lame: port to audioencoder

This commit is contained in:
Mark Nauwelaerts 2011-09-23 15:26:48 +02:00
parent 5c322ede2b
commit e15c5ae76e
2 changed files with 102 additions and 230 deletions

View file

@ -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)
{

View file

@ -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);