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gsmenc: port to audioencoder

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This commit is contained in:
Mark Nauwelaerts 2012-01-17 18:32:23 +01:00
parent 6b4038fed3
commit 0acfa3cc1e
2 changed files with 74 additions and 112 deletions
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166
ext/gsm/gstgsmenc.c
View file

@ -43,39 +43,12 @@ enum
ARG_0 ARG_0
}; };
static void gst_gsmenc_base_init (gpointer g_class); static gboolean gst_gsmenc_start (GstAudioEncoder * enc);
static void gst_gsmenc_class_init (GstGSMEnc * klass); static gboolean gst_gsmenc_stop (GstAudioEncoder * enc);
static void gst_gsmenc_init (GstGSMEnc * gsmenc); static gboolean gst_gsmenc_set_format (GstAudioEncoder * enc,
static void gst_gsmenc_finalize (GObject * object); GstAudioInfo * info);
static GstFlowReturn gst_gsmenc_handle_frame (GstAudioEncoder * enc,
static gboolean gst_gsmenc_setcaps (GstPad * pad, GstCaps * caps); GstBuffer * in_buf);
static GstFlowReturn gst_gsmenc_chain (GstPad * pad, GstBuffer * buf);
static GstElementClass *parent_class = NULL;
GType
gst_gsmenc_get_type (void)
{
static GType gsmenc_type = 0;
if (!gsmenc_type) {
static const GTypeInfo gsmenc_info = {
sizeof (GstGSMEncClass),
gst_gsmenc_base_init,
NULL,
(GClassInitFunc) gst_gsmenc_class_init,
NULL,
NULL,
sizeof (GstGSMEnc),
0,
(GInstanceInitFunc) gst_gsmenc_init,
};
gsmenc_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstGSMEnc", &gsmenc_info, 0);
}
return gsmenc_type;
}
static GstStaticPadTemplate gsmenc_src_template = static GstStaticPadTemplate gsmenc_src_template =
GST_STATIC_PAD_TEMPLATE ("src", GST_STATIC_PAD_TEMPLATE ("src",
@ -95,6 +68,9 @@ GST_STATIC_PAD_TEMPLATE ("sink",
"depth = (int) 16, " "rate = (int) 8000, " "channels = (int) 1") "depth = (int) 16, " "rate = (int) 8000, " "channels = (int) 1")
); );
GST_BOILERPLATE (GstGSMEnc, gst_gsmenc, GstAudioEncoder,
GST_TYPE_AUDIO_ENCODER);
static void static void
gst_gsmenc_base_init (gpointer g_class) gst_gsmenc_base_init (gpointer g_class)
{ {
@ -110,34 +86,32 @@ gst_gsmenc_base_init (gpointer g_class)
} }
static void static void
gst_gsmenc_class_init (GstGSMEnc * klass) gst_gsmenc_class_init (GstGSMEncClass * klass)
{ {
GObjectClass *gobject_class; GstAudioEncoderClass *base_class;
gobject_class = (GObjectClass *) klass; base_class = (GstAudioEncoderClass *) klass;
parent_class = g_type_class_peek_parent (klass); base_class->start = GST_DEBUG_FUNCPTR (gst_gsmenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmenc_stop);
gobject_class->finalize = gst_gsmenc_finalize; base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmenc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmenc_handle_frame);
GST_DEBUG_CATEGORY_INIT (gsmenc_debug, "gsmenc", 0, "GSM Encoder"); GST_DEBUG_CATEGORY_INIT (gsmenc_debug, "gsmenc", 0, "GSM Encoder");
} }
static void static void
gst_gsmenc_init (GstGSMEnc * gsmenc) gst_gsmenc_init (GstGSMEnc * gsmenc, GstGSMEncClass * klass)
{ {
}
static gboolean
gst_gsmenc_start (GstAudioEncoder * enc)
{
GstGSMEnc *gsmenc = GST_GSMENC (enc);
gint use_wav49; gint use_wav49;
/* create the sink and src pads */ GST_DEBUG_OBJECT (enc, "start");
gsmenc->sinkpad =
gst_pad_new_from_static_template (&gsmenc_sink_template, "sink");
gst_pad_set_chain_function (gsmenc->sinkpad, gst_gsmenc_chain);
gst_pad_set_setcaps_function (gsmenc->sinkpad, gst_gsmenc_setcaps);
gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->sinkpad);
gsmenc->srcpad =
gst_pad_new_from_static_template (&gsmenc_src_template, "src");
gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->srcpad);
gsmenc->state = gsm_create (); gsmenc->state = gsm_create ();
@ -145,78 +119,72 @@ gst_gsmenc_init (GstGSMEnc * gsmenc)
use_wav49 = 0; use_wav49 = 0;
gsm_option (gsmenc->state, GSM_OPT_WAV49, &use_wav49); gsm_option (gsmenc->state, GSM_OPT_WAV49, &use_wav49);
gsmenc->adapter = gst_adapter_new (); return TRUE;
