lamemp3enc: port to audioencoder

This commit is contained in:
Mark Nauwelaerts 2011-09-26 14:44:01 +02:00
parent 40274ad92f
commit 2bfec2cccc
3 changed files with 119 additions and 420 deletions

View file

@ -1,8 +1,10 @@
plugin_LTLIBRARIES = libgstlame.la
libgstlame_la_SOURCES = gstlame.c gstlamemp3enc.c plugin.c
libgstlame_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(LAME_CFLAGS)
libgstlame_la_LIBADD = $(LAME_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstpbutils-@GST_MAJORMINOR@ $(GST_LIBS)
libgstlame_la_CFLAGS = -DGST_USE_UNSTABLE_API \
$(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(LAME_CFLAGS)
libgstlame_la_LIBADD = $(LAME_LIBS) $(GST_PLUGINS_BASE_LIBS) \
-lgstaudio-$(GST_MAJORMINOR) $(GST_LIBS)
libgstlame_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstlame_la_LIBTOOLFLAGS = --tag=disable-static

View file

@ -68,8 +68,6 @@
#include "gstlamemp3enc.h"
#include <gst/gst-i18n-plugin.h>
#include <gst/pbutils/descriptions.h>
/* lame < 3.98 */
#ifndef HAVE_LAME_SET_VBR_QUALITY
#define lame_set_VBR_quality(flags,q) lame_set_VBR_q((flags),(int)(q))
@ -182,15 +180,19 @@ static void gst_lamemp3enc_base_init (gpointer g_class);
static void gst_lamemp3enc_class_init (GstLameMP3EncClass * klass);
static void gst_lamemp3enc_init (GstLameMP3Enc * gst_lame);
static gboolean gst_lamemp3enc_start (GstAudioEncoder * enc);
static gboolean gst_lamemp3enc_stop (GstAudioEncoder * enc);
static gboolean gst_lamemp3enc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_lamemp3enc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
static void gst_lamemp3enc_flush (GstAudioEncoder * enc);
static void gst_lamemp3enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_lamemp3enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_lamemp3enc_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_lamemp3enc_chain (GstPad * pad, GstBuffer * buf);
static gboolean gst_lamemp3enc_setup (GstLameMP3Enc * lame, GstTagList ** tags);
static GstStateChangeReturn gst_lamemp3enc_change_state (GstElement * element,
GstStateChange transition);
static GstElementClass *parent_class = NULL;
@ -211,17 +213,10 @@ gst_lamemp3enc_get_type (void)
0,
(GInstanceInitFunc) gst_lamemp3enc_init,
};
static const GInterfaceInfo preset_info = {
NULL,
NULL,
NULL
};
gst_lamemp3enc_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstLameMP3Enc",
g_type_register_static (GST_TYPE_AUDIO_ENCODER, "GstLameMP3Enc",
&gst_lamemp3enc_info, 0);
g_type_add_interface_static (gst_lamemp3enc_type, GST_TYPE_PRESET,
&preset_info);
}
return gst_lamemp3enc_type;
}
@ -263,9 +258,11 @@ gst_lamemp3enc_class_init (GstLameMP3EncClass * 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);
@ -273,64 +270,89 @@ gst_lamemp3enc_class_init (GstLameMP3EncClass * klass)
gobject_class->get_property = gst_lamemp3enc_get_property;
gobject_class->finalize = gst_lamemp3enc_finalize;
base_class->start = GST_DEBUG_FUNCPTR (gst_lamemp3enc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_lamemp3enc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_lamemp3enc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_lamemp3enc_handle_frame);
base_class->flush = GST_DEBUG_FUNCPTR (gst_lamemp3enc_flush);
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_TARGET,
g_param_spec_enum ("target", "Target",
"Optimize for quality or bitrate", GST_TYPE_LAMEMP3ENC_TARGET,
DEFAULT_TARGET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
DEFAULT_TARGET,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BITRATE,
g_param_spec_int ("bitrate", "Bitrate (kb/s)",
"Bitrate in kbit/sec (Only valid if target is bitrate, for CBR one "
"of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, "
"256 or 320)", 8, 320, DEFAULT_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_CBR,
g_param_spec_boolean ("cbr", "CBR", "Enforce constant bitrate encoding "
