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Merge branch 'master' into 0.11

Browse source 
Conflicts:
	ext/wavpack/gstwavpackenc.c
	tests/check/elements/audioiirfilter.c
	tests/examples/v4l2/probe.c
This commit is contained in:
Mark Nauwelaerts 2012-03-01 11:29:50 +01:00
commit f189f62b13
29 changed files with 1249 additions and 505 deletions
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4
ext/annodex/gstannodex.c
View file

@ -21,6 +21,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

4
ext/annodex/gstcmmlparser.c
View file

@ -21,6 +21,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include <stdarg.h>
#include <gst/gst.h>

4
ext/annodex/gstcmmltag.c
View file

@ -21,6 +21,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

4
ext/pulse/pulseprobe.c
View file

@ -21,6 +21,10 @@
* USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

358
ext/wavpack/gstwavpackdec.c
View file

@ -55,8 +55,6 @@
#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
@ -73,22 +71,34 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) [ 1, 32 ], "
"width = (int) 8, depth = (int) 8, "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true")
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true; "
"audio/x-raw-int, "
"width = (int) 16, depth = (int) 16, "
"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; ")
);
static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps);
static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
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 GstStateChangeReturn gst_wavpack_dec_change_state (GstElement * element,
GstStateChange transition);
static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT);
GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstAudioDecoder,
GST_TYPE_AUDIO_DECODER);
static void
gst_wavpack_dec_base_init (gpointer klass)
@ -110,11 +120,14 @@ static void
gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state);
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
@ -123,33 +136,15 @@ 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->channel_mask = 0;
dec->sample_rate = 0;
dec->depth = 0;
gst_segment_init (&dec->segment, GST_FORMAT_TIME);
dec->next_block_index = 0;
}
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_setcaps_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_set_caps));
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 ();
@ -168,26 +163,79 @@ gst_wavpack_dec_finalize (GObject * object)
}
static gboolean
gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
gst_wavpack_dec_start (GstAudioDecoder * dec)
{
GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
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);
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)
{
GstCaps *caps;
/* 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);
/* should always succeed */
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
gst_caps_unref (caps);
}
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)) {
GstCaps *caps;
GstAudioChannelPosition *pos;
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, 32,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
/* If we already have the channel layout set from upstream
* take this */
if (gst_structure_has_field (structure, "channel-positions")) {
@ -204,19 +252,13 @@ gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
g_free (pos);
}
GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
/* should always succeed */
gst_pad_set_caps (dec->srcpad, caps);
gst_caps_unref (caps);
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);
}
gst_object_unref (dec);
return TRUE;
}
@ -227,28 +269,26 @@ gst_wavpack_dec_post_tags (GstWavpackDec * dec)
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) {
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) {
guint64 bitrate;
list = gst_tag_list_new ();
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)
gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstWavpackDec *dec;
GstBuffer *outbuf = NULL;
@ -256,8 +296,13 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
WavpackHeader wph;
int32_t decoded, unpacked_size;
gboolean format_changed;
gint width, depth, i, max;
gint32 *dec_data = NULL;
guint8 *out_data;
dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
dec = GST_WAVPACK_DEC (bdec);
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
/* check input, we only accept framed input with complete chunks */
if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
@ -285,46 +330,31 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
dec->context = WavpackOpenFileInputEx (dec->stream_reader,
&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
/* expect this to work */
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;
}
GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg);
goto context_failed;
}
}
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)) ||
(dec->depth != WavpackGetBytesPerSample (dec->context) * 8) ||
#ifdef WAVPACK_OLD_API
(dec->channel_mask != dec->context->config.channel_mask);
#else
(dec->channel_mask != WavpackGetChannelMask (dec->context));
#endif
if (!GST_PAD_CAPS (dec->srcpad) || format_changed) {
GstCaps *caps;
if (!GST_PAD_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 = WavpackGetBitsPerSample (dec->context);
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, 32,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
dec->depth = WavpackGetBytesPerSample (dec->context) * 8;
#ifdef WAVPACK_OLD_API
channel_mask = dec->context->config.channel_mask;
@ -336,17 +366,7 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
dec->channel_mask = channel_mask;
/* Only set the channel layout for more than two channels
* otherwise things break unfortunately */
if (channel_mask != 0 && dec->channels > 2)
if (!gst_wavpack_set_channel_layout (caps, channel_mask))
GST_WARNING_OBJECT (dec, "Failed to set channel layout");
GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
/* should always succeed */
gst_pad_set_caps (dec->srcpad, caps);
gst_caps_unref (caps);
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 */
@ -354,35 +374,56 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
}
/* alloc output buffer */
unpacked_size = 4 * wph.block_samples * dec->channels;
ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf),
unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf);
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;
gst_buffer_copy_metadata (outbuf, buf, GST_BUFFER_COPY_TIMESTAMPS);
/* If we got a DISCONT buffer forward the flag. Nothing else
* has to be done as libwavpack doesn't store state between
* Wavpack blocks */
if (GST_BUFFER_IS_DISCONT (buf) || dec->next_block_index != wph.block_index)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
dec->next_block_index = wph.block_index + wph.block_samples;
dec_data = g_malloc (4 * wph.block_samples * dec->channels);
out_data = GST_BUFFER_DATA (outbuf);
/* decode */
decoded = WavpackUnpackSamples (dec->context,
(int32_t *) GST_BUFFER_DATA (outbuf), wph.block_samples);
decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
if ((outbuf = gst_audio_buffer_clip (outbuf, &dec->segment,
dec->sample_rate, 4 * dec->channels))) {
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);
width = dec->width;
depth = dec->depth;
max = dec->channels * wph.block_samples;
if (width == 8) {
gint8 *outbuffer = (gint8 *) out_data;
for (i = 0; i < max; i--) {
*outbuffer++ = (gint8) (dec_data[i]);
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) out_data;
for (i = 0; i < max; i++) {
*outbuffer++ = (gint8) (dec_data[i]);
}
} else if (dec->width == 32) {
gint32 *outbuffer = (gint32 *) out_data;
if (width != depth) {
for (i = 0; i < max; i++) {
*outbuffer++ = (gint32) (dec_data[i] << (width - depth));
}
} else {
for (i = 0; i < max; i++) {
*outbuffer++ = (gint32) dec_data[i];
}
}
} else {
g_assert_not_reached ();
}
g_free (dec_data);
ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
out:
@ -390,23 +431,25 @@ out:
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
}
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;
}
context_failed:
{
GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL),
("error creating Wavpack context"), ret);
goto out;
}
decode_error:
{
const gchar *reason = "unknown";
@ -420,86 +463,15 @@ decode_error:
} else {
reason = "couldn't create decoder context";
}
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Failed to decode wavpack stream: %s", reason));
GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
("decoding error: %s", reason), ret);
g_free (dec_data);
if (outbuf)
gst_buffer_unref (outbuf);
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_TIME);
}
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;
}
if (ret == GST_FLOW_OK)
gst_audio_decoder_finish_frame (bdec, NULL, 1);
return ret;
}
}
gboolean

