mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 02:31:03 +00:00
lame: port to audioencoder
This commit is contained in:
Mark Nauwelaerts
parent
5c322ede2b
commit
e15c5ae76e
2 changed files with 102 additions and 230 deletions
|
|
@ -21,7 +21,7 @@
|
|||
|
||||
/**
|
||||
* SECTION:element-lame
|
||||
* @see_also: lamemp3enc, mad, vorbisenc
|
||||
* @see_also: lame, mad, vorbisenc
|
||||
*
|
||||
* This element encodes raw integer audio into an MPEG-1 layer 3 (MP3) stream.
|
||||
* Note that <ulink url="http://en.wikipedia.org/wiki/MP3">MP3</ulink> is not
|
||||
|
|
@ -31,7 +31,7 @@
|
|||
*
|
||||
* <refsect2>
|
||||
* <title>Note</title>
|
||||
* This element is deprecated, use the lamemp3enc element instead
|
||||
* This element is deprecated, use the lame element instead
|
||||
* which provides a much simpler interface and results in better MP3 files.
|
||||
* </refsect2>
|
||||
*
|
||||
|
|
@ -309,15 +309,19 @@ static void gst_lame_base_init (gpointer g_class);
|
|||
static void gst_lame_class_init (GstLameClass * klass);
|
||||
static void gst_lame_init (GstLame * gst_lame);
|
||||
|
||||
static gboolean gst_lame_start (GstAudioEncoder * enc);
|
||||
static gboolean gst_lame_stop (GstAudioEncoder * enc);
|
||||
static gboolean gst_lame_set_format (GstAudioEncoder * enc,
|
||||
GstAudioInfo * info);
|
||||
static GstFlowReturn gst_lame_handle_frame (GstAudioEncoder * enc,
|
||||
GstBuffer * in_buf);
|
||||
static void gst_lame_flush (GstAudioEncoder * enc);
|
||||
|
||||
static void gst_lame_set_property (GObject * object, guint prop_id,
|
||||
const GValue * value, GParamSpec * pspec);
|
||||
static void gst_lame_get_property (GObject * object, guint prop_id,
|
||||
GValue * value, GParamSpec * pspec);
|
||||
static gboolean gst_lame_sink_event (GstPad * pad, GstEvent * event);
|
||||
static GstFlowReturn gst_lame_chain (GstPad * pad, GstBuffer * buf);
|
||||
static gboolean gst_lame_setup (GstLame * lame);
|
||||
static GstStateChangeReturn gst_lame_change_state (GstElement * element,
|
||||
GstStateChange transition);
|
||||
|
||||
static GstElementClass *parent_class = NULL;
|
||||
|
||||
|
|
@ -352,7 +356,8 @@ gst_lame_get_type (void)
|
|||
};
|
||||
|
||||
gst_lame_type =
|
||||
g_type_register_static (GST_TYPE_ELEMENT, "GstLame", &gst_lame_info, 0);
|
||||
g_type_register_static (GST_TYPE_AUDIO_ENCODER, "GstLame",
|
||||
&gst_lame_info, 0);
|
||||
g_type_add_interface_static (gst_lame_type, GST_TYPE_TAG_SETTER,
|
||||
&tag_setter_info);
|
||||
g_type_add_interface_static (gst_lame_type, GST_TYPE_PRESET, &preset_info);
|
||||
|
|
@ -397,9 +402,11 @@ gst_lame_class_init (GstLameClass * klass)
|
|||
{
|
||||
GObjectClass *gobject_class;
|
||||
GstElementClass *gstelement_class;
|
||||
GstAudioEncoderClass *base_class;
|
||||
|
||||
gobject_class = (GObjectClass *) klass;
|
||||
gstelement_class = (GstElementClass *) klass;
|
||||
base_class = (GstAudioEncoderClass *) klass;
|
||||
|
||||
parent_class = g_type_class_peek_parent (klass);