"(Only valid if target is bitrate)", DEFAULT_CBR,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_QUALITY,
g_param_spec_float ("quality", "Quality",
"VBR Quality from 0 to 10, 0 being the best "
"(Only valid if target is quality)", 0.0, 9.999,
DEFAULT_QUALITY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
DEFAULT_QUALITY,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
ARG_ENCODING_ENGINE_QUALITY, g_param_spec_enum ("encoding-engine-quality",
"Encoding Engine Quality", "Quality/speed of the encoding engine, "
"this does not affect the bitrate!",
GST_TYPE_LAMEMP3ENC_ENCODING_ENGINE_QUALITY,
DEFAULT_ENCODING_ENGINE_QUALITY,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MONO,
g_param_spec_boolean ("mono", "Mono", "Enforce mono encoding",
DEFAULT_MONO, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
DEFAULT_MONO,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_lamemp3enc_change_state);
static void
gst_lamemp3enc_init (GstLameMP3Enc * lame)
{
}
static gboolean
gst_lamemp3enc_src_setcaps (GstPad * pad, GstCaps * caps)
gst_lamemp3enc_start (GstAudioEncoder * enc)
{
GST_DEBUG_OBJECT (pad, "caps: %" GST_PTR_FORMAT, caps);
GstLameMP3Enc *lame = GST_LAMEMP3ENC (enc);
GST_DEBUG_OBJECT (lame, "start");
return TRUE;
}
static gboolean
gst_lamemp3enc_sink_setcaps (GstPad * pad, GstCaps * caps)
gst_lamemp3enc_stop (GstAudioEncoder * enc)
{
GstLameMP3Enc *lame = GST_LAMEMP3ENC (enc);
GST_DEBUG_OBJECT (lame, "stop");
gst_lamemp3enc_release_memory (lame);
return TRUE;
}
static gboolean
gst_lamemp3enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstLameMP3Enc *lame;
gint out_samplerate;
gint version;
GstStructure *structure;
GstCaps *othercaps;
GstClockTime latency;
GstTagList *tags = NULL;
lame = GST_LAMEMP3ENC (GST_PAD_PARENT (pad));
structure = gst_caps_get_structure (caps, 0);
lame = GST_LAMEMP3ENC (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_lamemp3enc_release_memory (lame);
GST_DEBUG_OBJECT (lame, "setting up lame");
if (!gst_lamemp3enc_setup (lame, &tags))
@ -362,39 +384,27 @@ gst_lamemp3enc_sink_setcaps (GstPad * pad, GstCaps * caps)
"rate", G_TYPE_INT, out_samplerate, NULL);
/* and use these caps */
gst_pad_set_caps (lame->srcpad, othercaps);
if (tags) {
gst_pb_utils_add_codec_description_to_tag_list (tags, GST_TAG_CODEC,
othercaps);
gst_pb_utils_add_codec_description_to_tag_list (tags, GST_TAG_AUDIO_CODEC,
othercaps);
}
gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), 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);
if (tags)
gst_element_found_tags_for_pad (GST_ELEMENT_CAST (lame), lame->srcpad,
tags);
gst_audio_encoder_merge_tags (enc, tags, GST_TAG_MERGE_REPLACE);
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),
("LAMEMP3ENC decided on a zero sample rate"));
if (tags)
gst_tag_list_free (tags);
GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS, (NULL),
("LAMEMP3ENC decided on a zero sample rate"));
return FALSE;
}
setup_failed:
@ -405,152 +415,6 @@ setup_failed:
}
}
static GstCaps *
gst_lamemp3enc_sink_getcaps (GstPad * pad)
{
const GstCaps *templ_caps;
GstLameMP3Enc *lame;
GstCaps *allowed = NULL;
GstCaps *caps, *filter_caps;
gint i, j;
lame = GST_LAMEMP3ENC (gst_pad_get_parent (pad));
/* we want to be able to communicate to upstream elements like audioconvert
* and audioresample any rate/channel restrictions downstream (e.g. muxer
* only accepting certain sample rates) */
templ_caps = gst_pad_get_pad_template_caps (pad);
allowed = gst_pad_get_allowed_caps (lame->srcpad);
if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) {
caps = gst_caps_copy (templ_caps);
goto done;
}
filter_caps = gst_caps_new_empty ();
for (i = 0; i < gst_caps_get_size (templ_caps); i++) {
GQuark q_name;
q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
/* pick rate + channel fields from allowed caps */
for (j = 0; j < gst_caps_get_size (allowed); j++) {