13
ext/wavpack/gstwavpackdec.h
View file

@ -24,6 +24,7 @@
#define __GST_WAVPACK_DEC_H__
#include <gst/gst.h>
#include <gst/audio/gstaudiodecoder.h>
#include <wavpack/wavpack.h>
@ -45,31 +46,25 @@ typedef struct _GstWavpackDecClass GstWavpackDecClass;
struct _GstWavpackDec
{
GstElement element;
GstAudioDecoder element;
/*< private > */
GstPad *sinkpad;
GstPad *srcpad;
WavpackContext *context;
WavpackStreamReader *stream_reader;
read_id wv_id;
GstSegment segment; /* used for clipping, TIME format */
guint32 next_block_index;
gint sample_rate;
gint depth;
gint width;
gint channels;
gint channel_mask;
gint error_count;
};
struct _GstWavpackDecClass
{
GstElementClass parent;
GstAudioDecoderClass parent;
};
GType gst_wavpack_dec_get_type (void);

409
ext/wavpack/gstwavpackenc.c
View file

@ -55,12 +55,18 @@
#include "gstwavpackenc.h"
#include "gstwavpackcommon.h"
static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps);
static gboolean gst_wavpack_enc_start (GstAudioEncoder * enc);
static gboolean gst_wavpack_enc_stop (GstAudioEncoder * enc);
static gboolean gst_wavpack_enc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_wavpack_enc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
static gboolean gst_wavpack_enc_sink_event (GstAudioEncoder * enc,
GstEvent * event);
static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count);
static gboolean gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event);
static GstStateChangeReturn gst_wavpack_enc_change_state (GstElement * element,
GstStateChange transition);
static GstFlowReturn gst_wavpack_enc_drain (GstWavpackEnc * enc);
static void gst_wavpack_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_wavpack_enc_get_property (GObject * object, guint prop_id,
@ -86,7 +92,7 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) [ 1, 32], "
"depth = (int) { 24, 32 }, "
"endianness = (int) BYTE_ORDER, "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE")
@ -97,7 +103,7 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) [ 1, 32 ], "
"channels = (int) [ 1, 2 ], "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
);
@ -196,21 +202,8 @@ gst_wavpack_enc_joint_stereo_mode_get_type (void)
return qtype;
}
static void
_do_init (GType object_type)
{
const GInterfaceInfo preset_interface_info = {
NULL, /* interface_init */
NULL, /* interface_finalize */
NULL /* interface_data */
};
g_type_add_interface_static (object_type, GST_TYPE_PRESET,
&preset_interface_info);
}
GST_BOILERPLATE_FULL (GstWavpackEnc, gst_wavpack_enc, GstElement,
GST_TYPE_ELEMENT, _do_init);
GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstAudioEncoder,
GST_TYPE_AUDIO_ENCODER);
static void
gst_wavpack_enc_base_init (gpointer klass)
@ -232,23 +225,24 @@ gst_wavpack_enc_base_init (gpointer klass)
"Sebastian Dröge <slomo@circular-chaos.org>");
}
static void
gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass);
parent_class = g_type_class_peek_parent (klass);
/* set state change handler */
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_wavpack_enc_change_state);
/* set property handlers */
gobject_class->set_property = gst_wavpack_enc_set_property;
gobject_class->get_property = gst_wavpack_enc_get_property;
base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_enc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_enc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_enc_handle_frame);
base_class->event = GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event);
/* install all properties */
g_object_class_install_property (gobject_class, ARG_MODE,
g_param_spec_enum ("mode", "Encoding mode",
@ -306,6 +300,9 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc)
g_checksum_free (enc->md5_context);
enc->md5_context = NULL;
}
if (enc->pending_segment)
gst_event_unref (enc->pending_segment);
enc->pending_segment = NULL;
if (enc->pending_buffer) {
gst_buffer_unref (enc->pending_buffer);
@ -332,18 +329,7 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc)
static void
gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
{
enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
gst_pad_set_setcaps_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps));
gst_pad_set_chain_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain));
gst_pad_set_event_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event));
gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
/* setup src pad */
enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
/* initialize object attributes */
enc->wp_config = NULL;
@ -367,37 +353,51 @@ gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
enc->md5 = FALSE;
enc->extra_processing = 0;
enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO;
/* require perfect ts */
gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
}
static gboolean
gst_wavpack_enc_start (GstAudioEncoder * enc)
{
GST_DEBUG_OBJECT (enc, "start");
return TRUE;
}
static gboolean
gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
gst_wavpack_enc_stop (GstAudioEncoder * enc)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
GstWavpackEnc *wpenc = GST_WAVPACK_ENC (enc);
GST_DEBUG_OBJECT (enc, "stop");
gst_wavpack_enc_reset (wpenc);
return TRUE;
}
static gboolean
gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
GstAudioChannelPosition *pos;
GstCaps *caps;
if (!gst_structure_get_int (structure, "channels", &enc->channels) ||
!gst_structure_get_int (structure, "rate", &enc->samplerate) ||
!gst_structure_get_int (structure, "depth", &enc->depth)) {
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("got invalid caps: %" GST_PTR_FORMAT, caps));
gst_object_unref (enc);
return FALSE;
}
/* we may be configured again, but that change should have cleanup context */
g_assert (enc->wp_context == NULL);
enc->channels = GST_AUDIO_INFO_CHANNELS (info);
enc->depth = GST_AUDIO_INFO_DEPTH (info);
enc->samplerate = GST_AUDIO_INFO_RATE (info);
pos = info->position;
g_assert (pos);
pos = gst_audio_get_channel_positions (structure);
/* If one channel is NONE they'll be all undefined */
if (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
g_free (pos);
pos = NULL;
}
if (pos == NULL) {
GST_ELEMENT_ERROR (enc, STREAM, FORMAT, (NULL),
("input has no valid channel layout"));
gst_object_unref (enc);
return FALSE;
goto invalid_channels;
}
enc->channel_mask =
@ -405,7 +405,6 @@ gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
enc->need_channel_remap =
gst_wavpack_set_channel_mapping (pos, enc->channels,
enc->channel_mapping);
g_free (pos);
/* set fixed src pad caps now that we know what we will get */
caps = gst_caps_new_simple ("audio/x-wavpack",
@ -416,18 +415,28 @@ gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
if (!gst_wavpack_set_channel_layout (caps, enc->channel_mask))
GST_WARNING_OBJECT (enc, "setting channel layout failed");
if (!gst_pad_set_caps (enc->srcpad, caps)) {
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("setting caps failed: %" GST_PTR_FORMAT, caps));
gst_caps_unref (caps);
gst_object_unref (enc);
return FALSE;
}
gst_pad_use_fixed_caps (enc->srcpad);
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps))
goto setting_src_caps_failed;
gst_caps_unref (caps);
gst_object_unref (enc);
/* no special feedback to base class; should provide all available samples */
return TRUE;
/* ERRORS */
setting_src_caps_failed:
{
GST_DEBUG_OBJECT (enc,
"Couldn't set caps on source pad: %" GST_PTR_FORMAT, caps);
gst_caps_unref (caps);
return FALSE;
}
invalid_channels:
{
GST_DEBUG_OBJECT (enc, "input has invalid channel layout");
return FALSE;
}
}
static void
@ -549,21 +558,14 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
GstBuffer *buffer;
GstPad *pad;
guchar *block = (guchar *) data;
gint samples = 0;
pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad;
pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc);
flow =
(wid->correction) ? &enc->wvcsrcpad_last_return : &enc->
srcpad_last_return;
*flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE,
count, GST_PAD_CAPS (pad), &buffer);
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
return FALSE;
}
buffer = gst_buffer_new_and_alloc (count);
g_memmove (GST_BUFFER_DATA (buffer), block, count);
if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
@ -599,12 +601,14 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
enc->pending_buffer = NULL;
enc->pending_offset = 0;
/* if it's the first wavpack block, send a NEW_SEGMENT event */
if (wph.block_index == 0) {
gst_pad_push_event (pad,
gst_event_new_new_segment (FALSE,
1.0, GST_FORMAT_TIME, 0, GST_BUFFER_OFFSET_NONE, 0));
/* only send segment on correction pad,
* regular pad is handled normally by baseclass */
if (wid->correction && enc->pending_segment) {
gst_pad_push_event (pad, enc->pending_segment);
enc->pending_segment = NULL;
}
if (wph.block_index == 0) {
/* save header for later reference, so we can re-send it later on
* EOS with fixed up values for total sample count etc. */
if (enc->first_block == NULL && !wid->correction) {
@ -614,29 +618,23 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
}
}
}
/* set buffer timestamp, duration, offset, offset_end from
* the wavpack header */
GST_BUFFER_TIMESTAMP (buffer) = enc->timestamp_offset +
gst_util_uint64_scale_int (GST_SECOND, wph.block_index,
enc->samplerate);
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale_int (GST_SECOND, wph.block_samples,
