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adpcmenc: port to audioencoder

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This commit is contained in:
Mark Nauwelaerts 2012-01-17 21:36:08 +01:00
parent bc1c77395e
commit 1a73bf0b79
2 changed files with 64 additions and 156 deletions
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5
gst/adpcmenc/Makefile.am
View file

@ -5,8 +5,9 @@ libgstadpcmenc_la_SOURCES = adpcmenc.c
# flags used to compile this plugin
# add other _CFLAGS and _LIBS as needed
libgstadpcmenc_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS)
libgstadpcmenc_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS)
libgstadpcmenc_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
libgstadpcmenc_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-@GST_MAJORMINOR@ \
$(GST_LIBS)
libgstadpcmenc_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstadpcmenc_la_LIBTOOLFLAGS = --tag=disable-static

215
gst/adpcmenc/adpcmenc.c
View file

@ -28,7 +28,7 @@
#endif
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
#include <gst/audio/gstaudioencoder.h>
#define GST_TYPE_ADPCM_ENC \
(adpcmenc_get_type ())
@ -113,17 +113,12 @@ adpcmenc_layout_get_type (void)
typedef struct _ADPCMEncClass
{
GstElementClass parent_class;
GstAudioEncoderClass parent_class;
} ADPCMEncClass;
typedef struct _ADPCMEnc
{
GstElement parent;
GstPad *sinkpad;
GstPad *srcpad;
GstCaps *output_caps;
GstAudioEncoder parent;
enum adpcm_layout layout;
int rate;
@ -133,19 +128,11 @@ typedef struct _ADPCMEnc
guint8 step_index[2];
gboolean is_setup;
GstClockTime timestamp;
GstClockTime base_timestamp;
guint64 out_samples;
GstAdapter *adapter;
} ADPCMEnc;
GType adpcmenc_get_type (void);
GST_BOILERPLATE (ADPCMEnc, adpcmenc, GstElement, GST_TYPE_ELEMENT);
GST_BOILERPLATE (ADPCMEnc, adpcmenc, GstAudioEncoder, GST_TYPE_AUDIO_ENCODER);
static gboolean
adpcmenc_setup (ADPCMEnc * enc)
{
@ -153,6 +140,7 @@ adpcmenc_setup (ADPCMEnc * enc)
const int ADPCM_SAMPLES_PER_BYTE = 2;
guint64 sample_bytes;
const char *layout;
GstCaps *caps;
switch (enc->layout) {
case LAYOUT_ADPCM_DVI:
@ -168,21 +156,14 @@ adpcmenc_setup (ADPCMEnc * enc)
return FALSE;
}
enc->output_caps = gst_caps_new_simple ("audio/x-adpcm",
caps = gst_caps_new_simple ("audio/x-adpcm",
"rate", G_TYPE_INT, enc->rate,
"channels", G_TYPE_INT, enc->channels,
"layout", G_TYPE_STRING, layout,
"block_align", G_TYPE_INT, enc->blocksize, NULL);
if (enc->output_caps) {
gst_pad_set_caps (enc->srcpad, enc->output_caps);
}
enc->is_setup = TRUE;
enc->timestamp = GST_CLOCK_TIME_NONE;
enc->base_timestamp = GST_CLOCK_TIME_NONE;
enc->adapter = gst_adapter_new ();
enc->out_samples = 0;
gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps);
gst_caps_unref (caps);
/* Step index state is carried between blocks. */
enc->step_index[0] = 0;
@ -191,37 +172,21 @@ adpcmenc_setup (ADPCMEnc * enc)
return TRUE;
}
static void
adpcmenc_teardown (ADPCMEnc * enc)
{
if (enc->output_caps) {
gst_caps_unref (enc->output_caps);
enc->output_caps = NULL;
}
if (enc->adapter) {
g_object_unref (enc->adapter);
enc->adapter = NULL;
}
enc->is_setup = FALSE;
}
static gboolean
adpcmenc_sink_setcaps (GstPad * pad, GstCaps * caps)
adpcmenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
ADPCMEnc *enc = (ADPCMEnc *) gst_pad_get_parent (pad);
GstStructure *structure = gst_caps_get_structure (caps, 0);
ADPCMEnc *enc = (ADPCMEnc *) (benc);
if (!gst_structure_get_int (structure, "rate", &enc->rate))
return FALSE;
if (!gst_structure_get_int (structure, "channels", &enc->channels))
enc->rate = GST_AUDIO_INFO_RATE (info);
enc->channels = GST_AUDIO_INFO_CHANNELS (info);
if (!adpcmenc_setup (enc))
return FALSE;
if (enc->is_setup) {
adpcmenc_teardown (enc);
}
adpcmenc_setup (enc);
gst_object_unref (enc);
/* report needs to base class */
gst_audio_encoder_set_frame_samples_min (benc, enc->samples_per_block);
gst_audio_encoder_set_frame_samples_max (benc, enc->samples_per_block);
gst_audio_encoder_set_frame_max (benc, 1);
return TRUE;
}
@ -368,148 +333,86 @@ adpcmenc_encode_ima_block (ADPCMEnc * enc, const gint16 * samples,
return TRUE;
}
static GstFlowReturn
static GstBuffer *
adpcmenc_encode_block (ADPCMEnc * enc, const gint16 * samples, int blocksize)
{
gboolean res;
gboolean res = FALSE;
GstBuffer *outbuf = NULL;
if (enc->layout == LAYOUT_ADPCM_DVI) {
outbuf = gst_buffer_new_and_alloc (enc->blocksize);
res = adpcmenc_encode_ima_block (enc, samples, GST_BUFFER_DATA (outbuf));
} else {
/* should not happen afaics */
g_assert_not_reached ();
GST_WARNING_OBJECT (enc, "Unknown layout");
return GST_FLOW_ERROR;
res = FALSE;
}
if (!res) {
gst_buffer_unref (outbuf);
if (outbuf)
gst_buffer_unref (outbuf);
outbuf = NULL;
GST_WARNING_OBJECT (enc, "Encode of block failed");
