mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 15:51:11 +00:00
gsmdec: port to audiodecoder
This commit is contained in:
Mark Nauwelaerts
parent
0acfa3cc1e
commit
bc1c77395e
2 changed files with 92 additions and 195 deletions
|
|
@ -43,43 +43,16 @@ enum
|
|||
ARG_0
|
||||
};
|
||||
|
||||
static void gst_gsmdec_base_init (gpointer g_class);
|
||||
static void gst_gsmdec_class_init (GstGSMDec * klass);
|
||||
static void gst_gsmdec_init (GstGSMDec * gsmdec);
|
||||
static void gst_gsmdec_finalize (GObject * object);
|
||||
|
||||
static gboolean gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps);
|
||||
static gboolean gst_gsmdec_sink_event (GstPad * pad, GstEvent * event);
|
||||
static GstFlowReturn gst_gsmdec_chain (GstPad * pad, GstBuffer * buf);
|
||||
|
||||
static GstElementClass *parent_class = NULL;
|
||||
static gboolean gst_gsmdec_start (GstAudioDecoder * dec);
|
||||
static gboolean gst_gsmdec_stop (GstAudioDecoder * dec);
|
||||
static gboolean gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
|
||||
static GstFlowReturn gst_gsmdec_parse (GstAudioDecoder * dec,
|
||||
GstAdapter * adapter, gint * offset, gint * length);
|
||||
static GstFlowReturn gst_gsmdec_handle_frame (GstAudioDecoder * dec,
|
||||
GstBuffer * in_buf);
|
||||
|
||||
/*static guint gst_gsmdec_signals[LAST_SIGNAL] = { 0 }; */
|
||||
|
||||
GType
|
||||
gst_gsmdec_get_type (void)
|
||||
{
|
||||
static GType gsmdec_type = 0;
|
||||
|
||||
if (!gsmdec_type) {
|
||||
static const GTypeInfo gsmdec_info = {
|
||||
sizeof (GstGSMDecClass),
|
||||
gst_gsmdec_base_init,
|
||||
NULL,
|
||||
(GClassInitFunc) gst_gsmdec_class_init,
|
||||
NULL,
|
||||
NULL,
|
||||
sizeof (GstGSMDec),
|
||||
0,
|
||||
(GInstanceInitFunc) gst_gsmdec_init,
|
||||
};
|
||||
|
||||
gsmdec_type =
|
||||
g_type_register_static (GST_TYPE_ELEMENT, "GstGSMDec", &gsmdec_info, 0);
|
||||
}
|
||||
return gsmdec_type;
|
||||
}
|
||||
|
||||
#define ENCODED_SAMPLES 160
|
||||
|
||||
static GstStaticPadTemplate gsmdec_sink_template =
|
||||
|
|
@ -101,6 +74,9 @@ GST_STATIC_PAD_TEMPLATE ("src",
|
|||
"depth = (int) 16, " "rate = (int) [1, MAX], " "channels = (int) 1")
|
||||
);
|
||||
|
||||
GST_BOILERPLATE (GstGSMDec, gst_gsmdec, GstAudioDecoder,
|
||||
GST_TYPE_AUDIO_DECODER);
|
||||
|
||||
static void
|
||||
gst_gsmdec_base_init (gpointer g_class)
|
||||
{
|
||||
|
|
@ -116,63 +92,60 @@ gst_gsmdec_base_init (gpointer g_class)
|
|||
}
|
||||
|
||||
static void
|
||||
gst_gsmdec_class_init (GstGSMDec * klass)
|
||||
gst_gsmdec_class_init (GstGSMDecClass * klass)
|
||||
{
|
||||
GObjectClass *gobject_class;
|
||||
GstAudioDecoderClass *base_class;
|
||||
|
||||
gobject_class = (GObjectClass *) klass;
|
||||
base_class = (GstAudioDecoderClass *) klass;
|
||||
|
||||
parent_class = g_type_class_peek_parent (klass);
|
||||
|
||||
gobject_class->finalize = gst_gsmdec_finalize;
|
||||
base_class->start = GST_DEBUG_FUNCPTR (gst_gsmdec_start);
|
||||
base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmdec_stop);
|
||||
base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmdec_set_format);
|
||||
base_class->parse = GST_DEBUG_FUNCPTR (gst_gsmdec_parse);
|
||||
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmdec_handle_frame);
|
||||
|
||||
GST_DEBUG_CATEGORY_INIT (gsmdec_debug, "gsmdec", 0, "GSM Decoder");
|
||||
}
|
||||
|
||||
static void
|
||||
gst_gsmdec_init (GstGSMDec * gsmdec)
|
||||
gst_gsmdec_init (GstGSMDec * gsmdec, GstGSMDecClass * klass)
|
||||
{
|
||||
/* create the sink and src pads */
|
||||
gsmdec->sinkpad =
|
||||
gst_pad_new_from_static_template (&gsmdec_sink_template, "sink");
|
||||
gst_pad_set_setcaps_function (gsmdec->sinkpad, gst_gsmdec_sink_setcaps);
|
||||
gst_pad_set_event_function (gsmdec->sinkpad, gst_gsmdec_sink_event);
|
||||
gst_pad_set_chain_function (gsmdec->sinkpad, gst_gsmdec_chain);
|
||||
gst_element_add_pad (GST_ELEMENT (gsmdec), gsmdec->sinkpad);
|
||||
|
||||
gsmdec->srcpad =
|
||||
gst_pad_new_from_static_template (&gsmdec_src_template, "src");
|
||||
gst_element_add_pad (GST_ELEMENT (gsmdec), gsmdec->srcpad);
|
||||
|
||||
gsmdec->state = gsm_create ();
|
||||
|
||||
gsmdec->adapter = gst_adapter_new ();
|
||||
gsmdec->next_of = 0;
|
||||
gsmdec->next_ts = 0;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_gsmdec_finalize (GObject * object)
|
||||
{
|
||||
GstGSMDec *gsmdec;
|
||||
|
||||
gsmdec = GST_GSMDEC (object);
|
||||
|
||||
g_object_unref (gsmdec->adapter);
|
||||
gsm_destroy (gsmdec->state);
|
||||
|
||||
G_OBJECT_CLASS (parent_class)->finalize (object);
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
|
||||
gst_gsmdec_start (GstAudioDecoder * dec)
|
||||
{
|
||||
GstGSMDec *gsmdec = GST_GSMDEC (dec);
|
||||
|
||||
GST_DEBUG_OBJECT (dec, "start");
|
||||
|
||||
gsmdec->state = gsm_create ();
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_gsmdec_stop (GstAudioDecoder * dec)
|
||||
{
|
||||
GstGSMDec *gsmdec = GST_GSMDEC (dec);
|
||||
|
||||
GST_DEBUG_OBJECT (dec, "stop");
|
||||
|
||||
gsm_destroy (gsmdec->state);
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
|
||||
{
|
||||
GstGSMDec *gsmdec;
|
||||
GstCaps *srccaps;
|
||||
GstStructure *s;
|
||||
gboolean ret = FALSE;
|
||||
gint rate;
|
||||
|
||||
gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
|
||||
gsmdec = GST_GSMDEC (dec);
|
||||
|
||||
s = gst_caps_get_structure (caps, 0);
|
||||
if (s == NULL)
|
||||
|
|
@ -186,7 +159,9 @@ gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
|
|||
else
|
||||
goto wrong_caps;
|
||||
|
||||
if (!gst_structure_get_int (s, "rate", &gsmdec->rate)) {
|
||||
gsmdec->needed = 33;
|
||||
|
||||
if (!gst_structure_get_int (s, "rate", &rate)) {
|
||||
GST_WARNING_OBJECT (gsmdec, "missing sample rate parameter from sink caps");
|
||||
goto beach;
|
||||
}
|
||||
|
|
@ -194,21 +169,16 @@ gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
|
|||
/* MSGSM needs different framing */
|
||||
gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49);
|
||||
|
||||
gsmdec->duration = gst_util_uint64_scale (ENCODED_SAMPLES,
|
||||
GST_SECOND, gsmdec->rate);
|
||||
|
||||
/* Setting up src caps based on the input sample rate. */
|
||||
srccaps = gst_caps_new_simple ("audio/x-raw-int",
|
||||
"endianness", G_TYPE_INT, G_BYTE_ORDER,
|
||||
"signed", G_TYPE_BOOLEAN, TRUE,
|
||||
"width", G_TYPE_INT, 16,
|
||||
"depth", G_TYPE_INT, 16,
|
||||
"rate", G_TYPE_INT, gsmdec->rate, "channels", G_TYPE_INT, 1, NULL);
|
||||
|
||||
ret = gst_pad_set_caps (gsmdec->srcpad, srccaps);
|
||||
"rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, 1, NULL);
|
||||
|
||||
ret = gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), srccaps);
|
||||
gst_caps_unref (srccaps);
|
||||
gst_object_unref (gsmdec);
|
||||
|
||||
return ret;
|
||||
|
||||
|
|
@ -218,127 +188,66 @@ wrong_caps:
|
|||
GST_ERROR_OBJECT (gsmdec, "invalid caps received");
|
||||
|
||||
beach:
|
||||
gst_object_unref (gsmdec);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_gsmdec_sink_event (GstPad * pad, GstEvent * event)
|
||||
static GstFlowReturn
|
||||
gst_gsmdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
|
||||
gint * offset, gint * length)
|
||||
{
|
||||
gboolean res;
|
||||
GstGSMDec *gsmdec;
|
||||
GstGSMDec *gsmdec = GST_GSMDEC (dec);
|
||||
guint size;
|
||||
|
||||
gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
|
||||
size = gst_adapter_available (adapter);
|
||||
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
|
||||
|
||||
switch (GST_EVENT_TYPE (event)) {
|
||||
case GST_EVENT_FLUSH_START:
|
||||
res = gst_pad_push_event (gsmdec->srcpad, event);
|
||||
break;
|
||||
case GST_EVENT_FLUSH_STOP:
|
||||
gst_segment_init (&gsmdec->segment, GST_FORMAT_UNDEFINED);
|
||||
res = gst_pad_push_event (gsmdec->srcpad, event);
|
||||
break;
|
||||
case GST_EVENT_NEWSEGMENT:
|
||||
{
|
||||
gboolean update;
|
||||
GstFormat format;
|
||||
gdouble rate, arate;
|
||||
gint64 start, stop, time;
|
||||
|
||||
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
|
||||
&start, &stop, &time);
|
||||
|
||||
/* now configure the values */
|
||||
gst_segment_set_newsegment_full (&gsmdec->segment, update,
|
||||
rate, arate, format, start, stop, time);
|
||||
|
||||
/* and forward */
|
||||
res = gst_pad_push_event (gsmdec->srcpad, event);
|
||||
break;
|
||||
}
|
||||
case GST_EVENT_EOS:
|
||||
default:
|
||||
res = gst_pad_push_event (gsmdec->srcpad, event);
|
||||
break;
|
||||
/* WAV49 requires alternating 33 and 32 bytes of input */
|
||||
if (gsmdec->use_wav49) {
|
||||
gsmdec->needed = (gsmdec->needed == 33 ? 32 : 33);
|
||||
}
|
||||
|
||||
gst_object_unref (gsmdec);
|
||||
if (size < gsmdec->needed)
|
||||
return GST_FLOW_UNEXPECTED;
|
||||
|
||||
return res;
|
||||
*offset = 0;
|
||||
*length = gsmdec->needed;
|
||||
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
gst_gsmdec_chain (GstPad * pad, GstBuffer * buf)
|
||||
gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
|
||||
{
|
||||
GstGSMDec *gsmdec;
|
||||
gsm_byte *data;
|
||||
GstFlowReturn ret = GST_FLOW_OK;
|
||||
GstClockTime timestamp;
|
||||
gint needed;
|
||||
GstBuffer *outbuf;
|
||||
|
||||
gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
|
||||
/* no fancy draining */
|
||||
if (G_UNLIKELY (!buffer))
|
||||
return GST_FLOW_OK;
|
||||
|
||||
timestamp = GST_BUFFER_TIMESTAMP (buf);
|
||||
gsmdec = GST_GSMDEC (dec);
|
||||
|
||||
if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
|
||||
gst_adapter_clear (gsmdec->adapter);
|
||||
gsmdec->next_ts = GST_CLOCK_TIME_NONE;
|
||||
/* FIXME, do some good offset */
|
||||
gsmdec->next_of = 0;
|
||||
}
|
||||
gst_adapter_push (gsmdec->adapter, buf);
|
||||
/* always the same amount of output samples */
|
||||
outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
|
||||
|
||||
needed = 33;
|
||||
/* do we have enough bytes to read a frame */
|
||||
while (gst_adapter_available (gsmdec->adapter) >= needed) {
|
||||
GstBuffer *outbuf;
|
||||
|
||||
/* always the same amount of output samples */
|
||||
outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
|
||||
|
||||
/* If we are not given any timestamp, interpolate from last seen
|
||||
* timestamp (if any). */
|
||||
if (timestamp == GST_CLOCK_TIME_NONE)
|
||||
timestamp = gsmdec->next_ts;
|
||||
|
||||
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
|
||||
|
||||
/* interpolate in the next run */
|
||||
if (timestamp != GST_CLOCK_TIME_NONE)
|
||||
gsmdec->next_ts = timestamp + gsmdec->duration;
|
||||
timestamp = GST_CLOCK_TIME_NONE;
|
||||
|
||||
GST_BUFFER_DURATION (outbuf) = gsmdec->duration;
|
||||
GST_BUFFER_OFFSET (outbuf) = gsmdec->next_of;
|
||||
if (gsmdec->next_of != -1)
|
||||
gsmdec->next_of += ENCODED_SAMPLES;
|
||||
GST_BUFFER_OFFSET_END (outbuf) = gsmdec->next_of;
|
||||
|
||||
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmdec->srcpad));
|
||||
|
||||
/* now encode frame into the output buffer */
|
||||
data = (gsm_byte *) gst_adapter_peek (gsmdec->adapter, needed);
|
||||
if (gsm_decode (gsmdec->state, data,
|
||||
(gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) {
|
||||
/* invalid frame */
|
||||
GST_WARNING_OBJECT (gsmdec, "tried to decode an invalid frame, skipping");
|
||||
}
|
||||
gst_adapter_flush (gsmdec->adapter, needed);
|
||||
|
||||
/* WAV49 requires alternating 33 and 32 bytes of input */
|
||||
if (gsmdec->use_wav49)
|
||||
needed = (needed == 33 ? 32 : 33);
|
||||
|
||||
GST_DEBUG_OBJECT (gsmdec, "Pushing buffer of size %d ts %" GST_TIME_FORMAT,
|
||||
GST_BUFFER_SIZE (outbuf),
|
||||
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
|
||||
|
||||
/* push */
|
||||
ret = gst_pad_push (gsmdec->srcpad, outbuf);
|
||||
/* now encode frame into the output buffer */
|
||||
data = (gsm_byte *) GST_BUFFER_DATA (buffer);
|
||||
if (gsm_decode (gsmdec->state, data,
|
||||
(gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) {
|
||||
/* invalid frame */
|
||||
GST_AUDIO_DECODER_ERROR (gsmdec, 1, STREAM, DECODE, (NULL),
|
||||
("tried to decode an invalid frame"), ret);
|
||||
if (ret != GST_FLOW_OK)
|
||||
goto exit;
|
||||
gst_buffer_unref (outbuf);
|
||||
outbuf = NULL;
|
||||
}
|
||||
|
||||
gst_object_unref (gsmdec);
|
||||
gst_audio_decoder_finish_frame (dec, outbuf, 1);
|
||||
|
||||
exit:
|
||||
return ret;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -21,7 +21,7 @@
|
|||
#define __GST_GSMDEC_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/base/gstadapter.h>
|
||||
#include <gst/audio/gstaudiodecoder.h>
|
||||
|
||||
#ifdef GSM_HEADER_IN_SUBDIR
|
||||
#include <gsm/gsm.h>
|
||||
|
|
@ -47,28 +47,16 @@ typedef struct _GstGSMDecClass GstGSMDecClass;
|
|||
|
||||
struct _GstGSMDec
|
||||
{
|
||||
GstElement element;
|
||||
|
||||
/* pads */
|
||||
GstPad *sinkpad, *srcpad;
|
||||
GstAudioDecoder element;
|
||||
|
||||
gsm state;
|
||||
gint use_wav49;
|
||||
gint64 next_of;
|
||||
GstClockTime next_ts;
|
||||
|
||||
GstAdapter *adapter;
|
||||
|
||||
GstSegment segment;
|
||||
|
||||
gint rate;
|
||||
|
||||
GstClockTime duration;
|
||||
gint needed;
|
||||
};
|
||||
|
||||
struct _GstGSMDecClass
|
||||
{
|
||||
GstElementClass parent_class;
|
||||
GstAudioDecoderClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_gsmdec_get_type (void);
|
||||
|
|
|
|||
Loading…
Reference in a new issue