enc->samplerate);
GST_BUFFER_OFFSET (buffer) = wph.block_index;
GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples;
samples = wph.block_samples;
} else {
/* if it's something else set no timestamp and duration on the buffer */
GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count);
GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE;
}
if (wid->correction || wid->passthrough) {
/* push the buffer and forward errors */
GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes",
GST_BUFFER_SIZE (buffer));
*flow = gst_pad_push (pad, buffer);
} else {
GST_DEBUG_OBJECT (enc, "handing frame of %d bytes",
GST_BUFFER_SIZE (buffer));
*flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer,
samples);
}
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
@ -666,18 +664,25 @@ gst_wavpack_enc_fix_channel_order (GstWavpackEnc * enc, gint32 * data,
}
static GstFlowReturn
gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
uint32_t sample_count = GST_BUFFER_SIZE (buf) / 4;
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
uint32_t sample_count;
GstFlowReturn ret;
/* base class ensures configuration */
g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED);
/* reset the last returns to GST_FLOW_OK. This is only set to something else
* while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
* so not valid anymore */
enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
GST_DEBUG ("got %u raw samples", sample_count);
if (G_UNLIKELY (!buf))
return gst_wavpack_enc_drain (enc);
sample_count = GST_BUFFER_SIZE (buf) / 4;
GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count);
/* check if we already have a valid WavpackContext, otherwise make one */
if (!enc->wp_context) {
@ -685,13 +690,8 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
enc->wp_context =
WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id,
(enc->correction_mode > 0) ? &enc->wvc_id : NULL);
if (!enc->wp_context) {
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("error creating Wavpack context"));
gst_object_unref (enc);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
if (!enc->wp_context)
goto context_failed;
/* set the WavpackConfig according to our parameters */
gst_wavpack_enc_set_wp_config (enc);
@ -701,76 +701,12 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
if (!WavpackSetConfiguration (enc->wp_context,
enc->wp_config, (uint32_t) (-1))
|| !WavpackPackInit (enc->wp_context)) {
GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
("error setting up wavpack encoding context"));
WavpackCloseFile (enc->wp_context);
gst_object_unref (enc);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
goto config_failed;
}
GST_DEBUG ("setup of encoding context successfull");
GST_DEBUG_OBJECT (enc, "setup of encoding context successfull");
}
/* Save the timestamp of the first buffer. This will be later
* used as offset for all following buffers */
if (enc->timestamp_offset == GST_CLOCK_TIME_NONE) {
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
enc->timestamp_offset = GST_BUFFER_TIMESTAMP (buf);
enc->next_ts = GST_BUFFER_TIMESTAMP (buf);
} else {
enc->timestamp_offset = 0;
enc->next_ts = 0;
}
}
/* Check if we have a continous stream, if not drop some samples or the buffer or
* insert some silence samples */
if (enc->next_ts != GST_CLOCK_TIME_NONE &&
GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) {
guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf);
guint64 diff_bytes;
GST_WARNING_OBJECT (enc, "Buffer is older than previous "
"timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT
"), cannot handle. Clipping buffer.",
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (enc->next_ts));
diff_bytes =
GST_CLOCK_TIME_TO_FRAMES (diff, enc->samplerate) * enc->channels * 2;
if (diff_bytes >= GST_BUFFER_SIZE (buf)) {
gst_buffer_unref (buf);
return GST_FLOW_OK;
}
buf = gst_buffer_make_metadata_writable (buf);
GST_BUFFER_DATA (buf) += diff_bytes;
GST_BUFFER_SIZE (buf) -= diff_bytes;
GST_BUFFER_TIMESTAMP (buf) += diff;
if (GST_BUFFER_DURATION_IS_VALID (buf))
GST_BUFFER_DURATION (buf) -= diff;
}
/* Allow a diff of at most 5 ms */
if (enc->next_ts != GST_CLOCK_TIME_NONE
&& GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts &&
GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > 5 * GST_MSECOND) {
GST_WARNING_OBJECT (enc,
"Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT,
GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, 5 * GST_MSECOND);
WavpackFlushSamples (enc->wp_context);
enc->timestamp_offset += (GST_BUFFER_TIMESTAMP (buf) - enc->next_ts);
}
}
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)
&& GST_BUFFER_DURATION_IS_VALID (buf))
enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
else
enc->next_ts = GST_CLOCK_TIME_NONE;
if (enc->need_channel_remap) {
buf = gst_buffer_make_writable (buf);
gst_wavpack_enc_fix_channel_order (enc, (gint32 *) GST_BUFFER_DATA (buf),
@ -787,7 +723,7 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
/* encode and handle return values from encoding */
if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf),
sample_count / enc->channels)) {
GST_DEBUG ("encoding samples successful");
GST_DEBUG_OBJECT (enc, "encoding samples successful");
ret = GST_FLOW_OK;
} else {
if ((enc->srcpad_last_return == GST_FLOW_RESEND) ||
@ -803,15 +739,35 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
(enc->wvcsrcpad_last_return == GST_FLOW_FLUSHING)) {
ret = GST_FLOW_FLUSHING;
} else {
GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
("encoding samples failed"));
ret = GST_FLOW_ERROR;
goto encoding_failed;
}
}
gst_buffer_unref (buf);
gst_object_unref (enc);
exit:
return ret;
/* ERRORS */
encoding_failed:
{
GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
("encoding samples failed"));
ret = GST_FLOW_ERROR;
goto exit;
}
config_failed:
{
GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
("error setting up wavpack encoding context"));
ret = GST_FLOW_ERROR;
goto exit;
}
context_failed:
{
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("error creating Wavpack context"));
ret = GST_FLOW_ERROR;
goto exit;
}
}
static void
@ -828,7 +784,7 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
WavpackUpdateNumSamples (enc->wp_context, enc->first_block);
/* try to seek to the beginning of the output */
ret = gst_pad_push_event (enc->srcpad, event);
ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), event);
if (ret) {
/* try to rewrite the first block */
GST_DEBUG_OBJECT (enc, "rewriting first block ...");
@ -836,22 +792,22 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
ret = gst_wavpack_enc_push_block (&enc->wv_id,
enc->first_block, enc->first_block_size);
enc->wv_id.passthrough = FALSE;
g_free (enc->first_block);
enc->first_block = NULL;
} else {
GST_WARNING_OBJECT (enc, "rewriting of first block failed. "
"Seeking to first block failed!");
}
}
static gboolean
gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event)
static GstFlowReturn
gst_wavpack_enc_drain (GstWavpackEnc * enc)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
gboolean ret = TRUE;
if (!enc->wp_context)
return GST_FLOW_OK;
GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event));
GST_DEBUG_OBJECT (enc, "draining");
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* Encode all remaining samples and flush them to the src pads */
WavpackFlushSamples (enc->wp_context);
@ -869,9 +825,10 @@ gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event)
gsize digest_len = sizeof (md5_digest);
g_checksum_get_digest (enc->md5_context, md5_digest, &digest_len);
if (digest_len == sizeof (md5_digest))
if (digest_len == sizeof (md5_digest)) {
WavpackStoreMD5Sum (enc->wp_context, md5_digest);
else
WavpackFlushSamples (enc->wp_context);
} else
GST_WARNING_OBJECT (enc, "Calculating MD5 digest failed");
}
@ -885,62 +842,34 @@ gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event)
enc->wp_context = NULL;
}
ret = gst_pad_event_default (pad, event);
break;
return GST_FLOW_OK;
}
static gboolean
gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
GST_DEBUG_OBJECT (enc, "Received %s event on sinkpad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
if (enc->wp_context) {
GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
"already started");
}
/* drop NEWSEGMENT events, we create our own when pushing
* the first buffer to the pads */
gst_event_unref (event);
ret = TRUE;
break;
default:
ret = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (enc);
return ret;
}
static GstStateChangeReturn
gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstWavpackEnc *enc = GST_WAVPACK_ENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
/* set the last returned GstFlowReturns of the two pads to GST_FLOW_OK
* as they're only set to something else in WavpackPackSamples() or more
* specific gst_wavpack_enc_push_block() and nothing happened there yet */
enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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:
gst_wavpack_enc_reset (enc);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
/* peek and hold NEWSEGMENT events for sending on correction pad */
if (enc->pending_segment)
gst_event_unref (enc->pending_segment);
enc->pending_segment = gst_event_ref (event);
break;
default:
break;
}
return ret;
/* baseclass handles rest */
return FALSE;
}
static void