return GST_FLOW_ERROR;
}
gst_buffer_set_caps (outbuf, enc->output_caps);
GST_BUFFER_TIMESTAMP (outbuf) = enc->timestamp;
enc->out_samples += enc->samples_per_block;
enc->timestamp = enc->base_timestamp +
gst_util_uint64_scale_int (enc->out_samples, GST_SECOND, enc->rate);
GST_BUFFER_DURATION (outbuf) = enc->timestamp - GST_BUFFER_TIMESTAMP (outbuf);
return gst_pad_push (enc->srcpad, outbuf);
return outbuf;
}
static GstFlowReturn
adpcmenc_chain (GstPad * pad, GstBuffer * buf)
adpcmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
{
ADPCMEnc *enc = (ADPCMEnc *) gst_pad_get_parent (pad);
ADPCMEnc *enc = (ADPCMEnc *) (benc);
GstFlowReturn ret = GST_FLOW_OK;
gint16 *samples;
GstBuffer *databuf = NULL;
GstBuffer *outbuf;
int input_bytes_per_block;
const int BYTES_PER_SAMPLE = 2;
if (enc->base_timestamp == GST_CLOCK_TIME_NONE) {
enc->base_timestamp = GST_BUFFER_TIMESTAMP (buf);
if (enc->base_timestamp == GST_CLOCK_TIME_NONE)
enc->base_timestamp = 0;
enc->timestamp = enc->base_timestamp;
/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buffer == NULL)) {
GST_DEBUG_OBJECT (benc, "no data");
goto done;
}
gst_adapter_push (enc->adapter, buf);
input_bytes_per_block =
enc->samples_per_block * BYTES_PER_SAMPLE * enc->channels;
while (gst_adapter_available (enc->adapter) >= input_bytes_per_block) {
databuf = gst_adapter_take_buffer (enc->adapter, input_bytes_per_block);
samples = (gint16 *) GST_BUFFER_DATA (databuf);
ret = adpcmenc_encode_block (enc, samples, enc->blocksize);
gst_buffer_unref (databuf);
if (ret != GST_FLOW_OK)
goto done;
if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < input_bytes_per_block)) {
GST_DEBUG_OBJECT (enc, "discarding trailing data %d",
GST_BUFFER_SIZE (buffer));
ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
goto done;
}
samples = (gint16 *) GST_BUFFER_DATA (buffer);
outbuf = adpcmenc_encode_block (enc, samples, enc->blocksize);
ret = gst_audio_encoder_finish_frame (benc, outbuf, enc->samples_per_block);
done:
gst_object_unref (enc);
return ret;
}
static gboolean
adpcmenc_sink_event (GstPad * pad, GstEvent * event)
adpcmenc_start (GstAudioEncoder * enc)
{
ADPCMEnc *enc = (ADPCMEnc *) gst_pad_get_parent (pad);
gboolean res;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
enc->out_samples = 0;
enc->timestamp = GST_CLOCK_TIME_NONE;
enc->base_timestamp = GST_CLOCK_TIME_NONE;
gst_adapter_clear (enc->adapter);
/* Fall through */
default:
res = gst_pad_push_event (enc->srcpad, event);
break;
}
gst_object_unref (enc);
return res;
GST_DEBUG_OBJECT (enc, "start");
return TRUE;
}
static GstStateChangeReturn
adpcmenc_change_state (GstElement * element, GstStateChange transition)
static gboolean
adpcmenc_stop (GstAudioEncoder * enc)
{
GstStateChangeReturn ret;
ADPCMEnc *enc = (ADPCMEnc *) element;
GST_DEBUG_OBJECT (enc, "stop");
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
adpcmenc_teardown (enc);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static void
adpcmenc_dispose (GObject * obj)
{
G_OBJECT_CLASS (parent_class)->dispose (obj);
return TRUE;
}
static void
adpcmenc_init (ADPCMEnc * enc, ADPCMEncClass * klass)
{
enc->sinkpad =
gst_pad_new_from_static_template (&adpcmenc_sink_template, "sink");
gst_pad_set_setcaps_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (adpcmenc_sink_setcaps));
gst_pad_set_chain_function (enc->sinkpad, GST_DEBUG_FUNCPTR (adpcmenc_chain));
gst_pad_set_event_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (adpcmenc_sink_event));
gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
enc->srcpad =
gst_pad_new_from_static_template (&adpcmenc_src_template, "src");
gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
/* Set defaults. */
enc->blocksize = DEFAULT_ADPCM_BLOCK_SIZE;
enc->layout = DEFAULT_ADPCM_LAYOUT;
@ -519,11 +422,16 @@ static void
adpcmenc_class_init (ADPCMEncClass * klass)
{
GObjectClass *gobjectclass = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) klass;
gobjectclass->set_property = adpcmenc_set_property;
gobjectclass->get_property = adpcmenc_get_property;
base_class->start = GST_DEBUG_FUNCPTR (adpcmenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (adpcmenc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (adpcmenc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (adpcmenc_handle_frame);
g_object_class_install_property (gobjectclass, ARG_LAYOUT,
g_param_spec_enum ("layout", "Layout",
"Layout for output stream",
@ -537,10 +445,9 @@ adpcmenc_class_init (ADPCMEncClass * klass)
DEFAULT_ADPCM_BLOCK_SIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gobjectclass->dispose = adpcmenc_dispose;
gstelement_class->change_state = adpcmenc_change_state;
} static void
}
static void
adpcmenc_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);