|
||||
|
||||
|
|
@ -407,6 +414,12 @@ gst_lame_class_init (GstLameClass * klass)
|
|||
gobject_class->get_property = gst_lame_get_property;
|
||||
gobject_class->finalize = gst_lame_finalize;
|
||||
|
||||
base_class->start = GST_DEBUG_FUNCPTR (gst_lame_start);
|
||||
base_class->stop = GST_DEBUG_FUNCPTR (gst_lame_stop);
|
||||
base_class->set_format = GST_DEBUG_FUNCPTR (gst_lame_set_format);
|
||||
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_lame_handle_frame);
|
||||
base_class->flush = GST_DEBUG_FUNCPTR (gst_lame_flush);
|
||||
|
||||
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BITRATE,
|
||||
g_param_spec_int ("bitrate", "Bitrate (kb/s)",
|
||||
"Bitrate in kbit/sec (8, 16, 24, 32, 40, 48, 56, 64, 80, 96, "
|
||||
|
|
@ -565,39 +578,30 @@ gst_lame_class_init (GstLameClass * klass)
|
|||
GST_TYPE_LAME_PRESET, gst_lame_default_settings.preset,
|
||||
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
||||
#endif
|
||||
|
||||
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_lame_change_state);
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_lame_src_setcaps (GstPad * pad, GstCaps * caps)
|
||||
{
|
||||
GST_DEBUG_OBJECT (pad, "caps: %" GST_PTR_FORMAT, caps);
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_lame_sink_setcaps (GstPad * pad, GstCaps * caps)
|
||||
gst_lame_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
|
||||
{
|
||||
GstLame *lame;
|
||||
gint out_samplerate;
|
||||
gint version;
|
||||
GstStructure *structure;
|
||||
GstCaps *othercaps;
|
||||
GstClockTime latency;
|
||||
|
||||
lame = GST_LAME (GST_PAD_PARENT (pad));
|
||||
structure = gst_caps_get_structure (caps, 0);
|
||||
lame = GST_LAME (enc);
|
||||
|
||||
if (!gst_structure_get_int (structure, "rate", &lame->samplerate))
|
||||
goto no_rate;
|
||||
if (!gst_structure_get_int (structure, "channels", &lame->num_channels))
|
||||
goto no_channels;
|
||||
/* parameters already parsed for us */
|
||||
lame->samplerate = GST_AUDIO_INFO_RATE (info);
|
||||
lame->num_channels = GST_AUDIO_INFO_CHANNELS (info);
|
||||
|
||||
/* but we might be asked to reconfigure, so reset */
|
||||
gst_lame_release_memory (lame);
|
||||
|
||||
GST_DEBUG_OBJECT (lame, "setting up lame");
|
||||
if (!gst_lame_setup (lame))
|
||||
goto setup_failed;
|
||||
|
||||
|
||||
out_samplerate = lame_get_out_samplerate (lame->lgf);
|
||||
if (out_samplerate == 0)
|
||||
goto zero_output_rate;
|
||||
|
|
@ -624,21 +628,18 @@ gst_lame_sink_setcaps (GstPad * pad, GstCaps * caps)
|
|||
"rate", G_TYPE_INT, out_samplerate, NULL);
|
||||
|
||||
/* and use these caps */
|
||||
gst_pad_set_caps (lame->srcpad, othercaps);
|
||||
gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (lame), othercaps);
|
||||
gst_caps_unref (othercaps);
|
||||
|
||||
/* base class feedback:
|
||||
* - we will handle buffers, just hand us all available
|
||||
* - report latency */
|
||||
latency = gst_util_uint64_scale_int (lame_get_framesize (lame->lgf),
|
||||
