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gstreamer/ext/lame/gstlamemp3enc.c
Sebastian Dröge c286272792 lamemp3enc: Improve debugging a bit
2009-05-14 12:32:54 +02:00

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2004> Wim Taymans <wim@fluendo.com>
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
* Copyright (C) <2009> Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-lamemp3enc
* @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
* a free format, there are licensing and patent issues to take into
* consideration. See <ulink url="http://www.vorbis.com/">Ogg/Vorbis</ulink>
* for a royalty free (and often higher quality) alternative.
*
* <refsect2>
* <title>Output sample rate</title>
* If no fixed output sample rate is negotiated on the element's src pad,
* the element will choose an optimal sample rate to resample to internally.
* For example, a 16-bit 44.1 KHz mono audio stream encoded at 48 kbit will
* get resampled to 32 KHz. Use filter caps on the src pad to force a
* particular sample rate.
* </refsect2>
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lamemp3enc ! filesink location=sine.mp3
* ]| Encode a test sine signal to MP3.
* |[
* gst-launch -v alsasrc ! audioconvert ! lamemp3enc target=bitrate bitrate=192 ! filesink location=alsasrc.mp3
* ]| Record from a sound card using ALSA and encode to MP3 with an average bitrate of 192kbps
* |[
* gst-launch -v filesrc location=music.wav ! decodebin ! audioconvert ! audioresample ! lamemp3enc target=quality quality=0 ! id3v2mux ! filesink location=music.mp3
* ]| Transcode from a .wav file to MP3 (the id3v2mux element is optional) with best VBR quality
* |[
* gst-launch -v cdda://5 ! audioconvert ! lamemp3enc target=bitrate cbr=true bitrate=192 ! filesink location=track5.mp3
* ]| Encode Audio CD track 5 to MP3 with a constant bitrate of 192kbps
* |[
* gst-launch -v audiotestsrc num-buffers=10 ! audio/x-raw-int,rate=44100,channels=1 ! lamemp3enc target=bitrate cbr=true bitrate=48 ! filesink location=test.mp3
* ]| Encode to a fixed sample rate
* </refsect2>
*
* Since: 0.10.12
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstlamemp3enc.h"
#include <gst/gst-i18n-plugin.h>
/* lame < 3.98 */
#ifndef HAVE_LAME_SET_VBR_QUALITY
#define lame_set_VBR_quality(flags,q) lame_set_VBR_q((flags),(int)(q))
#endif
GST_DEBUG_CATEGORY_STATIC (debug);
#define GST_CAT_DEFAULT debug
/* elementfactory information */
/* LAMEMP3ENC can do MPEG-1, MPEG-2, and MPEG-2.5, so it has 9 possible
* sample rates it supports */
static GstStaticPadTemplate gst_lamemp3enc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
"signed = (boolean) true, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
"channels = (int) [ 1, 2 ]")
);
static GstStaticPadTemplate gst_lamemp3enc_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, "
"mpegversion = (int) 1, "
"layer = (int) 3, "
"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
"channels = (int) [ 1, 2 ]")
);
/********** Define useful types for non-programmatic interfaces **********/
enum
{
LAMEMP3ENC_TARGET_QUALITY = 0,
LAMEMP3ENC_TARGET_BITRATE
};
#define GST_TYPE_LAMEMP3ENC_TARGET (gst_lamemp3enc_target_get_type())
static GType
gst_lamemp3enc_target_get_type (void)
{
static GType lame_target_type = 0;
static GEnumValue lame_targets[] = {
{LAMEMP3ENC_TARGET_QUALITY, "Quality", "quality"},
{LAMEMP3ENC_TARGET_BITRATE, "Bitrate", "bitrate"},
{0, NULL, NULL}
};
if (!lame_target_type) {
lame_target_type =
g_enum_register_static ("GstLameMP3EncTarget", lame_targets);
}
return lame_target_type;
}
enum
{
LAMEMP3ENC_ENCODING_ENGINE_QUALITY_FAST = 0,
LAMEMP3ENC_ENCODING_ENGINE_QUALITY_STANDARD,
LAMEMP3ENC_ENCODING_ENGINE_QUALITY_HIGH
};