7
ext/wavpack/gstwavpackenc.h
View file

@ -23,6 +23,7 @@
#define __GST_WAVPACK_ENC_H__
#include <gst/gst.h>
#include <gst/audio/gstaudioencoder.h>
#include <wavpack/wavpack.h>
@ -50,10 +51,9 @@ typedef struct
struct _GstWavpackEnc
{
GstElement element;
GstAudioEncoder element;
/*< private > */
GstPad *sinkpad, *srcpad;
GstPad *wvcsrcpad;
GstFlowReturn srcpad_last_return;
@ -86,6 +86,7 @@ struct _GstWavpackEnc
GstBuffer *pending_buffer;
gint32 pending_offset;
GstEvent *pending_segment;
GstClockTime timestamp_offset;
GstClockTime next_ts;
@ -93,7 +94,7 @@ struct _GstWavpackEnc
struct _GstWavpackEncClass
{
GstElementClass parent;
GstAudioEncoderClass parent;
};
GType gst_wavpack_enc_get_type (void);

4
gst/audiofx/audiofirfilter.c
View file

@ -45,6 +45,10 @@
* </refsect2>
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

4
gst/audiofx/audioiirfilter.c
View file

@ -41,6 +41,10 @@
* </refsect2>
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

4
gst/audioparsers/Makefile.am
View file

@ -3,7 +3,7 @@ plugin_LTLIBRARIES = libgstaudioparsers.la
libgstaudioparsers_la_SOURCES = \
gstaacparse.c gstamrparse.c gstac3parse.c \
gstdcaparse.c gstflacparse.c gstmpegaudioparse.c \
plugin.c
gstwavpackparse.c plugin.c
libgstaudioparsers_la_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
@ -15,4 +15,4 @@ libgstaudioparsers_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstaudioparsers_la_LIBTOOLFLAGS = --tag=disable-static
noinst_HEADERS = gstaacparse.h gstamrparse.h gstac3parse.h \
gstdcaparse.h gstflacparse.h gstmpegaudioparse.h
gstdcaparse.h gstflacparse.h gstmpegaudioparse.h gstwavpackparse.h