GST_SECOND, lame->samplerate);
|
||||
gst_audio_encoder_set_latency (enc, latency, latency);
|
||||
|
||||
return TRUE;
|
||||
|
||||
no_rate:
|
||||
{
|
||||
GST_ERROR_OBJECT (lame, "input caps have no sample rate field");
|
||||
return FALSE;
|
||||
}
|
||||
no_channels:
|
||||
{
|
||||
GST_ERROR_OBJECT (lame, "input caps have no channels field");
|
||||
return FALSE;
|
||||
}
|
||||
zero_output_rate:
|
||||
{
|
||||
GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS, (NULL),
|
||||
|
|
@ -658,26 +659,6 @@ gst_lame_init (GstLame * lame)
|
|||
{
|
||||
GST_DEBUG_OBJECT (lame, "starting initialization");
|
||||
|
||||
lame->sinkpad =
|
||||
gst_pad_new_from_static_template (&gst_lame_sink_template, "sink");
|
||||
gst_pad_set_event_function (lame->sinkpad,
|
||||
GST_DEBUG_FUNCPTR (gst_lame_sink_event));
|
||||
gst_pad_set_chain_function (lame->sinkpad,
|
||||
GST_DEBUG_FUNCPTR (gst_lame_chain));
|
||||
gst_pad_set_setcaps_function (lame->sinkpad,
|
||||
GST_DEBUG_FUNCPTR (gst_lame_sink_setcaps));
|
||||
gst_element_add_pad (GST_ELEMENT (lame), lame->sinkpad);
|
||||
|
||||
lame->srcpad =
|
||||
gst_pad_new_from_static_template (&gst_lame_src_template, "src");
|
||||
gst_pad_set_setcaps_function (lame->srcpad,
|
||||
GST_DEBUG_FUNCPTR (gst_lame_src_setcaps));
|
||||
gst_element_add_pad (GST_ELEMENT (lame), lame->srcpad);
|
||||
|
||||
lame->samplerate = 44100;
|
||||
lame->num_channels = 2;
|
||||
lame->setup = FALSE;
|
||||
|
||||
/* Set default settings */
|
||||
lame->bitrate = gst_lame_default_settings.bitrate;
|
||||
lame->compression_ratio = gst_lame_default_settings.compression_ratio;
|
||||
|
|
@ -714,6 +695,27 @@ gst_lame_init (GstLame * lame)
|
|||
GST_DEBUG_OBJECT (lame, "done initializing");
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_lame_start (GstAudioEncoder * enc)
|
||||
{
|
||||
GstLame *lame = GST_LAME (enc);
|
||||
|
||||
GST_DEBUG_OBJECT (lame, "start");
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_lame_stop (GstAudioEncoder * enc)
|
||||
{
|
||||
GstLame *lame = GST_LAME (enc);
|
||||
|
||||
GST_DEBUG_OBJECT (lame, "stop");
|
||||
|
||||
gst_lame_release_memory (lame);
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
|
||||
/* <php-emulation-mode>three underscores for ___rate is really really really
|
||||
* private as opposed to one underscore<php-emulation-mode> */
|
||||
/* call this MACRO outside of the NULL state so that we have a higher chance
|
||||
|
|
@ -979,108 +981,54 @@ gst_lame_get_property (GObject * object, guint prop_id, GValue * value,
|
|||
}
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_lame_sink_event (GstPad * pad, GstEvent * event)
|
||||
static GstFlowReturn
|
||||
gst_lame_flush_full (GstLame * lame, gboolean push)
|
||||
{
|
||||
gboolean ret;
|
||||
GstLame *lame;
|
||||
GstBuffer *buf;
|
||||
gint size;
|
||||
GstFlowReturn result = GST_FLOW_OK;
|
||||
|
||||
lame = GST_LAME (gst_pad_get_parent (pad));
|
||||
if (!lame->lgf)
|
||||
return GST_FLOW_OK;
|
||||
|
