 +190,51 @@ gst_wavpack_dec_stop (GstAudioDecoder * dec)
static void static void
gst_wavpack_dec_negotiate (GstWavpackDec * dec) gst_wavpack_dec_negotiate (GstWavpackDec * dec)
{ {
GstCaps *caps; GstAudioInfo info;
GstAudioFormat fmt;
GstAudioChannelPosition pos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
/* arrange for 1, 2 or 4-byte width == depth output */ /* arrange for 1, 2 or 4-byte width == depth output */
if (dec->depth == 24) dec->width = dec->depth;
dec->width = 32; switch (dec->depth) {
else case 8:
dec->width = dec->depth; 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:
g_assert_not_reached ();
break;
}
caps = gst_caps_new_simple ("audio/x-raw-int", g_assert (dec->channel_mask != 0);
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->channels,
"depth", G_TYPE_INT, dec->width,
"width", G_TYPE_INT, dec->width,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
/* Only set the channel layout for more than two channels if (!gst_wavpack_get_channel_positions (dec->channels,
* otherwise things break unfortunately */ dec->channel_mask, pos))
if (dec->channel_mask != 0 && dec->channels > 2) GST_WARNING_OBJECT (dec, "Failed to set channel layout");
if (!gst_wavpack_set_channel_layout (caps, dec->channel_mask))
GST_WARNING_OBJECT (dec, "Failed to set channel layout");
GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps); 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 */ /* should always succeed */
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps); gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
gst_caps_unref (caps);
} }
static gboolean static gboolean
gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps) gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{ {
GstWavpackDec *dec = GST_WAVPACK_DEC (bdec); /* pretty much nothing to do here,
GstStructure *structure = gst_caps_get_structure (caps, 0); * we'll parse it all from the stream and setup then */
/* Check if we can set the caps here already */
if (gst_structure_get_int (structure, "channels", &dec->channels) &&
gst_structure_get_int (structure, "rate", &dec->sample_rate) &&
gst_structure_get_int (structure, "width", &dec->depth)) {
GstAudioChannelPosition *pos;
/* If we already have the channel layout set from upstream
* take this */
if (gst_structure_has_field (structure, "channel-positions")) {
pos = gst_audio_get_channel_positions (structure);
if (pos != NULL && dec->channels > 2) {
GstStructure *new_str = gst_caps_get_structure (caps, 0);
gst_audio_set_channel_positions (new_str, pos);
dec->channel_mask =
gst_wavpack_get_channel_mask_from_positions (pos, dec->channels);
}
if (pos != NULL)
g_free (pos);
}
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);
}
return TRUE; return TRUE;
} }
@ -272,13 +247,13 @@ gst_wavpack_dec_post_tags (GstWavpackDec * dec)
gint64 duration, size; gint64 duration, size;
/* try to estimate the average bitrate */ /* try to estimate the average bitrate */
if (gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec), if (gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
&format_bytes, &size) && format_bytes, &size) &&
gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec), gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
&format_time, &duration) && size > 0 && duration > 0) { format_time, &duration) && size > 0 && duration > 0) {
guint64 bitrate; guint64 bitrate;
list = gst_tag_list_new (); list = gst_tag_list_new_empty ();
bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration); bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration);
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE, gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
@ -298,29 +273,32 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
WavpackHeader wph; WavpackHeader wph;
int32_t decoded, unpacked_size; int32_t decoded, unpacked_size;
gboolean format_changed; gboolean format_changed;
gint width, depth, i, max; gint width, depth, i, j, max;
gint32 *dec_data = NULL; gint32 *dec_data = NULL;
guint8 *out_data; guint8 *out_data;
GstMapInfo map, omap;
dec = GST_WAVPACK_DEC (bdec); dec = GST_WAVPACK_DEC (bdec);
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR); 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 */ /* check input, we only accept framed input with complete chunks */
if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader)) if (map.size < sizeof (WavpackHeader))
goto input_not_framed; goto input_not_framed;
if (!gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf))) if (!gst_wavpack_read_header (&wph, map.data))
goto invalid_header; goto invalid_header;
if (GST_BUFFER_SIZE (buf) < wph.ckSize + 4 * 1 + 4) if (map.size < wph.ckSize + 4 * 1 + 4)
goto input_not_framed; goto input_not_framed;
if (!(wph.flags & INITIAL_BLOCK)) if (!(wph.flags & INITIAL_BLOCK))
goto input_not_framed; goto input_not_framed;
dec->wv_id.buffer = GST_BUFFER_DATA (buf); dec->wv_id.buffer = map.data;
dec->wv_id.length = GST_BUFFER_SIZE (buf); dec->wv_id.length = map.size;
dec->wv_id.position = 0; dec->wv_id.position = 0;
/* create a new wavpack context if there is none yet but if there /* create a new wavpack context if there is none yet but if there