648
gst/audioparsers/gstwavpackparse.c Normal file
View file

@ -0,0 +1,648 @@
/* GStreamer Wavpack parser
* Copyright (C) 2012 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* Copyright (C) 2012 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* 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-wavpackparse
* @short_description: Wavpack parser
* @see_also: #GstAmrParse, #GstAACParse
*
* This is an Wavpack parser.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch filesrc location=abc.wavpack ! wavpackparse ! wavpackdec ! audioresample ! audioconvert ! autoaudiosink
* ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstwavpackparse.h"
#include <gst/base/gstbytereader.h>
#include <gst/audio/multichannel.h>
GST_DEBUG_CATEGORY_STATIC (wavpack_parse_debug);
#define GST_CAT_DEFAULT wavpack_parse_debug
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) [ 1, 32 ], "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE; "
"audio/x-wavpack-correction, " "framed = (boolean) TRUE")
);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack"));
static void gst_wavpack_parse_finalize (GObject * object);
static gboolean gst_wavpack_parse_start (GstBaseParse * parse);
static gboolean gst_wavpack_parse_stop (GstBaseParse * parse);
static gboolean gst_wavpack_parse_check_valid_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, guint * size, gint * skipsize);
static GstFlowReturn gst_wavpack_parse_parse_frame (GstBaseParse * parse,
GstBaseParseFrame * frame);
static GstCaps *gst_wavpack_parse_get_sink_caps (GstBaseParse * parse);
/* FIXME remove when all properly renamed */
typedef GstWavpackParse GstWavpackParse2;
typedef GstWavpackParseClass GstWavpackParse2Class;
GST_BOILERPLATE (GstWavpackParse2, gst_wavpack_parse, GstBaseParse,
GST_TYPE_BASE_PARSE);
static void
gst_wavpack_parse_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_static_pad_template (element_class, &sink_template);
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_set_details_simple (element_class,
"Wavpack audio stream parser", "Codec/Parser/Audio",
"Wavpack parser", "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
}
static void
gst_wavpack_parse_class_init (GstWavpackParseClass * klass)
{
GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (wavpack_parse_debug, "wavpackparse", 0,
"Wavpack audio stream parser");
object_class->finalize = gst_wavpack_parse_finalize;
parse_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_parse_start);
parse_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_parse_stop);
parse_class->check_valid_frame =
GST_DEBUG_FUNCPTR (gst_wavpack_parse_check_valid_frame);
parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_wavpack_parse_parse_frame);
parse_class->get_sink_caps =
GST_DEBUG_FUNCPTR (gst_wavpack_parse_get_sink_caps);
}
static void
gst_wavpack_parse_reset (GstWavpackParse * wvparse)
{
wvparse->channels = -1;
wvparse->channel_mask = 0;
wvparse->sample_rate = -1;
wvparse->width = -1;
wvparse->total_samples = 0;
}
static void
gst_wavpack_parse_init (GstWavpackParse * wvparse, GstWavpackParseClass * klass)
{
gst_wavpack_parse_reset (wvparse);
}
static void
gst_wavpack_parse_finalize (GObject * object)
{
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_wavpack_parse_start (GstBaseParse * parse)
{
GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse);
GST_DEBUG_OBJECT (parse, "starting");
gst_wavpack_parse_reset (wvparse);
/* need header at least */
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (wvparse),
sizeof (WavpackHeader));
/* inform baseclass we can come up with ts, based on counters in packets */
gst_base_parse_set_has_timing_info (GST_BASE_PARSE_CAST (wvparse), TRUE);
gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (wvparse), TRUE);
return TRUE;
}
static gboolean
gst_wavpack_parse_stop (GstBaseParse * parse)
{
GST_DEBUG_OBJECT (parse, "stopping");
return TRUE;
}
static gint
gst_wavpack_get_default_channel_mask (gint nchannels)
{
gint channel_mask = 0;
/* Set the default channel mask for the given number of channels.
* It's the same as for WAVE_FORMAT_EXTENDED:
* http://www.microsoft.com/whdc/device/audio/multichaud.mspx
*/
switch (nchannels) {
case 11:
channel_mask |= 0x00400;
channel_mask |= 0x00200;
case 9:
channel_mask |= 0x00100;
case 8:
channel_mask |= 0x00080;
channel_mask |= 0x00040;
case 6:
channel_mask |= 0x00020;
channel_mask |= 0x00010;
case 4:
channel_mask |= 0x00008;
case 3:
channel_mask |= 0x00004;
case 2:
channel_mask |= 0x00002;
channel_mask |= 0x00001;
break;
case 1:
/* For mono use front center */
channel_mask |= 0x00004;
break;
}
return channel_mask;
}
static const struct
{
const guint32 ms_mask;
const GstAudioChannelPosition gst_pos;
} layout_mapping[] = {
{
0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, {
0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
0x00008, GST_AUDIO_CHANNEL_POSITION_LFE}, {
0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, {
0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, {
0x00080, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER}, {
0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, {
0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, {
0x00800, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_CENTER */
{
0x01000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_LEFT */
{
0x02000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_CENTER */
{
0x04000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_RIGHT */
{
0x08000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_LEFT */
{
0x10000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_CENTER */
{
0x20000, GST_AUDIO_CHANNEL_POSITION_INVALID} /* TOP_BACK_RIGHT */
};
#define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping)
static gboolean
gst_wavpack_set_channel_layout (GstCaps * caps, gint layout)
{
GstAudioChannelPosition pos[MAX_CHANNEL_POSITIONS];
GstStructure *s;
gint num_channels, i, p;
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "channels", &num_channels))
g_return_val_if_reached (FALSE);
if (num_channels == 1 && layout == 0x00004) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
gst_audio_set_channel_positions (s, pos);
return TRUE;
}
p = 0;
for (i = 0; i < MAX_CHANNEL_POSITIONS; ++i) {
if ((layout & layout_mapping[i].ms_mask) != 0) {
if (p >= num_channels) {
GST_WARNING ("More bits set in the channel layout map than there "
"are channels! Broken file");
return FALSE;
}
if (layout_mapping[i].gst_pos == GST_AUDIO_CHANNEL_POSITION_INVALID) {
GST_WARNING ("Unsupported channel position (mask 0x%08x) in channel "
"layout map - ignoring those channels", layout_mapping[i].ms_mask);
/* what to do? just ignore it and let downstream deal with a channel
* layout that has INVALID positions in it for now ... */
}
pos[p] = layout_mapping[i].gst_pos;
++p;
}
}
if (p != num_channels) {
GST_WARNING ("Only %d bits set in the channel layout map, but there are "
"supposed to be %d channels! Broken file", p, num_channels);
return FALSE;
}
gst_audio_set_channel_positions (s, pos);
return TRUE;
}
static const guint32 sample_rates[] = {
6000, 8000, 9600, 11025, 12000, 16000, 22050,
24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000
};
#define CHECK(call) { \
if (!call) \
goto read_failed; \
}
/* caller ensures properly sync'ed with enough data */
static gboolean
gst_wavpack_parse_frame_metadata (GstWavpackParse * parse, GstBuffer * buf,
gint skip, WavpackHeader * wph, WavpackInfo * wpi)
{
GstByteReader br;
gint i;
g_return_val_if_fail (wph != NULL || wpi != NULL, FALSE);
g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= skip + sizeof (WavpackHeader),
FALSE);
gst_byte_reader_init (&br, GST_BUFFER_DATA (buf) + skip, wph->ckSize + 8);
/* skip past header */
gst_byte_reader_skip_unchecked (&br, sizeof (WavpackHeader));
/* get some basics from header */
i = (wph->flags >> 23) & 0xF;
if (!wpi->rate)
wpi->rate = (i < G_N_ELEMENTS (sample_rates)) ? sample_rates[i] : 44100;
wpi->width = ((wph->flags & 0x3) + 1) * 8;
if (!wpi->channels)
wpi->channels = (wph->flags & 0x4) ? 1 : 2;
if (!wpi->channel_mask)
wpi->channel_mask = 5 - wpi->channels;
/* need to dig metadata blocks for some more */
while (gst_byte_reader_get_remaining (&br)) {
gint size = 0;
guint16 size2 = 0;
guint8 c, id;
const guint8 *data;
GstByteReader mbr;
CHECK (gst_byte_reader_get_uint8 (&br, &id));
CHECK (gst_byte_reader_get_uint8 (&br, &c));