||||
switch (GST_EVENT_TYPE (event)) {
|
||||
case GST_EVENT_EOS:{
|
||||
GST_DEBUG_OBJECT (lame, "handling EOS event");
|
||||
buf = gst_buffer_new_and_alloc (7200);
|
||||
size = lame_encode_flush (lame->lgf, GST_BUFFER_DATA (buf), 7200);
|
||||
|
||||
if (lame->lgf != NULL) {
|
||||
GstBuffer *buf;
|
||||
gint size;
|
||||
|
||||
buf = gst_buffer_new_and_alloc (7200);
|
||||
size = lame_encode_flush (lame->lgf, GST_BUFFER_DATA (buf), 7200);
|
||||
|
||||
if (size > 0 && lame->last_flow == GST_FLOW_OK) {
|
||||
gint64 duration;
|
||||
|
||||
duration = gst_util_uint64_scale (size, 8 * GST_SECOND,
|
||||
1000 * lame->bitrate);
|
||||
|
||||
if (lame->last_ts == GST_CLOCK_TIME_NONE) {
|
||||
lame->last_ts = lame->eos_ts;
|
||||
lame->last_duration = duration;
|
||||
} else {
|
||||
lame->last_duration += duration;
|
||||
}
|
||||
|
||||
GST_BUFFER_TIMESTAMP (buf) = lame->last_ts;
|
||||
GST_BUFFER_DURATION (buf) = lame->last_duration;
|
||||
lame->last_ts = GST_CLOCK_TIME_NONE;
|
||||
GST_BUFFER_SIZE (buf) = size;
|
||||
GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
|
||||
gst_buffer_set_caps (buf, GST_PAD_CAPS (lame->srcpad));
|
||||
gst_pad_push (lame->srcpad, buf);
|
||||
} else {
|
||||
GST_DEBUG_OBJECT (lame, "no final packet (size=%d, last_flow=%s)",
|
||||
size, gst_flow_get_name (lame->last_flow));
|
||||
gst_buffer_unref (buf);
|
||||
}
|
||||
}
|
||||
|
||||
ret = gst_pad_event_default (pad, event);
|
||||
break;
|
||||
}
|
||||
case GST_EVENT_FLUSH_START:
|
||||
GST_DEBUG_OBJECT (lame, "handling FLUSH start event");
|
||||
/* forward event */
|
||||
ret = gst_pad_push_event (lame->srcpad, event);
|
||||
break;
|
||||
case GST_EVENT_FLUSH_STOP:
|
||||
{
|
||||
guchar *mp3_data = NULL;
|
||||
gint mp3_buffer_size;
|
||||
|
||||
GST_DEBUG_OBJECT (lame, "handling FLUSH stop event");
|
||||
|
||||
if (lame->lgf) {
|
||||
/* clear buffers if we already have lame set up */
|
||||
mp3_buffer_size = 7200;
|
||||
mp3_data = g_malloc (mp3_buffer_size);
|
||||
lame_encode_flush (lame->lgf, mp3_data, mp3_buffer_size);
|
||||
g_free (mp3_data);
|
||||
}
|
||||
|
||||
ret = gst_pad_push_event (lame->srcpad, event);
|
||||
break;
|
||||
}
|
||||
case GST_EVENT_TAG:
|
||||
GST_DEBUG_OBJECT (lame, "ignoring TAG event, passing it on");
|
||||
ret = gst_pad_push_event (lame->srcpad, event);
|
||||
break;
|
||||
default:
|
||||
ret = gst_pad_event_default (pad, event);
|
||||
break;
|
||||
if (size > 0 && push) {
|
||||
GST_BUFFER_SIZE (buf) = size;
|
||||
GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
|
||||
result = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame), buf, -1);
|
||||
} else {
|
||||
GST_DEBUG_OBJECT (lame, "no final packet (size=%d, push=%d)", size, push);
|
||||
gst_buffer_unref (buf);
|
||||
result = GST_FLOW_OK;
|
||||
}
|
||||
gst_object_unref (lame);
|
||||
return ret;
|
||||
return result;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_lame_flush (GstAudioEncoder * enc)
|
||||
{
|
||||
