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voamrwbenc: port to audioencoder

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This commit is contained in:
Mark Nauwelaerts 2011-11-17 23:03:05 +01:00
parent 53723f81eb
commit 97279f1dfd
3 changed files with 88 additions and 162 deletions
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6
ext/voamrwbenc/Makefile.am
View file

@ -4,8 +4,10 @@ libgstvoamrwbenc_la_SOURCES = \
gstvoamrwb.c \ gstvoamrwb.c \
gstvoamrwbenc.c gstvoamrwbenc.c
libgstvoamrwbenc_la_CFLAGS = $(GST_CFLAGS) $(VOAMRWBENC_CFLAGS) libgstvoamrwbenc_la_CFLAGS = -DGST_USE_UNSTABLE_API \
libgstvoamrwbenc_la_LIBADD = $(GST_BASE_LIBS) $(VOAMRWBENC_LIBS) $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(VOAMRWBENC_CFLAGS)
libgstvoamrwbenc_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
-lgstaudio-$(GST_MAJORMINOR) $(GST_BASE_LIBS) $(VOAMRWBENC_LIBS)
libgstvoamrwbenc_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) libgstvoamrwbenc_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstvoamrwbenc_la_LIBTOOLFLAGS = --tag=disable-static libgstvoamrwbenc_la_LIBTOOLFLAGS = --tag=disable-static