@ -351,7 +329,8 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
(dec->channel_mask != WavpackGetChannelMask (dec->context)); (dec->channel_mask != WavpackGetChannelMask (dec->context));
#endif #endif
if (!GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)) || format_changed) { if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (dec)) ||
format_changed) {
gint channel_mask; gint channel_mask;
dec->sample_rate = WavpackGetSampleRate (dec->context); dec->sample_rate = WavpackGetSampleRate (dec->context);
@ -376,58 +355,72 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
} }
/* alloc output buffer */ /* alloc output buffer */
unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
ret = gst_pad_alloc_buffer (GST_AUDIO_DECODER_SRC_PAD (dec),
GST_BUFFER_OFFSET (buf), unpacked_size,
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
if (ret != GST_FLOW_OK)
goto out;
dec_data = g_malloc (4 * wph.block_samples * dec->channels); dec_data = g_malloc (4 * wph.block_samples * dec->channels);
out_data = GST_BUFFER_DATA (outbuf);
/* decode */ /* decode */
decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples); decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
if (decoded != wph.block_samples) if (decoded != wph.block_samples)
goto decode_error; 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; width = dec->width;
depth = dec->depth; depth = dec->depth;
max = dec->channels * wph.block_samples; max = dec->channels * wph.block_samples;
if (width == 8) { if (width == 8) {
gint8 *outbuffer = (gint8 *) out_data; gint8 *outbuffer = (gint8 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i++) { for (i = 0; i < max; i += dec->channels) {
*outbuffer++ = (gint8) (dec_data[i]); for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint8) (dec_data[i + reorder_map[j]]);
} }
} else if (width == 16) { } else if (width == 16) {
gint16 *outbuffer = (gint16 *) out_data; gint16 *outbuffer = (gint16 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i++) { for (i = 0; i < max; i += dec->channels) {
*outbuffer++ = (gint16) (dec_data[i]); for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint16) (dec_data[i + reorder_map[j]]);
} }
} else if (dec->width == 32) { } else if (dec->width == 32) {
gint32 *outbuffer = (gint32 *) out_data; gint32 *outbuffer = (gint32 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
if (width != depth) { if (width != depth) {
for (i = 0; i < max; i++) { for (i = 0; i < max; i += dec->channels) {
*outbuffer++ = (gint32) (dec_data[i] << (width - depth)); for (j = 0; j < dec->channels; j++)
*outbuffer++ =
(gint32) (dec_data[i + reorder_map[j]] << (width - depth));
} }
} else { } else {
for (i = 0; i < max; i++) { for (i = 0; i < max; i += dec->channels) {
*outbuffer++ = (gint32) dec_data[i]; for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint32) (dec_data[i + reorder_map[j]]);
} }
} }
} else { } else {
g_assert_not_reached (); g_assert_not_reached ();
} }
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unmap (buf, &map);
buf = NULL;
g_free (dec_data); g_free (dec_data);
ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1); ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
out: out:
if (buf)
gst_buffer_unmap (buf, &map);
if (G_UNLIKELY (ret != GST_FLOW_OK)) { if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret)); GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
@ -439,12 +432,14 @@ out:
input_not_framed: input_not_framed:
{ {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input")); GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
return GST_FLOW_ERROR; ret = GST_FLOW_ERROR;
goto out;
} }
invalid_header: invalid_header:
{ {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header")); GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
return GST_FLOW_ERROR; ret = GST_FLOW_ERROR;
goto out;
} }
context_failed: context_failed:
{ {
@ -468,11 +463,9 @@ decode_error:
GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL), GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
("decoding error: %s", reason), ret); ("decoding error: %s", reason), ret);
g_free (dec_data); g_free (dec_data);
if (outbuf)
gst_buffer_unref (outbuf);
if (ret == GST_FLOW_OK) if (ret == GST_FLOW_OK)
gst_audio_decoder_finish_frame (bdec, NULL, 1); gst_audio_decoder_finish_frame (bdec, NULL, 1);
return ret; goto out;
} }
} }
@ -482,7 +475,7 @@ gst_wavpack_dec_plugin_init (GstPlugin * plugin)
if (!gst_element_register (plugin, "wavpackdec", if (!gst_element_register (plugin, "wavpackdec",
GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC)) GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC))
return FALSE; return FALSE;
GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpack_dec", 0, GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpackdec", 0,
"Wavpack decoder"); "Wavpack decoder");
return TRUE; return TRUE;
} }

3
ext/wavpack/gstwavpackdec.h
View File

@ -60,6 +60,9 @@ struct _GstWavpackDec
gint width; gint width;
gint channels; gint channels;
gint channel_mask; gint channel_mask;
gint channel_reorder_map[64];
}; };
struct _GstWavpackDecClass struct _GstWavpackDecClass