#include <gst/gst.h>
#include <gst/audio/audio.h>
-#include <gst/audio/multichannel.h>
#include <math.h>
#include <string.h>
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
- "width = (int) [ 1, 32 ], "
+ "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-int, "
- "width = (int) 8, depth = (int) 8, "
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) S8, "
+ "layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], "
- "rate = (int) [ 6000, 192000 ], "
- "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; "
- "audio/x-raw-int, "
- "width = (int) 16, depth = (int) 16, "
+ "rate = (int) [ 6000, 192000 ]; "
+ "audio/x-raw, "
+ "format = (string) " GST_AUDIO_NE (S16) ", "
+ "layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], "
- "rate = (int) [ 6000, 192000 ], "
- "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; "
- "audio/x-raw-int, "
- "width = (int) 32, depth = (int) 32, "
- "channels = (int) [ 1, 8 ], "
- "rate = (int) [ 6000, 192000 ], "
- "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; ")
+ "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 void gst_wavpack_dec_finalize (GObject * object);
static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
-GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstAudioDecoder,
- GST_TYPE_AUDIO_DECODER);
+#define gst_wavpack_dec_parent_class parent_class
+G_DEFINE_TYPE (GstWavpackDec, gst_wavpack_dec, GST_TYPE_AUDIO_DECODER);
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_static_pad_template_get (&src_factory));
"Decodes Wavpack audio data",
"Arwed v. Merkatz <v.merkatz@gmx.net>, "
"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;
}
static void
-gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass)
+gst_wavpack_dec_init (GstWavpackDec * dec)
{
dec->context = NULL;
dec->stream_reader = gst_wavpack_stream_reader_new ();
static void
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 */
- if (dec->depth == 24)
- dec->width = 32;
- else
- dec->width = 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->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
- * otherwise things break unfortunately */
- if (dec->channel_mask != 0 && dec->channels > 2)
- 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);
+ 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:
+ 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_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
- gst_caps_unref (caps);
+ gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
}
static gboolean
gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
- GstWavpackDec *dec = GST_WAVPACK_DEC (bdec);
- GstStructure *structure = gst_caps_get_structure (caps, 0);
-
- /* 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);
- }
+ /* pretty much nothing to do here,
+ * we'll parse it all from the stream and setup then */
return TRUE;
}
gint64 duration, size;
/* try to estimate the average bitrate */
- if (gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
- &format_bytes, &size) &&
- gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
- &format_time, &duration) && size > 0 && duration > 0) {
+ 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 ();
+ 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,
WavpackHeader wph;
int32_t decoded, unpacked_size;
gboolean format_changed;
- gint width, depth, i, max;
+ 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 (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
+ if (map.size < sizeof (WavpackHeader))
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;
- if (GST_BUFFER_SIZE (buf) < wph.ckSize + 4 * 1 + 4)
+ 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 = GST_BUFFER_DATA (buf);
- dec->wv_id.length = GST_BUFFER_SIZE (buf);
+ 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
(dec->channel_mask != WavpackGetChannelMask (dec->context));
#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;
dec->sample_rate = WavpackGetSampleRate (dec->context);
}
/* 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);
- out_data = GST_BUFFER_DATA (outbuf);
/* 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++) {
- *outbuffer++ = (gint8) (dec_data[i]);
+ 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++) {
- *outbuffer++ = (gint16) (dec_data[i]);
+ 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++) {
- *outbuffer++ = (gint32) (dec_data[i] << (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++) {
- *outbuffer++ = (gint32) dec_data[i];
+ 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));
input_not_framed:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
- return GST_FLOW_ERROR;
+ ret = GST_FLOW_ERROR;
+ goto out;
}
invalid_header:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
- return GST_FLOW_ERROR;
+ ret = GST_FLOW_ERROR;
+ goto out;
}
context_failed:
{
GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
("decoding error: %s", reason), ret);
g_free (dec_data);
- if (outbuf)
- gst_buffer_unref (outbuf);
if (ret == GST_FLOW_OK)
gst_audio_decoder_finish_frame (bdec, NULL, 1);
- return ret;
+ goto out;
}
}
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,
+ GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpackdec", 0,
"Wavpack decoder");
return TRUE;
}
projects / platform / upstream / gstreamer.git / commitdiff