#define GST_TYPE_LAMEMP3ENC_ENCODING_ENGINE_QUALITY (gst_lamemp3enc_encoding_engine_quality_get_type())
static GType
gst_lamemp3enc_encoding_engine_quality_get_type (void)
{
static GType lame_encoding_engine_quality_type = 0;
static GEnumValue lame_encoding_engine_quality[] = {
{0, "Fast", "fast"},
{1, "Standard", "standard"},
{2, "High", "high"},
{0, NULL, NULL}
};
if (!lame_encoding_engine_quality_type) {
lame_encoding_engine_quality_type =
g_enum_register_static ("GstLameMP3EncEncodingEngineQuality",
lame_encoding_engine_quality);
}
return lame_encoding_engine_quality_type;
}
/********** Standard stuff for signals and arguments **********/
enum
{
ARG_0,
ARG_TARGET,
ARG_BITRATE,
ARG_CBR,
ARG_QUALITY,
ARG_ENCODING_ENGINE_QUALITY,
ARG_MONO
};
#define DEFAULT_TARGET LAMEMP3ENC_TARGET_QUALITY
#define DEFAULT_BITRATE 128
#define DEFAULT_CBR FALSE
#define DEFAULT_QUALITY 4
#define DEFAULT_ENCODING_ENGINE_QUALITY LAMEMP3ENC_ENCODING_ENGINE_QUALITY_STANDARD
#define DEFAULT_MONO FALSE
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 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);
static GstStateChangeReturn gst_lamemp3enc_change_state (GstElement * element,
GstStateChange transition);
static GstElementClass *parent_class = NULL;
GType
gst_lamemp3enc_get_type (void)
{
static GType gst_lamemp3enc_type = 0;
if (!gst_lamemp3enc_type) {
static const GTypeInfo gst_lamemp3enc_info = {
sizeof (GstLameMP3EncClass),
gst_lamemp3enc_base_init,
NULL,
(GClassInitFunc) gst_lamemp3enc_class_init,
NULL,
NULL,
sizeof (GstLameMP3Enc),
0,
(GInstanceInitFunc) gst_lamemp3enc_init,
};
static const GInterfaceInfo preset_info = {
NULL,
NULL,
NULL
};
gst_lamemp3enc_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstLameMP3Enc",
&gst_lamemp3enc_info, 0);
g_type_add_interface_static (gst_lamemp3enc_type, GST_TYPE_PRESET,
&preset_info);
}
return gst_lamemp3enc_type;
}
static void
gst_lamemp3enc_release_memory (GstLameMP3Enc * lame)
{
if (lame->lgf) {
lame_close (lame->lgf);
lame->lgf = NULL;
}
}
static void
gst_lamemp3enc_finalize (GObject * obj)
{
gst_lamemp3enc_release_memory (GST_LAMEMP3ENC (obj));
G_OBJECT_CLASS (parent_class)->finalize (obj);
}
static void
gst_lamemp3enc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_lamemp3enc_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_lamemp3enc_sink_template));
gst_element_class_set_details_simple (element_class, "L.A.M.E. mp3 encoder",
"Codec/Encoder/Audio",
"High-quality free MP3 encoder",
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
}
static void
gst_lamemp3enc_class_init (GstLameMP3EncClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property = gst_lamemp3enc_set_property;
gobject_class->get_property = gst_lamemp3enc_get_property;
gobject_class->finalize = gst_lamemp3enc_finalize;
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_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_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_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_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_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MONO,
g_param_spec_boolean ("mono", "Mono", "Enforce mono encoding",
DEFAULT_MONO, G_PARAM_READWRITE));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_lamemp3enc_change_state);
}
static gboolean
gst_lamemp3enc_src_setcaps (GstPad * pad, GstCaps * caps)
{
GST_DEBUG_OBJECT (pad, "caps: %" GST_PTR_FORMAT, caps);
return TRUE;
}
static gboolean
gst_lamemp3enc_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstLameMP3Enc *lame;
gint out_samplerate;
gint version;
GstStructure *structure;
GstCaps *othercaps;
lame = GST_LAMEMP3ENC (GST_PAD_PARENT (pad));
structure = gst_caps_get_structure (caps, 0);
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;