if (id & ID_LARGE)
CHECK (gst_byte_reader_get_uint16_le (&br, &size2));
size = size2;
size <<= 8;
size += c;
size <<= 1;
if (id & ID_ODD_SIZE)
size--;
CHECK (gst_byte_reader_get_data (&br, size + (size & 1), &data));
gst_byte_reader_init (&mbr, data, size);
switch (id) {
case ID_WVC_BITSTREAM:
GST_LOG_OBJECT (parse, "correction bitstream");
wpi->correction = TRUE;
break;
case ID_WV_BITSTREAM:
case ID_WVX_BITSTREAM:
break;
case ID_SAMPLE_RATE:
if (size == 3) {
CHECK (gst_byte_reader_get_uint24_le (&mbr, &wpi->rate));
GST_LOG_OBJECT (parse, "updated with custom rate %d", wpi->rate);
} else {
GST_DEBUG_OBJECT (parse, "unexpected size for SAMPLE_RATE metadata");
}
break;
case ID_CHANNEL_INFO:
{
guint16 channels;
guint32 mask = 0;
if (size == 6) {
CHECK (gst_byte_reader_get_uint16_le (&mbr, &channels));
channels = channels & 0xFFF;
CHECK (gst_byte_reader_get_uint24_le (&mbr, &mask));
} else if (size) {
CHECK (gst_byte_reader_get_uint8 (&mbr, &c));
channels = c;
while (gst_byte_reader_get_uint8 (&mbr, &c))
mask |= (((guint32) c) << 8);
} else {
GST_DEBUG_OBJECT (parse, "unexpected size for CHANNEL_INFO metadata");
break;
}
wpi->channels = channels;
wpi->channel_mask = mask;
break;
}
default:
GST_LOG_OBJECT (parse, "unparsed ID 0x%x", id);
break;
}
}
return TRUE;
/* ERRORS */
read_failed:
{
GST_DEBUG_OBJECT (parse, "short read while parsing metadata");
/* let's look the other way anyway */
return TRUE;
}
}
/* caller ensures properly sync'ed with enough data */
static gboolean
gst_wavpack_parse_frame_header (GstWavpackParse * parse, GstBuffer * buf,
gint skip, WavpackHeader * _wph)
{
GstByteReader br = GST_BYTE_READER_INIT_FROM_BUFFER (buf);
WavpackHeader wph;
g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= skip + sizeof (WavpackHeader),
FALSE);
/* marker */
gst_byte_reader_skip_unchecked (&br, skip + 4);
/* read */
gst_byte_reader_get_uint32_le (&br, &wph.ckSize);
gst_byte_reader_get_uint16_le (&br, &wph.version);
gst_byte_reader_get_uint8 (&br, &wph.track_no);
gst_byte_reader_get_uint8 (&br, &wph.index_no);
gst_byte_reader_get_uint32_le (&br, &wph.total_samples);
gst_byte_reader_get_uint32_le (&br, &wph.block_index);
gst_byte_reader_get_uint32_le (&br, &wph.block_samples);
gst_byte_reader_get_uint32_le (&br, &wph.flags);
gst_byte_reader_get_uint32_le (&br, &wph.crc);
/* dump */
GST_LOG_OBJECT (parse, "size %d", wph.ckSize);
GST_LOG_OBJECT (parse, "version 0x%x", wph.version);
GST_LOG_OBJECT (parse, "total samples %d", wph.total_samples);
GST_LOG_OBJECT (parse, "block index %d", wph.block_index);
GST_LOG_OBJECT (parse, "block samples %d", wph.block_samples);
GST_LOG_OBJECT (parse, "flags 0x%x", wph.flags);
GST_LOG_OBJECT (parse, "crc 0x%x", wph.flags);
if (!parse->total_samples && wph.block_index == 0 && wph.total_samples != -1) {
GST_DEBUG_OBJECT (parse, "determined duration of %u samples",
wph.total_samples);
parse->total_samples = wph.total_samples;
}
if (_wph)
*_wph = wph;
return TRUE;
}
static gboolean
gst_wavpack_parse_check_valid_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
{
GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse);
GstBuffer *buf = frame->buffer;
GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buf);
gint off;
gboolean lost_sync, draining, final;
guint frmsize = 0;
WavpackHeader wph;
WavpackInfo wpi = { 0, };
if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader)))
return FALSE;
/* scan for 'wvpk' marker */
off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffffffff, 0x7776706b,
0, GST_BUFFER_SIZE (buf));
GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
/* didn't find anything that looks like a sync word, skip */
if (off < 0) {
*skipsize = GST_BUFFER_SIZE (buf) - 3;
goto skip;
}
/* possible frame header, but not at offset 0? skip bytes before sync */
if (off > 0) {
*skipsize = off;
goto skip;
}
/* make sure the values in the frame header look sane */
gst_wavpack_parse_frame_header (wvparse, buf, 0, &wph);
frmsize = wph.ckSize + 8;
/* need the entire frame for parsing */
if (gst_byte_reader_get_remaining (&reader) < frmsize)
goto more;
/* got a frame, now we can dig for some more metadata */
GST_LOG_OBJECT (parse, "got frame");
gst_wavpack_parse_frame_metadata (wvparse, buf, 0, &wph, &wpi);
lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
draining = GST_BASE_PARSE_DRAINING (parse);
while (!(final = (wph.flags & FLAG_FINAL_BLOCK)) || (lost_sync && !draining)) {
guint32 word = 0;
GST_LOG_OBJECT (wvparse, "checking next frame syncword; "
"lost_sync: %d, draining: %d, final: %d", lost_sync, draining, final);
if (!gst_byte_reader_skip (&reader, wph.ckSize + 8) ||
!gst_byte_reader_peek_uint32_be (&reader, &word)) {
GST_DEBUG_OBJECT (wvparse, "... but not sufficient data");
frmsize += 4;
goto more;
} else {
if (word != 0x7776706b) {
GST_DEBUG_OBJECT (wvparse, "0x%x not OK", word);
*skipsize = off + 2;
goto skip;
}
/* need to parse each frame/block for metadata if several ones */
if (!final) {
gint av;
GST_LOG_OBJECT (wvparse, "checking frame at offset %d (0x%x)",
frmsize, frmsize);
av = gst_byte_reader_get_remaining (&reader);
if (av < sizeof (WavpackHeader)) {
frmsize += sizeof (WavpackHeader);
goto more;
}
gst_wavpack_parse_frame_header (wvparse, buf, frmsize, &wph);
off = frmsize;
frmsize += wph.ckSize + 8;
if (av < wph.ckSize + 8)
goto more;
gst_wavpack_parse_frame_metadata (wvparse, buf, off, &wph, &wpi);
/* could also check for matching block_index and block_samples ?? */
}
}
/* resynced if we make it here */
lost_sync = FALSE;
}
/* found frame (up to final), record gathered metadata */
wvparse->wpi = wpi;
wvparse->wph = wph;
*framesize = frmsize;
gst_base_parse_set_min_frame_size (parse, sizeof (WavpackHeader));
return TRUE;
skip:
GST_LOG_OBJECT (wvparse, "skipping %d", *skipsize);
return FALSE;
more:
GST_LOG_OBJECT (wvparse, "need at least %u", frmsize);
gst_base_parse_set_min_frame_size (parse, frmsize);
*skipsize = 0;
return FALSE;
}
static GstFlowReturn
gst_wavpack_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
{
GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse);
GstBuffer *buf = frame->buffer;
guint rate, chans, width, mask;
/* re-use previously parsed data */
rate = wvparse->wpi.rate;
width = wvparse->wpi.width;
chans = wvparse->wpi.channels;
mask = wvparse->wpi.channel_mask;
GST_LOG_OBJECT (parse, "rate: %u, width: %u, chans: %u", rate, width, chans);
GST_BUFFER_TIMESTAMP (buf) =
gst_util_uint64_scale_int (wvparse->wph.block_index, GST_SECOND, rate);
GST_BUFFER_DURATION (buf) =
gst_util_uint64_scale_int (wvparse->wph.block_index +
wvparse->wph.block_samples, GST_SECOND, rate) -
GST_BUFFER_TIMESTAMP (buf);
if (G_UNLIKELY (wvparse->sample_rate != rate || wvparse->channels != chans
|| wvparse->width != width || wvparse->channel_mask != mask)) {
GstCaps *caps;
if (wvparse->wpi.correction) {
caps = gst_caps_new_simple ("audio/x-wavpack-correction",
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
} else {
caps = gst_caps_new_simple ("audio/x-wavpack",
"channels", G_TYPE_INT, chans,
"rate", G_TYPE_INT, rate,
"width", G_TYPE_INT, width, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
if (!mask)
mask = gst_wavpack_get_default_channel_mask (wvparse->channels);
if (mask != 0) {
if (!gst_wavpack_set_channel_layout (caps, mask)) {
GST_WARNING_OBJECT (wvparse, "Failed to set channel layout");
}
}
}
gst_buffer_set_caps (buf, caps);
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
gst_caps_unref (caps);
wvparse->sample_rate = rate;
wvparse->channels = chans;
wvparse->width = width;
wvparse->channel_mask = mask;
if (wvparse->total_samples) {
GST_DEBUG_OBJECT (wvparse, "setting duration");
gst_base_parse_set_duration (GST_BASE_PARSE (wvparse),
GST_FORMAT_TIME, gst_util_uint64_scale_int (wvparse->total_samples,
GST_SECOND, wvparse->sample_rate), 0);
}
}
return GST_FLOW_OK;
}
static GstCaps *
gst_wavpack_parse_get_sink_caps (GstBaseParse * parse)
{
GstCaps *peercaps;
GstCaps *res;
peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (peercaps) {
guint i, n;
/* Remove the framed field */
peercaps = gst_caps_make_writable (peercaps);
n = gst_caps_get_size (peercaps);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (peercaps, i);
gst_structure_remove_field (s, "framed");
}
res =
gst_caps_intersect_full (peercaps,
gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
} else {
res =
gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
(parse)));
}
return res;
}