230
ext/voamrwbenc/gstvoamrwbenc.c
View file

@ -109,31 +109,15 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_DEBUG_CATEGORY_STATIC (gst_voamrwbenc_debug); GST_DEBUG_CATEGORY_STATIC (gst_voamrwbenc_debug);
#define GST_CAT_DEFAULT gst_voamrwbenc_debug #define GST_CAT_DEFAULT gst_voamrwbenc_debug
static void gst_voamrwbenc_finalize (GObject * object); static gboolean gst_voamrwbenc_start (GstAudioEncoder * enc);
static gboolean gst_voamrwbenc_stop (GstAudioEncoder * enc);
static gboolean gst_voamrwbenc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_voamrwbenc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
static GstFlowReturn gst_voamrwbenc_chain (GstPad * pad, GstBuffer * buffer); GST_BOILERPLATE (GstVoAmrWbEnc, gst_voamrwbenc, GstAudioEncoder,
static gboolean gst_voamrwbenc_setcaps (GstPad * pad, GstCaps * caps); GST_TYPE_AUDIO_ENCODER);
static GstStateChangeReturn gst_voamrwbenc_state_change (GstElement * element,
GstStateChange transition);
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_DEBUG_CATEGORY_INIT (gst_voamrwbenc_debug, "amrwbenc", 0,
"AMR-WB audio encoder");
}
GST_BOILERPLATE_FULL (GstVoAmrWbEnc, gst_voamrwbenc, GstElement,
GST_TYPE_ELEMENT, _do_init);
static void static void
gst_voamrwbenc_set_property (GObject * object, guint prop_id, gst_voamrwbenc_set_property (GObject * object, guint prop_id,
@ -191,71 +175,74 @@ static void
gst_voamrwbenc_class_init (GstVoAmrWbEncClass * klass) gst_voamrwbenc_class_init (GstVoAmrWbEncClass * klass)
{ {
GObjectClass *object_class = G_OBJECT_CLASS (klass); GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
object_class->finalize = gst_voamrwbenc_finalize;
object_class->set_property = gst_voamrwbenc_set_property; object_class->set_property = gst_voamrwbenc_set_property;
object_class->get_property = gst_voamrwbenc_get_property; object_class->get_property = gst_voamrwbenc_get_property;
base_class->start = GST_DEBUG_FUNCPTR (gst_voamrwbenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_voamrwbenc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_voamrwbenc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_voamrwbenc_handle_frame);
g_object_class_install_property (object_class, PROP_BANDMODE, g_object_class_install_property (object_class, PROP_BANDMODE,
g_param_spec_enum ("band-mode", "Band Mode", g_param_spec_enum ("band-mode", "Band Mode",
"Encoding Band Mode (Kbps)", GST_VOAMRWBENC_BANDMODE_TYPE, "Encoding Band Mode (Kbps)", GST_VOAMRWBENC_BANDMODE_TYPE,
BANDMODE_DEFAULT, BANDMODE_DEFAULT,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
element_class->change_state = GST_DEBUG_FUNCPTR (gst_voamrwbenc_state_change);
} }
static void static void
gst_voamrwbenc_init (GstVoAmrWbEnc * amrwbenc, GstVoAmrWbEncClass * klass) gst_voamrwbenc_init (GstVoAmrWbEnc * amrwbenc, GstVoAmrWbEncClass * klass)
{ {
/* create the sink pad */
amrwbenc->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_setcaps_function (amrwbenc->sinkpad, gst_voamrwbenc_setcaps);
gst_pad_set_chain_function (amrwbenc->sinkpad, gst_voamrwbenc_chain);
gst_element_add_pad (GST_ELEMENT (amrwbenc), amrwbenc->sinkpad);
/* create the src pad */
amrwbenc->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_use_fixed_caps (amrwbenc->srcpad);
gst_element_add_pad (GST_ELEMENT (amrwbenc), amrwbenc->srcpad);
amrwbenc->adapter = gst_adapter_new ();
/* init rest */ /* init rest */
amrwbenc->handle = NULL; amrwbenc->handle = NULL;
amrwbenc->channels = 0; amrwbenc->channels = 0;
amrwbenc->rate = 0; amrwbenc->rate = 0;
amrwbenc->ts = 0;
}
static void
gst_voamrwbenc_finalize (GObject * object)
{
GstVoAmrWbEnc *amrwbenc;
amrwbenc = GST_VOAMRWBENC (object);
g_object_unref (G_OBJECT (amrwbenc->adapter));
amrwbenc->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
} }
static gboolean static gboolean
gst_voamrwbenc_setcaps (GstPad * pad, GstCaps * caps) gst_voamrwbenc_start (GstAudioEncoder * enc)
{
GstVoAmrWbEnc *voamrwbenc = GST_VOAMRWBENC (enc);
GST_DEBUG_OBJECT (enc, "start");
if (!(voamrwbenc->handle = E_IF_init ()))
return FALSE;
voamrwbenc->rate = 0;
voamrwbenc->channels = 0;
return TRUE;
}
static gboolean
gst_voamrwbenc_stop (GstAudioEncoder * enc)
{
GstVoAmrWbEnc *voamrwbenc = GST_VOAMRWBENC (enc);
GST_DEBUG_OBJECT (enc, "stop");
if (voamrwbenc->handle) {
E_IF_exit (voamrwbenc->handle);
voamrwbenc->handle = NULL;
}
return TRUE;
}
static gboolean
gst_voamrwbenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{ {
GstStructure *structure;
GstVoAmrWbEnc *amrwbenc; GstVoAmrWbEnc *amrwbenc;
GstCaps *copy; GstCaps *copy;
amrwbenc = GST_VOAMRWBENC (GST_PAD_PARENT (pad)); amrwbenc = GST_VOAMRWBENC (benc);
structure = gst_caps_get_structure (caps, 0);
/* get channel count */ /* get channel count */
gst_structure_get_int (structure, "channels", &amrwbenc->channels); amrwbenc->channels = GST_AUDIO_INFO_CHANNELS (info);
gst_structure_get_int (structure, "rate", &amrwbenc->rate); amrwbenc->rate = GST_AUDIO_INFO_RATE (info);
/* this is not wrong but will sound bad */ /* this is not wrong but will sound bad */
if (amrwbenc->channels != 1) { if (amrwbenc->channels != 1) {
@ -270,116 +257,59 @@ gst_voamrwbenc_setcaps (GstPad * pad, GstCaps * caps)
"channels", G_TYPE_INT, amrwbenc->channels, "channels", G_TYPE_INT, amrwbenc->channels,
"rate", G_TYPE_INT, amrwbenc->rate, NULL); "rate", G_TYPE_INT, amrwbenc->rate, NULL);
gst_pad_set_caps (amrwbenc->srcpad, copy); gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (amrwbenc), copy);
gst_caps_unref (copy); gst_caps_unref (copy);
/* report needs to base class: one frame at a time */
gst_audio_encoder_set_frame_samples_min (benc, L_FRAME16k);
gst_audio_encoder_set_frame_samples_max (benc, L_FRAME16k);
gst_audio_encoder_set_frame_max (benc, 1);
return TRUE; return TRUE;
} }
static GstFlowReturn static GstFlowReturn
gst_voamrwbenc_chain (GstPad * pad, GstBuffer * buffer) gst_voamrwbenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
{ {
GstVoAmrWbEnc *amrwbenc; GstVoAmrWbEnc *amrwbenc;
GstFlowReturn ret = GST_FLOW_OK; GstFlowReturn ret = GST_FLOW_OK;
const int buffer_size = sizeof (short) * L_FRAME16k; const int buffer_size = sizeof (short) * L_FRAME16k;
GstBuffer *out;
gint outsize;
amrwbenc = GST_VOAMRWBENC (gst_pad_get_parent (pad)); amrwbenc = GST_VOAMRWBENC (benc);
g_return_val_if_fail (amrwbenc->handle, GST_FLOW_WRONG_STATE); g_return_val_if_fail (amrwbenc->handle, GST_FLOW_NOT_NEGOTIATED);
if (amrwbenc->rate == 0 || amrwbenc->channels == 0) { if (amrwbenc->rate == 0 || amrwbenc->channels == 0) {
ret = GST_FLOW_NOT_NEGOTIATED; ret = GST_FLOW_NOT_NEGOTIATED;
goto done; goto done;
} }
/* discontinuity clears adapter, FIXME, maybe we can set some /* we don't deal with squeezing remnants, so simply discard those */
* encoder flag to mask the discont. */ if (G_UNLIKELY (buffer == NULL)) {
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) { GST_DEBUG_OBJECT (amrwbenc, "no data");
gst_adapter_clear (amrwbenc->adapter); goto done;
amrwbenc->ts = 0;
amrwbenc->discont = TRUE;
} }
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < buffer_size)) {
amrwbenc->ts = GST_BUFFER_TIMESTAMP (buffer); GST_DEBUG_OBJECT (amrwbenc, "discarding trailing data %d",
buffer ? GST_BUFFER_SIZE (buffer) : 0);
ret = GST_FLOW_OK; ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
gst_adapter_push (amrwbenc->adapter, buffer); goto done;
/* Collect samples until we have enough for an output frame */
while (gst_adapter_available (amrwbenc->adapter) >= buffer_size) {
GstBuffer *out;
guint8 *data;
gint outsize;
out = gst_buffer_new_and_alloc (buffer_size);
GST_BUFFER_DURATION (out) = GST_SECOND * L_FRAME16k /
(amrwbenc->rate * amrwbenc->channels);
GST_BUFFER_TIMESTAMP (out) = amrwbenc->ts;
if (amrwbenc->ts != -1) {
amrwbenc->ts += GST_BUFFER_DURATION (out);
}
if (amrwbenc->discont) {
GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
amrwbenc->discont = FALSE;
}
gst_buffer_set_caps (out, gst_pad_get_caps (amrwbenc->srcpad));
data = (guint8 *) gst_adapter_peek (amrwbenc->adapter, buffer_size);
/* encode */
outsize =
E_IF_encode (amrwbenc->handle, amrwbenc->bandmode, (const short *) data,
(unsigned char *) GST_BUFFER_DATA (out), 0);
gst_adapter_flush (amrwbenc->adapter, buffer_size);
GST_BUFFER_SIZE (out) = outsize;
/* play */
if ((ret = gst_pad_push (amrwbenc->srcpad, out)) != GST_FLOW_OK)
break;
} }
out = gst_buffer_new_and_alloc (buffer_size);
/* encode */
outsize = E_IF_encode (amrwbenc->handle, amrwbenc->bandmode,
(const short *) GST_BUFFER_DATA (buffer),
(unsigned char *) GST_BUFFER_DATA (out), 0);
GST_LOG_OBJECT (amrwbenc, "encoded to %d bytes", outsize);
GST_BUFFER_SIZE (out) = outsize;
ret = gst_audio_encoder_finish_frame (benc, out, L_FRAME16k);
done: done:
gst_object_unref (amrwbenc);
return ret;
}
static GstStateChangeReturn
gst_voamrwbenc_state_change (GstElement * element, GstStateChange transition)
{
GstVoAmrWbEnc *amrwbenc;
GstStateChangeReturn ret;
amrwbenc = GST_VOAMRWBENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!(amrwbenc->handle = E_IF_init ()))
return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
amrwbenc->rate = 0;
amrwbenc->channels = 0;
amrwbenc->ts = 0;
amrwbenc->discont = FALSE;
gst_adapter_clear (amrwbenc->adapter);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
E_IF_exit (amrwbenc->handle);
break;
default:
break;
}
return ret; return ret;
} }

