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amrnbenc: port to audioencoder

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This commit is contained in:
Mark Nauwelaerts 2011-09-27 17:01:59 +02:00
parent 62497d4ba8
commit dc3013f925
3 changed files with 94 additions and 173 deletions
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6
ext/amrnb/Makefile.am
View file

@ -5,8 +5,10 @@ libgstamrnb_la_SOURCES = \
amrnbdec.c \
amrnbenc.c
libgstamrnb_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AMRNB_CFLAGS)
libgstamrnb_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) $(AMRNB_LIBS)
libgstamrnb_la_CFLAGS = -DGST_USE_UNSTABLE_API $(GST_PLUGINS_BASE_CFLAGS) \
$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AMRNB_CFLAGS)
libgstamrnb_la_LIBADD = $(GST_BASE_LIBS) -lgstaudio-@GST_MAJORMINOR@ \
$(GST_LIBS) $(AMRNB_LIBS)
libgstamrnb_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstamrnb_la_LIBTOOLFLAGS = --tag=disable-static

244
ext/amrnb/amrnbenc.c
View file

@ -92,31 +92,15 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_DEBUG_CATEGORY_STATIC (gst_amrnbenc_debug);
#define GST_CAT_DEFAULT gst_amrnbenc_debug
static void gst_amrnbenc_finalize (GObject * object);
static gboolean gst_amrnbenc_start (GstAudioEncoder * enc);
static gboolean gst_amrnbenc_stop (GstAudioEncoder * enc);
static gboolean gst_amrnbenc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_amrnbenc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
static GstFlowReturn gst_amrnbenc_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_amrnbenc_setcaps (GstPad * pad, GstCaps * caps);
static GstStateChangeReturn gst_amrnbenc_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_amrnbenc_debug, "amrnbenc", 0,
"AMR-NB audio encoder");
}
GST_BOILERPLATE_FULL (GstAmrnbEnc, gst_amrnbenc, GstElement, GST_TYPE_ELEMENT,
_do_init);
GST_BOILERPLATE (GstAmrnbEnc, gst_amrnbenc, GstAudioEncoder,
GST_TYPE_AUDIO_ENCODER);
static void
gst_amrnbenc_set_property (GObject * object, guint prop_id,
@ -172,11 +156,15 @@ static void
gst_amrnbenc_class_init (GstAmrnbEncClass * 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->set_property = gst_amrnbenc_set_property;
object_class->get_property = gst_amrnbenc_get_property;
object_class->finalize = gst_amrnbenc_finalize;
base_class->start = GST_DEBUG_FUNCPTR (gst_amrnbenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_amrnbenc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrnbenc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrnbenc_handle_frame);
g_object_class_install_property (object_class, PROP_BANDMODE,
g_param_spec_enum ("band-mode", "Band Mode",
@ -184,57 +172,53 @@ gst_amrnbenc_class_init (GstAmrnbEncClass * klass)
BANDMODE_DEFAULT,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
element_class->change_state = GST_DEBUG_FUNCPTR (gst_amrnbenc_state_change);
GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0,
"AMR-NB audio encoder");
}
static void
gst_amrnbenc_init (GstAmrnbEnc * amrnbenc, GstAmrnbEncClass * klass)
{
/* create the sink pad */
amrnbenc->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_setcaps_function (amrnbenc->sinkpad, gst_amrnbenc_setcaps);
gst_pad_set_chain_function (amrnbenc->sinkpad, gst_amrnbenc_chain);
gst_element_add_pad (GST_ELEMENT (amrnbenc), amrnbenc->sinkpad);
/* create the src pad */
amrnbenc->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_use_fixed_caps (amrnbenc->srcpad);
gst_element_add_pad (GST_ELEMENT (amrnbenc), amrnbenc->srcpad);
amrnbenc->adapter = gst_adapter_new ();
/* init rest */
amrnbenc->handle = NULL;
}
static void
gst_amrnbenc_finalize (GObject * object)
{
GstAmrnbEnc *amrnbenc;
amrnbenc = GST_AMRNBENC (object);
g_object_unref (G_OBJECT (amrnbenc->adapter));
amrnbenc->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_amrnbenc_setcaps (GstPad * pad, GstCaps * caps)
gst_amrnbenc_start (GstAudioEncoder * enc)
{
GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);
GST_DEBUG_OBJECT (amrnbenc, "start");
if (!(amrnbenc->handle = Encoder_Interface_init (0)))
return FALSE;
return TRUE;
}
static gboolean
gst_amrnbenc_stop (GstAudioEncoder * enc)
{
GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);
GST_DEBUG_OBJECT (amrnbenc, "stop");
Encoder_Interface_exit (amrnbenc->handle);
return TRUE;
}
static gboolean
gst_amrnbenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstStructure *structure;
GstAmrnbEnc *amrnbenc;
GstCaps *copy;
amrnbenc = GST_AMRNBENC (GST_PAD_PARENT (pad));
amrnbenc = GST_AMRNBENC (enc);
structure = gst_caps_get_structure (caps, 0);
/* get channel count */
gst_structure_get_int (structure, "channels", &amrnbenc->channels);
gst_structure_get_int (structure, "rate", &amrnbenc->rate);
/* parameters already parsed for us */
amrnbenc->rate = GST_AUDIO_INFO_RATE (info);
amrnbenc->channels = GST_AUDIO_INFO_CHANNELS (info);
/* we do not really accept other input, but anyway ... */
/* this is not wrong but will sound bad */
if (amrnbenc->channels != 1) {
g_warning ("amrnbdec is only optimized for mono channels");
@ -248,124 +232,64 @@ gst_amrnbenc_setcaps (GstPad * pad, GstCaps * caps)