gst_lame_flush_full (GST_LAME (enc), FALSE);
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
gst_lame_chain (GstPad * pad, GstBuffer * buf)
|
||||
gst_lame_handle_frame (GstAudioEncoder * enc, GstBuffer * buf)
|
||||
{
|
||||
GstLame *lame;
|
||||
guchar *mp3_data;
|
||||
GstBuffer *mp3_buf;
|
||||
gint mp3_buffer_size, mp3_size;
|
||||
gint64 duration;
|
||||
GstFlowReturn result;
|
||||
gint num_samples;
|
||||
guint8 *data;
|
||||
guint size;
|
||||
|
||||
lame = GST_LAME (GST_PAD_PARENT (pad));
|
||||
lame = GST_LAME (enc);
|
||||
|
||||
GST_LOG_OBJECT (lame, "entered chain");
|
||||
|
||||
if (!lame->setup)
|
||||
goto not_setup;
|
||||
/* squeeze remaining and push */
|
||||
if (G_UNLIKELY (buf == NULL))
|
||||
return gst_lame_flush_full (lame, TRUE);
|
||||
|
||||
data = GST_BUFFER_DATA (buf);
|
||||
size = GST_BUFFER_SIZE (buf);
|
||||
|
|
@ -1089,7 +1037,8 @@ gst_lame_chain (GstPad * pad, GstBuffer * buf)
|
|||
|
||||
/* allocate space for output */
|
||||
mp3_buffer_size = 1.25 * num_samples + 7200;
|
||||
mp3_data = g_malloc (mp3_buffer_size);
|
||||
mp3_buf = gst_buffer_new_and_alloc (mp3_buffer_size);
|
||||
mp3_data = GST_BUFFER_DATA (mp3_buf);
|
||||
|
||||
/* lame seems to be too stupid to get mono interleaved going */
|
||||
if (lame->num_channels == 1) {
|
||||
|
|
@ -1105,69 +1054,23 @@ gst_lame_chain (GstPad * pad, GstBuffer * buf)
|
|||
GST_LOG_OBJECT (lame, "encoded %d bytes of audio to %d bytes of mp3",
|
||||
size, mp3_size);
|
||||
|
||||
duration = gst_util_uint64_scale_int (size, GST_SECOND,
|
||||
2 * lame->samplerate * lame->num_channels);
|
||||
|
||||
if (GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE &&
|
||||
GST_BUFFER_DURATION (buf) != duration) {
|
||||
GST_DEBUG_OBJECT (lame, "incoming buffer had incorrect duration %"
|
||||
GST_TIME_FORMAT ", outgoing buffer will have correct duration %"
|
||||
GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_TIME_ARGS (duration));
|
||||
}
|
||||
|
||||
if (lame->last_ts == GST_CLOCK_TIME_NONE) {
|
||||
lame->last_ts = GST_BUFFER_TIMESTAMP (buf);
|
||||
lame->last_offs = GST_BUFFER_OFFSET (buf);
|
||||
lame->last_duration = duration;
|
||||
} else {
|
||||
lame->last_duration += duration;
|
||||
}
|
||||
|
||||
gst_buffer_unref (buf);
|
||||
|
||||
if (mp3_size < 0) {
|
||||
g_warning ("error %d", mp3_size);
|
||||
}
|
||||
|
||||
if (mp3_size > 0) {
|
||||
GstBuffer *outbuf;
|
||||
|
||||
outbuf = gst_buffer_new ();
|
||||
GST_BUFFER_DATA (outbuf) = mp3_data;
|
||||
GST_BUFFER_MALLOCDATA (outbuf) = mp3_data;
|
||||
GST_BUFFER_SIZE (outbuf) = mp3_size;
|
||||
GST_BUFFER_TIMESTAMP (outbuf) = lame->last_ts;
|
||||
GST_BUFFER_OFFSET (outbuf) = lame->last_offs;
|
||||
GST_BUFFER_DURATION (outbuf) = lame->last_duration;
|
||||
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (lame->srcpad));
|
||||
|
||||
result = gst_pad_push (lame->srcpad, outbuf);
|
||||
lame->last_flow = result;
|
||||
if (result != GST_FLOW_OK) {
|
||||