134
gst/audioparsers/gstwavpackparse.h Normal file
View file

@ -0,0 +1,134 @@
/* GStreamer Wavpack parser
* Copyright (C) 2012 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* Copyright (C) 2012 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* 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.
*/
#ifndef __GST_WAVPACK_PARSE_H__
#define __GST_WAVPACK_PARSE_H__
#include <gst/gst.h>
#include <gst/base/gstbaseparse.h>
G_BEGIN_DECLS
#define GST_TYPE_WAVPACK_PARSE \
(gst_wavpack_parse_get_type())
#define GST_WAVPACK_PARSE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_WAVPACK_PARSE, GstWavpackParse))
#define GST_WAVPACK_PARSE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_WAVPACK_PARSE, GstWavpackParseClass))
#define GST_IS_WAVPACK_PARSE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_WAVPACK_PARSE))
#define GST_IS_WAVPACK_PARSE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_WAVPACK_PARSE))
#define ID_UNIQUE 0x3f
#define ID_OPTIONAL_DATA 0x20
#define ID_ODD_SIZE 0x40
#define ID_LARGE 0x80
#define ID_DUMMY 0x0
#define ID_ENCODER_INFO 0x1
#define ID_DECORR_TERMS 0x2
#define ID_DECORR_WEIGHTS 0x3
#define ID_DECORR_SAMPLES 0x4
#define ID_ENTROPY_VARS 0x5
#define ID_HYBRID_PROFILE 0x6
#define ID_SHAPING_WEIGHTS 0x7
#define ID_FLOAT_INFO 0x8
#define ID_INT32_INFO 0x9
#define ID_WV_BITSTREAM 0xa
#define ID_WVC_BITSTREAM 0xb
#define ID_WVX_BITSTREAM 0xc
#define ID_CHANNEL_INFO 0xd
#define ID_RIFF_HEADER (ID_OPTIONAL_DATA | 0x1)
#define ID_RIFF_TRAILER (ID_OPTIONAL_DATA | 0x2)
#define ID_REPLAY_GAIN (ID_OPTIONAL_DATA | 0x3)
#define ID_CUESHEET (ID_OPTIONAL_DATA | 0x4)
#define ID_CONFIG_BLOCK (ID_OPTIONAL_DATA | 0x5)
#define ID_MD5_CHECKSUM (ID_OPTIONAL_DATA | 0x6)
#define ID_SAMPLE_RATE (ID_OPTIONAL_DATA | 0x7)
#define FLAG_FINAL_BLOCK (1 << 12)
typedef struct {
char ckID [4]; /* "wvpk" */
guint32 ckSize; /* size of entire block (minus 8, of course) */
guint16 version; /* 0x402 to 0x410 are currently valid for decode */
guchar track_no; /* track number (0 if not used, like now) */
guchar index_no; /* track sub-index (0 if not used, like now) */
guint32 total_samples; /* total samples for entire file, but this is
* only valid if block_index == 0 and a value of
* -1 indicates unknown length */
guint32 block_index; /* index of first sample in block relative to
* beginning of file (normally this would start
* at 0 for the first block) */
guint32 block_samples; /* number of samples in this block (0 = no audio) */
guint32 flags; /* various flags for id and decoding */
guint32 crc; /* crc for actual decoded data */
} WavpackHeader;
typedef struct {
gboolean correction;
guint rate;
guint width;
guint channels;
guint channel_mask;
} WavpackInfo;
typedef struct _GstWavpackParse GstWavpackParse;
typedef struct _GstWavpackParseClass GstWavpackParseClass;
/**
* GstWavpackParse:
*
* The opaque GstWavpackParse object
*/
struct _GstWavpackParse {
GstBaseParse baseparse;
/*< private >*/
gint sample_rate;
gint channels;
gint width;
gint channel_mask;
guint total_samples;
WavpackHeader wph;
WavpackInfo wpi;
};
/**
* GstWavpackParseClass:
* @parent_class: Element parent class.
*
* The opaque GstWavpackParseClass data structure.
*/
struct _GstWavpackParseClass {
GstBaseParseClass baseparse_class;
};
GType gst_wavpack_parse_get_type (void);
G_END_DECLS
#endif /* __GST_WAVPACK_PARSE_H__ */