"channels", G_TYPE_INT, amrnbenc->channels,
"rate", G_TYPE_INT, amrnbenc->rate, NULL);
/* precalc duration as it's constant now */
amrnbenc->duration = gst_util_uint64_scale_int (160, GST_SECOND,
amrnbenc->rate * amrnbenc->channels);
gst_pad_set_caps (amrnbenc->srcpad, copy);
gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (amrnbenc), copy);
gst_caps_unref (copy);
/* report needs to base class: hand one frame at a time */
gst_audio_encoder_set_frame_samples_min (enc, 160);
gst_audio_encoder_set_frame_samples_max (enc, 160);
gst_audio_encoder_set_frame_max (enc, 1);
return TRUE;
}
static GstFlowReturn
gst_amrnbenc_chain (GstPad * pad, GstBuffer * buffer)
gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
{
GstAmrnbEnc *amrnbenc;
GstFlowReturn ret;
GstBuffer *out;
guint8 *data;
gint outsize;
amrnbenc = GST_AMRNBENC (GST_PAD_PARENT (pad));
amrnbenc = GST_AMRNBENC (enc);
g_return_val_if_fail (amrnbenc->handle, GST_FLOW_WRONG_STATE);
if (amrnbenc->rate == 0 || amrnbenc->channels == 0)
goto not_negotiated;
/* discontinuity clears adapter, FIXME, maybe we can set some
* encoder flag to mask the discont. */
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (amrnbenc->adapter);
amrnbenc->ts = 0;
amrnbenc->discont = TRUE;
/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buffer == NULL)) {
GST_DEBUG_OBJECT (amrnbenc, "no data");
return GST_FLOW_OK;
}
/* take latest timestamp, FIXME timestamp is the one of the
* first buffer in the adapter. */
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
amrnbenc->ts = GST_BUFFER_TIMESTAMP (buffer);
if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < 320)) {
GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data %d",
buffer ? GST_BUFFER_SIZE (buffer) : 0);
return gst_audio_encoder_finish_frame (enc, NULL, -1);
}
ret = GST_FLOW_OK;
gst_adapter_push (amrnbenc->adapter, buffer);
/* get output, max size is 32 */
out = gst_buffer_new_and_alloc (32);
/* AMR encoder actually writes into the source data buffers it gets */
/* should be able to handle that with what we are given */
data = GST_BUFFER_DATA (buffer);
/* Collect samples until we have enough for an output frame */
while (gst_adapter_available (amrnbenc->adapter) >= 320) {
GstBuffer *out;
guint8 *data;
gint outsize;
/* encode */
outsize =
Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
(short *) data, (guint8 *) GST_BUFFER_DATA (out), 0);
/* get output, max size is 32 */
out = gst_buffer_new_and_alloc (32);
GST_BUFFER_DURATION (out) = amrnbenc->duration;
GST_BUFFER_TIMESTAMP (out) = amrnbenc->ts;
if (amrnbenc->ts != -1) {
amrnbenc->ts += amrnbenc->duration;
}
if (amrnbenc->discont) {
GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
amrnbenc->discont = FALSE;
}
gst_buffer_set_caps (out, GST_PAD_CAPS (amrnbenc->srcpad));
/* The AMR encoder actually writes into the source data buffers it gets */
data = gst_adapter_take (amrnbenc->adapter, 320);
/* encode */
outsize =
Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
(short *) data, (guint8 *) GST_BUFFER_DATA (out), 0);
g_free (data);
GST_LOG_OBJECT (amrnbenc, "output data size %d", outsize);
if (outsize) {
GST_BUFFER_SIZE (out) = outsize;
/* play */
if ((ret = gst_pad_push (amrnbenc->srcpad, out)) != GST_FLOW_OK)
break;
}
return ret;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (amrnbenc, STREAM, TYPE_NOT_FOUND,
(NULL), ("unknown type"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
static GstStateChangeReturn
gst_amrnbenc_state_change (GstElement * element, GstStateChange transition)
{
GstAmrnbEnc *amrnbenc;
GstStateChangeReturn ret;
amrnbenc = GST_AMRNBENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!(amrnbenc->handle = Encoder_Interface_init (0)))
return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
amrnbenc->rate = 0;
amrnbenc->channels = 0;
amrnbenc->ts = 0;
amrnbenc->discont = FALSE;
gst_adapter_clear (amrnbenc->adapter);
break;
default:
break;
ret = gst_audio_encoder_finish_frame (enc, out, 160);
} else {
/* should not happen (without dtx or so at least) */
GST_WARNING_OBJECT (amrnbenc, "no encoded data; discarding input");
gst_buffer_unref (out);
ret = gst_audio_encoder_finish_frame (enc, NULL, -1);
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
Encoder_Interface_exit (amrnbenc->handle);
break;
default:
break;
}
return ret;
}

17
ext/amrnb/amrnbenc.h
View file

@ -22,7 +22,7 @@
#include <gst/gst.h>
#include <interf_enc.h>
#include <gst/base/gstadapter.h>
#include <gst/audio/gstaudioencoder.h>
G_BEGIN_DECLS
@ -41,26 +41,21 @@ typedef struct _GstAmrnbEnc GstAmrnbEnc;
typedef struct _GstAmrnbEncClass GstAmrnbEncClass;
struct _GstAmrnbEnc {
GstElement element;
/* pads */
GstPad *sinkpad, *srcpad;
guint64 ts;
gboolean discont;
GstAdapter *adapter;
GstAudioEncoder element;
/* library handle */
void *handle;
/* input settings */
enum Mode bandmode;
gint channels, rate;
gint duration;
/* property */
enum Mode bandmode;
};
struct _GstAmrnbEncClass {
GstElementClass parent_class;
GstAudioEncoderClass parent_class;
};
GType gst_amrnbenc_get_type (void);