GST_DEBUG_OBJECT (lame, "flow return: %s", gst_flow_get_name (result));
|
||||
}
|
||||
|
||||
if (GST_CLOCK_TIME_IS_VALID (lame->last_ts))
|
||||
lame->eos_ts = lame->last_ts + lame->last_duration;
|
||||
else
|
||||
lame->eos_ts = GST_CLOCK_TIME_NONE;
|
||||
lame->last_ts = GST_CLOCK_TIME_NONE;
|
||||
if (G_LIKELY (mp3_size > 0)) {
|
||||
GST_BUFFER_SIZE (mp3_buf) = mp3_size;
|
||||
result = gst_audio_encoder_finish_frame (enc, mp3_buf, -1);
|
||||
} else {
|
||||
g_free (mp3_data);
|
||||
if (mp3_size < 0) {
|
||||
/* eat error ? */
|
||||
g_warning ("error %d", mp3_size);
|
||||
}
|
||||
result = GST_FLOW_OK;
|
||||
gst_buffer_unref (mp3_buf);
|
||||
}
|
||||
|
||||
return result;
|
||||
|
||||
/* ERRORS */
|
||||
not_setup:
|
||||
{
|
||||
gst_buffer_unref (buf);
|
||||
GST_ELEMENT_ERROR (lame, CORE, NEGOTIATION, (NULL),
|
||||
("encoder not initialized (input is not audio?)"));
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
}
|
||||
|
||||
/* set up the encoder state */
|
||||
|
|
@ -1204,7 +1107,7 @@ gst_lame_setup (GstLame * lame)
|
|||
lame_set_in_samplerate (lame->lgf, lame->samplerate);
|
||||
|
||||
/* let lame choose default samplerate unless outgoing sample rate is fixed */
|
||||
allowed_caps = gst_pad_get_allowed_caps (lame->srcpad);
|
||||
allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (lame));
|
||||
|
||||
if (allowed_caps != NULL) {
|
||||
GstStructure *structure;
|
||||
|
|
@ -1294,37 +1197,6 @@ gst_lame_setup (GstLame * lame)
|
|||
#undef CHECK_ERROR
|
||||
}
|
||||
|
||||
static GstStateChangeReturn
|
||||
gst_lame_change_state (GstElement * element, GstStateChange transition)
|
||||
{
|
||||
GstLame *lame;
|
||||
GstStateChangeReturn result;
|
||||
|
||||
lame = GST_LAME (element);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
lame->last_flow = GST_FLOW_OK;
|
||||
lame->last_ts = GST_CLOCK_TIME_NONE;
|
||||
lame->eos_ts = GST_CLOCK_TIME_NONE;
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_READY_TO_NULL:
|
||||
gst_lame_release_memory (lame);
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
return result;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_lame_get_default_settings (void)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -27,6 +27,7 @@
|
|||
G_BEGIN_DECLS
|
||||
|
||||
#include <lame/lame.h>
|
||||
#include <gst/audio/gstaudioencoder.h>
|
||||
|
||||
#define GST_TYPE_LAME \
|
||||
(gst_lame_get_type())
|
||||
|
|
@ -48,10 +49,9 @@ typedef struct _GstLameClass GstLameClass;
|
|||
* Opaque data structure.
|
||||
*/
|
||||
struct _GstLame {
|
||||
GstElement element;
|
||||
GstAudioEncoder element;
|
||||
|
||||
/*< private >*/
|
||||
GstPad *srcpad, *sinkpad;
|
||||
|
||||
gint samplerate;
|
||||
gint num_channels;
|
||||
|
|
@ -100,7 +100,7 @@ struct _GstLame {
|
|||
};
|
||||
|
||||
struct _GstLameClass {
|
||||
GstElementClass parent_class;
|
||||
GstAudioEncoderClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_lame_get_type(void);
|
||||
|
|
|
|||
Loading…
Reference in a new issue