const GstStructure *allowed_s = gst_caps_get_structure (allowed, j);
const GValue *val;
GstStructure *s;
s = gst_structure_id_empty_new (q_name);
if ((val = gst_structure_get_value (allowed_s, "rate")))
gst_structure_set_value (s, "rate", val);
if ((val = gst_structure_get_value (allowed_s, "channels")))
gst_structure_set_value (s, "channels", val);
gst_caps_merge_structure (filter_caps, s);
}
}
caps = gst_caps_intersect (filter_caps, templ_caps);
gst_caps_unref (filter_caps);
done:
gst_caps_replace (&allowed, NULL);
gst_object_unref (lame);
return caps;
}
static gint64
gst_lamemp3enc_get_latency (GstLameMP3Enc * lame)
{
return gst_util_uint64_scale_int (lame_get_framesize (lame->lgf),
GST_SECOND, lame->samplerate);
}
static gboolean
gst_lamemp3enc_src_query (GstPad * pad, GstQuery * query)
{
gboolean res = TRUE;
GstLameMP3Enc *lame;
GstPad *peerpad;
lame = GST_LAMEMP3ENC (gst_pad_get_parent (pad));
peerpad = gst_pad_get_peer (GST_PAD (lame->sinkpad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
if ((res = gst_pad_query (peerpad, query))) {
gboolean live;
GstClockTime min_latency, max_latency;
gint64 latency;
if (lame->lgf == NULL)
break;
gst_query_parse_latency (query, &live, &min_latency, &max_latency);
latency = gst_lamemp3enc_get_latency (lame);
/* add our latency */
min_latency += latency;
if (max_latency != -1)
max_latency += latency;
gst_query_set_latency (query, live, min_latency, max_latency);
}
break;
}
default:
res = gst_pad_query (peerpad, query);
break;
}
gst_object_unref (peerpad);
gst_object_unref (lame);
return res;
}
static void
gst_lamemp3enc_init (GstLameMP3Enc * lame)
{
GST_DEBUG_OBJECT (lame, "starting initialization");
lame->sinkpad =
gst_pad_new_from_static_template (&gst_lamemp3enc_sink_template, "sink");
gst_pad_set_event_function (lame->sinkpad,
GST_DEBUG_FUNCPTR (gst_lamemp3enc_sink_event));
gst_pad_set_chain_function (lame->sinkpad,
GST_DEBUG_FUNCPTR (gst_lamemp3enc_chain));
gst_pad_set_setcaps_function (lame->sinkpad,
GST_DEBUG_FUNCPTR (gst_lamemp3enc_sink_setcaps));
gst_pad_set_getcaps_function (lame->sinkpad,
GST_DEBUG_FUNCPTR (gst_lamemp3enc_sink_getcaps));
gst_element_add_pad (GST_ELEMENT (lame), lame->sinkpad);
lame->srcpad =
gst_pad_new_from_static_template (&gst_lamemp3enc_src_template, "src");
gst_pad_set_query_function (lame->srcpad,
GST_DEBUG_FUNCPTR (gst_lamemp3enc_src_query));
gst_pad_set_setcaps_function (lame->srcpad,
GST_DEBUG_FUNCPTR (gst_lamemp3enc_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->target = DEFAULT_TARGET;
lame->bitrate = DEFAULT_BITRATE;
lame->cbr = DEFAULT_CBR;
lame->quality = DEFAULT_QUALITY;
lame->encoding_engine_quality = DEFAULT_ENCODING_ENGINE_QUALITY;
lame->mono = DEFAULT_MONO;
GST_DEBUG_OBJECT (lame, "done initializing");
}
/* <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
@ -654,128 +518,64 @@ gst_lamemp3enc_get_property (GObject * object, guint prop_id, GValue * value,
}
}
static gboolean
gst_lamemp3enc_sink_event (GstPad * pad, GstEvent * event)
static GstFlowReturn
gst_lamemp3enc_flush_full (GstLameMP3Enc * lame, gboolean push)
{
gboolean ret;
GstLameMP3Enc *lame;
GstBuffer *buf;
gint size;
GstFlowReturn result = GST_FLOW_OK;
lame = GST_LAMEMP3ENC (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:{
GstTagList *tags;
gst_event_parse_tag (event, &tags);
tags = gst_tag_list_copy (tags);
gst_event_unref (event);
gst_tag_list_remove_tag (tags, GST_TAG_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_AUDIO_CODEC);
event = gst_event_new_tag (tags);
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_lamemp3enc_flush (GstAudioEncoder * enc)
{
gst_lamemp3enc_flush_full (GST_LAMEMP3ENC (enc), FALSE);
}
static GstFlowReturn
gst_lamemp3enc_chain (GstPad * pad, GstBuffer * buf)
gst_lamemp3enc_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
{
GstLameMP3Enc *lame;
guchar *mp3_data;