14
ext/voamrwbenc/gstvoamrwbenc.h
View file

@ -21,7 +21,8 @@
#define __GST_VOAMRWBENC_H__ #define __GST_VOAMRWBENC_H__
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/base/gstadapter.h> #include <gst/audio/gstaudioencoder.h>
#include <vo-amrwbenc/enc_if.h> #include <vo-amrwbenc/enc_if.h>
G_BEGIN_DECLS G_BEGIN_DECLS
@ -41,14 +42,7 @@ typedef struct _GstVoAmrWbEnc GstVoAmrWbEnc;
typedef struct _GstVoAmrWbEncClass GstVoAmrWbEncClass; typedef struct _GstVoAmrWbEncClass GstVoAmrWbEncClass;
struct _GstVoAmrWbEnc { struct _GstVoAmrWbEnc {
GstElement element; GstAudioEncoder element;
/* pads */
GstPad *sinkpad, *srcpad;
guint64 ts;
gboolean discont;
GstAdapter *adapter;
/* library handle */ /* library handle */
void *handle; void *handle;
@ -59,7 +53,7 @@ struct _GstVoAmrWbEnc {
}; };
struct _GstVoAmrWbEncClass { struct _GstVoAmrWbEncClass {
GstElementClass parent_class; GstAudioEncoderClass parent_class;
}; };
GType gst_voamrwbenc_get_type (void); GType gst_voamrwbenc_get_type (void);