GST_DEBUG_OBJECT (lame, "setting up lame");
if (!gst_lamemp3enc_setup (lame))
goto setup_failed;
out_samplerate = lame_get_out_samplerate (lame->lgf);
if (out_samplerate == 0)
goto zero_output_rate;
if (out_samplerate != lame->samplerate) {
GST_WARNING_OBJECT (lame,
"output samplerate %d is different from incoming samplerate %d",
out_samplerate, lame->samplerate);
}
version = lame_get_version (lame->lgf);
if (version == 0)
version = 2;
else if (version == 1)
version = 1;
else if (version == 2)
version = 3;
othercaps =
gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 1,
"mpegaudioversion", G_TYPE_INT, version,
"layer", G_TYPE_INT, 3,
"channels", G_TYPE_INT, lame->mono ? 1 : lame->num_channels,
"rate", G_TYPE_INT, out_samplerate, NULL);
/* and use these caps */
gst_pad_set_caps (lame->srcpad, othercaps);
gst_caps_unref (othercaps);
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"));
return FALSE;
}
setup_failed:
{
GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS,
(_("Failed to configure LAMEMP3ENC encoder. Check your encoding parameters.")), (NULL));
return FALSE;
}
}
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_element_add_pad (GST_ELEMENT (lame), lame->sinkpad);
lame->srcpad =
gst_pad_new_from_static_template (&gst_lamemp3enc_src_template, "src");
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
* of actually having a pipeline and bus to get the message through */
#define CHECK_AND_FIXUP_BITRATE(obj,param,rate) \
G_STMT_START { \
gint ___rate = rate; \
gint maxrate = 320; \
gint multiplier = 64; \
if (rate == 0) { \
___rate = rate; \
} else if (rate <= 64) { \
maxrate = 64; multiplier = 8; \
if ((rate % 8) != 0) ___rate = GST_ROUND_UP_8 (rate); \
} else if (rate <= 128) { \
maxrate = 128; multiplier = 16; \
if ((rate % 16) != 0) ___rate = GST_ROUND_UP_16 (rate); \
} else if (rate <= 256) { \
maxrate = 256; multiplier = 32; \
if ((rate % 32) != 0) ___rate = GST_ROUND_UP_32 (rate); \
} else if (rate <= 320) { \
maxrate = 320; multiplier = 64; \
if ((rate % 64) != 0) ___rate = GST_ROUND_UP_64 (rate); \
} \
if (___rate != rate) { \
GST_ELEMENT_WARNING (obj, LIBRARY, SETTINGS, \
(_("The requested bitrate %d kbit/s for property '%s' " \
"is not allowed. " \
"The bitrate was changed to %d kbit/s."), rate, \
param, ___rate), \
("A bitrate below %d should be a multiple of %d.", \
maxrate, multiplier)); \
rate = ___rate; \
} \
} G_STMT_END
static void
gst_lamemp3enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstLameMP3Enc *lame;
lame = GST_LAMEMP3ENC (object);
switch (prop_id) {
case ARG_TARGET:
lame->target = g_value_get_enum (value);
break;
case ARG_BITRATE:
lame->bitrate = g_value_get_int (value);
break;
case ARG_CBR:
lame->cbr = g_value_get_boolean (value);
break;
case ARG_QUALITY:
lame->quality = g_value_get_float (value);
break;
case ARG_ENCODING_ENGINE_QUALITY:
lame->encoding_engine_quality = g_value_get_enum (value);
break;
case ARG_MONO:
lame->mono = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_lamemp3enc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstLameMP3Enc *lame;
lame = GST_LAMEMP3ENC (object);
switch (prop_id) {
case ARG_TARGET:
g_value_set_enum (value, lame->target);
break;
case ARG_BITRATE:
g_value_set_int (value, lame->bitrate);
break;
case ARG_CBR:
g_value_set_boolean (value, lame->cbr);
break;
case ARG_QUALITY:
g_value_set_float (value, lame->quality);
break;
case ARG_ENCODING_ENGINE_QUALITY:
g_value_set_enum (value, lame->encoding_engine_quality);
break;
case ARG_MONO:
g_value_set_boolean (value, lame->mono);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_lamemp3enc_sink_event (GstPad * pad, GstEvent * event)
{
gboolean ret;
GstLameMP3Enc *lame;
lame = GST_LAMEMP3ENC (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:{
GST_DEBUG_OBJECT (lame, "handling EOS event");
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");
/* clear buffers */
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;
}
gst_object_unref (lame);
return ret;
}
static GstFlowReturn
gst_lamemp3enc_chain (GstPad * pad, GstBuffer * buf)
{
GstLameMP3Enc *lame;
guchar *mp3_data;
gint mp3_buffer_size, mp3_size;
gint64 duration;
GstFlowReturn result;
gint num_samples;
guint8 *data;
guint size;
lame = GST_LAMEMP3ENC (GST_PAD_PARENT (pad));
GST_LOG_OBJECT (lame, "entered chain");
if (!lame->setup)
goto not_setup;
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
num_samples = size / 2;
/* allocate space for output */
mp3_buffer_size = 1.25 * num_samples + 7200;
mp3_data = g_malloc (mp3_buffer_size);
/* lame seems to be too stupid to get mono interleaved going */
if (lame->num_channels == 1) {
mp3_size = lame_encode_buffer (lame->lgf,
(short int *) data,
(short int *) data, num_samples, mp3_data, mp3_buffer_size);
} else {
mp3_size = lame_encode_buffer_interleaved (lame->lgf,
(short int *) data,
num_samples / lame->num_channels, mp3_data, mp3_buffer_size);
}
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;
} else {
g_free (mp3_data);
result = GST_FLOW_OK;
}
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)
{
#define CHECK_ERROR(command) G_STMT_START {\
if ((command) < 0) { \
GST_ERROR_OBJECT (lame, "setup failed: " G_STRINGIFY (command)); \
return FALSE; \
} \
}G_STMT_END
int retval;
GstCaps *allowed_caps;
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)
return FALSE;
/* 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);
if (allowed_caps != NULL) {
GstStructure *structure;
gint samplerate;
structure = gst_caps_get_structure (allowed_caps, 0);
if (gst_structure_get_int (structure, "rate", &samplerate)) {
GST_DEBUG_OBJECT (lame, "Setting sample rate to %d as fixed in src caps",
samplerate);
lame_set_out_samplerate (lame->lgf, samplerate);
} else {
GST_DEBUG_OBJECT (lame, "Letting lame choose sample rate");
lame_set_out_samplerate (lame->lgf, 0);
}
gst_caps_unref (allowed_caps);
allowed_caps = NULL;
} else {
GST_DEBUG_OBJECT (lame, "No peer yet, letting lame choose sample rate");
lame_set_out_samplerate (lame->lgf, 0);
}
CHECK_ERROR (lame_set_num_channels (lame->lgf, lame->num_channels));
CHECK_ERROR (lame_set_bWriteVbrTag (lame->lgf, 0));
if (lame->target == LAMEMP3ENC_TARGET_QUALITY) {
CHECK_ERROR (lame_set_VBR (lame->lgf, vbr_default));
CHECK_ERROR (lame_set_VBR_quality (lame->lgf, lame->quality));
} else {
if (lame->cbr) {
CHECK_AND_FIXUP_BITRATE (lame, "bitrate", lame->bitrate);
CHECK_ERROR (lame_set_VBR (lame->lgf, vbr_off));
CHECK_ERROR (lame_set_brate (lame->lgf, lame->bitrate));
} else {
CHECK_ERROR (lame_set_VBR (lame->lgf, vbr_abr));
CHECK_ERROR (lame_set_VBR_mean_bitrate_kbps (lame->lgf, lame->bitrate));
}
}
if (lame->encoding_engine_quality == LAMEMP3ENC_ENCODING_ENGINE_QUALITY_FAST)
CHECK_ERROR (lame_set_quality (lame->lgf, 7));
else if (lame->encoding_engine_quality ==
LAMEMP3ENC_ENCODING_ENGINE_QUALITY_HIGH)
CHECK_ERROR (lame_set_quality (lame->lgf, 2));
/* else default */
if (lame->mono)
CHECK_ERROR (lame_set_mode (lame->lgf, MONO));
/* 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);
} else {
GST_ERROR_OBJECT (lame, "lame_init_params returned %d", retval);
}
GST_DEBUG_OBJECT (lame, "done with setup");
return lame->setup;
#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)
{
GST_DEBUG_CATEGORY_INIT (debug, "lamemp3enc", 0, "lame mp3 encoder");
if (!gst_element_register (plugin, "lamemp3enc", GST_RANK_PRIMARY,
GST_TYPE_LAMEMP3ENC))
return FALSE;
return TRUE;
}
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