gint mp3_buffer_size, mp3_size;
gint64 duration;
GstBuffer *mp3_buf;
GstFlowReturn result;
gint num_samples;
guint8 *data;
guint size;
lame = GST_LAMEMP3ENC (GST_PAD_PARENT (pad));
lame = GST_LAMEMP3ENC (enc);
GST_LOG_OBJECT (lame, "entered chain");
/* squeeze remaining and push */
if (G_UNLIKELY (in_buf == NULL))
return gst_lamemp3enc_flush_full (lame, TRUE);
if (!lame->setup)
goto not_setup;
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
data = GST_BUFFER_DATA (in_buf);
size = GST_BUFFER_SIZE (in_buf);
num_samples = size / 2;
/* 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) {
@ -791,75 +591,26 @@ gst_lamemp3enc_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;
if (G_LIKELY (mp3_size > 0)) {
GST_BUFFER_SIZE (mp3_buf) = mp3_size;
result = gst_audio_encoder_finish_frame (enc, mp3_buf, -1);
} 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 (mp3_size < 0) {
/* eat error ? */
g_warning ("error %d", mp3_size);
}
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;
} else {
g_free (mp3_data);
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 */
static gboolean
gst_lamemp3enc_setup (GstLameMP3Enc * lame, GstTagList ** tags)
{
gboolean res;
#define CHECK_ERROR(command) G_STMT_START {\
if ((command) < 0) { \
@ -877,14 +628,6 @@ gst_lamemp3enc_setup (GstLameMP3Enc * lame, GstTagList ** tags)
GST_DEBUG_OBJECT (lame, "starting setup");
/* check if we're already setup; if we are, we might want to check
* if this initialization is compatible with the previous one */
/* FIXME: do this */
if (lame->setup) {
GST_WARNING_OBJECT (lame, "already setup");
lame->setup = FALSE;
}
lame->lgf = lame_init ();
if (lame->lgf == NULL)
@ -892,15 +635,11 @@ gst_lamemp3enc_setup (GstLameMP3Enc * lame, GstTagList ** tags)
*tags = gst_tag_list_new ();
/* post latency message on the bus */
gst_element_post_message (GST_ELEMENT (lame),
gst_message_new_latency (GST_OBJECT (lame)));
/* copy the parameters over */
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;
@ -954,53 +693,22 @@ gst_lamemp3enc_setup (GstLameMP3Enc * lame, GstTagList ** tags)
/* initialize the lame encoder */
if ((retval = lame_init_params (lame->lgf)) >= 0) {
lame->setup = TRUE;
/* FIXME: it would be nice to print out the mode here */
GST_INFO
("lame encoder setup (target %s, quality %f, bitrate %d, %d Hz, %d channels)",
(lame->target == LAMEMP3ENC_TARGET_QUALITY) ? "quality" : "bitrate",
lame->quality, lame->bitrate, lame->samplerate, lame->num_channels);
res = TRUE;
} else {
GST_ERROR_OBJECT (lame, "lame_init_params returned %d", retval);
res = FALSE;
}
GST_DEBUG_OBJECT (lame, "done with setup");
return lame->setup;
return res;
#undef CHECK_ERROR
}
static GstStateChangeReturn
gst_lamemp3enc_change_state (GstElement * element, GstStateChange transition)
{
GstLameMP3Enc *lame;
GstStateChangeReturn result;
lame = GST_LAMEMP3ENC (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_lamemp3enc_release_memory (lame);
break;
default:
break;
}
return result;
}
gboolean
gst_lamemp3enc_register (GstPlugin * plugin)
{

View file

@ -24,6 +24,7 @@
#include <gst/gst.h>
#include <gst/audio/gstaudioencoder.h>
G_BEGIN_DECLS
@ -49,37 +50,25 @@ typedef struct _GstLameMP3EncClass GstLameMP3EncClass;
* Opaque data structure.
*/
struct _GstLameMP3Enc {
GstElement element;
GstAudioEncoder element;
/*< private >*/
GstPad *srcpad, *sinkpad;
gint samplerate;
gint num_channels;
gboolean setup;
/* properties */
gint target;
gint bitrate;
gboolean cbr;
gfloat quality;
gint encoding_engine_quality;
gboolean mono;
/* track this so we don't send a last buffer in eos handler after error */
GstFlowReturn last_flow;
lame_global_flags *lgf;
/* time tracker */
guint64 last_ts, last_offs, last_duration, eos_ts;
};
struct _GstLameMP3EncClass {
GstElementClass parent_class;
GstAudioEncoderClass parent_class;
};
GType gst_lamemp3enc_get_type(void);