gsmenc->next_ts = 0;
}
static void
gst_gsmenc_finalize (GObject * object)
{
GstGSMEnc *gsmenc;
gsmenc = GST_GSMENC (object);
g_object_unref (gsmenc->adapter);
gsm_destroy (gsmenc->state);
G_OBJECT_CLASS (parent_class)->finalize (object);
} }
static gboolean static gboolean
gst_gsmenc_setcaps (GstPad * pad, GstCaps * caps) gst_gsmenc_stop (GstAudioEncoder * enc)
{ {
GstGSMEnc *gsmenc; GstGSMEnc *gsmenc = GST_GSMENC (enc);
GstCaps *srccaps;
gsmenc = GST_GSMENC (gst_pad_get_parent (pad)); GST_DEBUG_OBJECT (enc, "stop");
gsm_destroy (gsmenc->state);
srccaps = gst_static_pad_template_get_caps (&gsmenc_src_template);
gst_pad_set_caps (gsmenc->srcpad, srccaps);
gst_object_unref (gsmenc);
return TRUE; return TRUE;
} }
static gboolean
gst_gsmenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstGSMEnc *gsmenc;
GstCaps *srccaps;
gsmenc = GST_GSMENC (benc);
srccaps = gst_static_pad_template_get_caps (&gsmenc_src_template);
gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (benc), srccaps);
/* report needs to base class */
gst_audio_encoder_set_frame_samples_min (benc, 160);
gst_audio_encoder_set_frame_samples_max (benc, 160);
gst_audio_encoder_set_frame_max (benc, 1);
return TRUE;
}
static GstFlowReturn static GstFlowReturn
gst_gsmenc_chain (GstPad * pad, GstBuffer * buf) gst_gsmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
{ {
GstGSMEnc *gsmenc; GstGSMEnc *gsmenc;
gsm_signal *data; gsm_signal *data;
GstFlowReturn ret = GST_FLOW_OK; GstFlowReturn ret = GST_FLOW_OK;
gsmenc = GST_GSMENC (gst_pad_get_parent (pad));
if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (gsmenc->adapter);
}
gst_adapter_push (gsmenc->adapter, buf);
while (gst_adapter_available (gsmenc->adapter) >= 320) {
GstBuffer *outbuf; GstBuffer *outbuf;
gsmenc = GST_GSMENC (benc);
/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buffer == NULL)) {
GST_DEBUG_OBJECT (gsmenc, "no data");
goto done;
}
if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < 320)) {
GST_DEBUG_OBJECT (gsmenc, "discarding trailing data %d",
GST_BUFFER_SIZE (buffer));
ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
goto done;
}
outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte)); outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
GST_BUFFER_TIMESTAMP (outbuf) = gsmenc->next_ts;
GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
gsmenc->next_ts += 20 * GST_MSECOND;
/* encode 160 16-bit samples into 33 bytes */ /* encode 160 16-bit samples into 33 bytes */
data = (gsm_signal *) gst_adapter_peek (gsmenc->adapter, 320); data = (gsm_signal *) GST_BUFFER_DATA (buffer);
gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf)); gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf));
gst_adapter_flush (gsmenc->adapter, 320);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmenc->srcpad)); GST_LOG_OBJECT (gsmenc, "encoded to %d bytes", GST_BUFFER_SIZE (outbuf));
GST_DEBUG_OBJECT (gsmenc, "Pushing buffer of size %d", ret = gst_audio_encoder_finish_frame (benc, outbuf, 160);
GST_BUFFER_SIZE (outbuf));
ret = gst_pad_push (gsmenc->srcpad, outbuf);
}
gst_object_unref (gsmenc);
done:
return ret; return ret;
} }

12
ext/gsm/gstgsmenc.h
View file

@ -21,7 +21,7 @@
#define __GST_GSMENC_H__ #define __GST_GSMENC_H__
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/base/gstadapter.h> #include <gst/audio/gstaudioencoder.h>
#ifdef GSM_HEADER_IN_SUBDIR #ifdef GSM_HEADER_IN_SUBDIR
#include <gsm/gsm.h> #include <gsm/gsm.h>
@ -47,20 +47,14 @@ typedef struct _GstGSMEncClass GstGSMEncClass;
struct _GstGSMEnc struct _GstGSMEnc
{ {
GstElement element; GstAudioEncoder element;
/* pads */
GstPad *sinkpad, *srcpad;
GstAdapter *adapter;
gsm state; gsm state;
GstClockTime next_ts;
gboolean firstBuf;
}; };
struct _GstGSMEncClass struct _GstGSMEncClass
{ {
GstElementClass parent_class; GstAudioEncoderClass parent_class;
}; };
GType gst_gsmenc_get_type (void); GType gst_gsmenc_get_type (void);