3
gst/audioparsers/plugin.c
View file

@ -27,6 +27,7 @@
#include "gstdcaparse.h"
#include "gstflacparse.h"
#include "gstmpegaudioparse.h"
#include "gstwavpackparse.h"
static gboolean
plugin_init (GstPlugin * plugin)
@ -45,6 +46,8 @@ plugin_init (GstPlugin * plugin)
GST_RANK_PRIMARY + 1, GST_TYPE_FLAC_PARSE);
ret &= gst_element_register (plugin, "mpegaudioparse",
GST_RANK_PRIMARY + 2, GST_TYPE_MPEG_AUDIO_PARSE);
ret &= gst_element_register (plugin, "wavpackparse2",
GST_RANK_SECONDARY, GST_TYPE_WAVPACK_PARSE);
return ret;
}

4
gst/interleave/interleave.c
View file

@ -57,6 +57,10 @@
* </refsect2>
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif

4
gst/rtpmanager/rtpsession.c
View file

@ -17,6 +17,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>

4
gst/udp/gstdynudpsink.c
View file

@ -21,6 +21,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

4
gst/udp/gstmultiudpsink.c
View file

@ -29,6 +29,10 @@
* It can be combined with rtp payload encoders to implement RTP streaming.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

4
sys/oss4/oss4-audio.c
View file

@ -17,6 +17,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

4
sys/oss4/oss4-property-probe.c
View file

@ -17,6 +17,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif

4
sys/v4l2/gstv4l2object.c
View file

@ -18,6 +18,10 @@
* USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif

4
tests/check/elements/interleave.c
View file

@ -18,6 +18,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif

17
tests/check/elements/wavpackdec.c
View file

@ -52,7 +52,7 @@ static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"width = (int) 16, "
"depth = (int) 16, "
"channels = (int) 1, "
"rate = (int) 44100, "
@ -112,23 +112,20 @@ GST_START_TEST (test_decode_frame)
memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame));
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackdec, bus);
/* should decode the buffer without problems */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL);
/* uncompressed data should be 102400 bytes */
fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), 102400);
fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), 51200);
/* and all 102400 bytes must be 0, i.e. silence */
for (i = 0; i < 102400; i++)
/* and all bytes must be 0, i.e. silence */
for (i = 0; i < 51200; i++)
fail_unless_equals_int (GST_BUFFER_DATA (outbuffer)[i], 0);
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
@ -166,14 +163,11 @@ GST_START_TEST (test_decode_frame_with_broken_header)
GST_BUFFER_DATA (inbuffer)[2] = 'e';
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackdec, bus);
/* should fail gracefully */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_ERROR);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
fail_if ((message = gst_bus_pop (bus)) == NULL);
fail_unless_message_error (message, STREAM, DECODE);
@ -204,14 +198,11 @@ GST_START_TEST (test_decode_frame_with_incomplete_frame)
memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame) - 2);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackdec, bus);
/* should fail gracefully */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_ERROR);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
fail_if ((message = gst_bus_pop (bus)) == NULL);
fail_unless_message_error (message, STREAM, DECODE);

13
tests/check/elements/wavpackenc.c
View file

@ -32,14 +32,14 @@ static GstBus *bus;
#define RAW_CAPS_STRING "audio/x-raw-int, " \
"width = (int) 32, " \
"depth = (int) 16, " \
"depth = (int) 32, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"endianness = (int) BYTE_ORDER, " \
"signed = (boolean) true"
#define WAVPACK_CAPS_STRING "audio/x-wavpack, " \
"width = (int) 16, " \
"width = (int) 32, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"framed = (boolean) true"
@ -48,7 +48,7 @@ static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) 16, "
"width = (int) 32, "
"channels = (int) 1, "
"rate = (int) 44100, " "framed = (boolean) true"));
@ -57,7 +57,7 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) 16, "
"depth = (int) 32, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true"));
@ -118,13 +118,10 @@ GST_START_TEST (test_encode_silence)
gst_caps_unref (caps);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackenc, bus);
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
fail_if (gst_pad_push_event (mysrcpad, eos) != TRUE);
@ -134,9 +131,7 @@ GST_START_TEST (test_encode_silence)
fail_if (outbuffer == NULL);
fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 0);
fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 0);
fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 5668934);
fail_unless_equals_int (GST_BUFFER_OFFSET_END (outbuffer), 250);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0,
"Failed to encode to valid Wavpack frames");

4
tests/examples/audiofx/firfilter-example.c
View file

@ -21,6 +21,10 @@
* by transforming the frequency response to the filter kernel.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include <math.h>

4
tests/examples/audiofx/iirfilter-example.c
View file

@ -23,6 +23,10 @@
* of the IIR filter that is used.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include <math.h>

4
tests/examples/pulse/pulse.c
View file

@ -17,6 +17,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <gst/gst.h>
#if 0

4
tests/examples/rtp/server-alsasrc-PCMA.c
View file

@ -17,6 +17,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include <math.h>

4
tests/icles/test-oss4.c
View file

@ -17,6 +17,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif