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gsmdec: port to audiodecoder

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This commit is contained in:
Mark Nauwelaerts 2012-01-17 18:33:09 +01:00
parent 0acfa3cc1e
commit bc1c77395e
2 changed files with 92 additions and 195 deletions
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267
ext/gsm/gstgsmdec.c
View file

@ -43,43 +43,16 @@ enum
ARG_0 ARG_0
}; };
static void gst_gsmdec_base_init (gpointer g_class); static gboolean gst_gsmdec_start (GstAudioDecoder * dec);
static void gst_gsmdec_class_init (GstGSMDec * klass); static gboolean gst_gsmdec_stop (GstAudioDecoder * dec);
static void gst_gsmdec_init (GstGSMDec * gsmdec); static gboolean gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
static void gst_gsmdec_finalize (GObject * object); static GstFlowReturn gst_gsmdec_parse (GstAudioDecoder * dec,
GstAdapter * adapter, gint * offset, gint * length);
static gboolean gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps); static GstFlowReturn gst_gsmdec_handle_frame (GstAudioDecoder * dec,
static gboolean gst_gsmdec_sink_event (GstPad * pad, GstEvent * event); GstBuffer * in_buf);
static GstFlowReturn gst_gsmdec_chain (GstPad * pad, GstBuffer * buf);
static GstElementClass *parent_class = NULL;
/*static guint gst_gsmdec_signals[LAST_SIGNAL] = { 0 }; */ /*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 #define ENCODED_SAMPLES 160
static GstStaticPadTemplate gsmdec_sink_template = static GstStaticPadTemplate gsmdec_sink_template =
@ -101,6 +74,9 @@ GST_STATIC_PAD_TEMPLATE ("src",
"depth = (int) 16, " "rate = (int) [1, MAX], " "channels = (int) 1") "depth = (int) 16, " "rate = (int) [1, MAX], " "channels = (int) 1")
); );
GST_BOILERPLATE (GstGSMDec, gst_gsmdec, GstAudioDecoder,
GST_TYPE_AUDIO_DECODER);
static void static void
gst_gsmdec_base_init (gpointer g_class) gst_gsmdec_base_init (gpointer g_class)
{ {
@ -116,63 +92,60 @@ gst_gsmdec_base_init (gpointer g_class)
} }
static void 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); base_class->start = GST_DEBUG_FUNCPTR (gst_gsmdec_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmdec_stop);
gobject_class->finalize = gst_gsmdec_finalize; 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"); GST_DEBUG_CATEGORY_INIT (gsmdec_debug, "gsmdec", 0, "GSM Decoder");
} }
static void 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 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; GstGSMDec *gsmdec;
GstCaps *srccaps; GstCaps *srccaps;
GstStructure *s; GstStructure *s;
gboolean ret = FALSE; gboolean ret = FALSE;
gint rate;
gsmdec = GST_GSMDEC (gst_pad_get_parent (pad)); gsmdec = GST_GSMDEC (dec);
s = gst_caps_get_structure (caps, 0); s = gst_caps_get_structure (caps, 0);
if (s == NULL) if (s == NULL)
@ -186,7 +159,9 @@ gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
else else
goto wrong_caps; 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"); GST_WARNING_OBJECT (gsmdec, "missing sample rate parameter from sink caps");
goto beach; goto beach;
} }
@ -194,21 +169,16 @@ gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
/* MSGSM needs different framing */ /* MSGSM needs different framing */
gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49); 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. */ /* Setting up src caps based on the input sample rate. */
srccaps = gst_caps_new_simple ("audio/x-raw-int", srccaps = gst_caps_new_simple ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER, "endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, "signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 16, "width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16,
"rate", G_TYPE_INT, gsmdec->rate, "channels", G_TYPE_INT, 1, NULL); "rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, 1, NULL);
ret = gst_pad_set_caps (gsmdec->srcpad, srccaps);
ret = gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), srccaps);
gst_caps_unref (srccaps); gst_caps_unref (srccaps);
gst_object_unref (gsmdec);
return ret; return ret;
@ -218,127 +188,66 @@ wrong_caps:
GST_ERROR_OBJECT (gsmdec, "invalid caps received"); GST_ERROR_OBJECT (gsmdec, "invalid caps received");
beach: beach:
gst_object_unref (gsmdec);
return ret; return ret;
} }
static gboolean static GstFlowReturn
gst_gsmdec_sink_event (GstPad * pad, GstEvent * event) gst_gsmdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
gint * offset, gint * length)
{ {
gboolean res; GstGSMDec *gsmdec = GST_GSMDEC (dec);
GstGSMDec *gsmdec; 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)) { /* WAV49 requires alternating 33 and 32 bytes of input */
case GST_EVENT_FLUSH_START: if (gsmdec->use_wav49) {
res = gst_pad_push_event (gsmdec->srcpad, event); gsmdec->needed = (gsmdec->needed == 33 ? 32 : 33);
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;
} }
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 static GstFlowReturn
gst_gsmdec_chain (GstPad * pad, GstBuffer * buf) gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{ {
GstGSMDec *gsmdec; GstGSMDec *gsmdec;
gsm_byte *data; gsm_byte *data;
GstFlowReturn ret = GST_FLOW_OK; GstFlowReturn ret = GST_FLOW_OK;
GstClockTime timestamp; GstBuffer *outbuf;
gint needed;
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)) { /* always the same amount of output samples */
gst_adapter_clear (gsmdec->adapter); outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
gsmdec->next_ts = GST_CLOCK_TIME_NONE;
/* FIXME, do some good offset */
gsmdec->next_of = 0;
}
gst_adapter_push (gsmdec->adapter, buf);
needed = 33; /* now encode frame into the output buffer */
/* do we have enough bytes to read a frame */ data = (gsm_byte *) GST_BUFFER_DATA (buffer);
while (gst_adapter_available (gsmdec->adapter) >= needed) { if (gsm_decode (gsmdec->state, data,
GstBuffer *outbuf; (gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) {
/* invalid frame */
/* always the same amount of output samples */ GST_AUDIO_DECODER_ERROR (gsmdec, 1, STREAM, DECODE, (NULL),
outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal)); ("tried to decode an invalid frame"), ret);
if (ret != GST_FLOW_OK)
/* If we are not given any timestamp, interpolate from last seen goto exit;
* timestamp (if any). */ gst_buffer_unref (outbuf);
if (timestamp == GST_CLOCK_TIME_NONE) outbuf = NULL;
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);
} }
gst_object_unref (gsmdec); gst_audio_decoder_finish_frame (dec, outbuf, 1);
exit:
return ret; return ret;
} }

20
ext/gsm/gstgsmdec.h
View file

@ -21,7 +21,7 @@
#define __GST_GSMDEC_H__ #define __GST_GSMDEC_H__
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/base/gstadapter.h> #include <gst/audio/gstaudiodecoder.h>
#ifdef GSM_HEADER_IN_SUBDIR #ifdef GSM_HEADER_IN_SUBDIR
#include <gsm/gsm.h> #include <gsm/gsm.h>
@ -47,28 +47,16 @@ typedef struct _GstGSMDecClass GstGSMDecClass;
struct _GstGSMDec struct _GstGSMDec
{ {
GstElement element; GstAudioDecoder element;
/* pads */
GstPad *sinkpad, *srcpad;
gsm state; gsm state;
gint use_wav49; gint use_wav49;
gint64 next_of; gint needed;
GstClockTime next_ts;
GstAdapter *adapter;
GstSegment segment;
gint rate;
GstClockTime duration;
}; };
struct _GstGSMDecClass struct _GstGSMDecClass
{ {
GstElementClass parent_class; GstAudioDecoderClass parent_class;
}; };
GType gst_gsmdec_get_type (void); GType gst_gsmdec_get_type (void);