gstreamer/ext/lame/gstlamemp3enc.c
Tim-Philipp Müller 7417ad6d5f lamemp3enc: implement sinkpad get_caps() function to proxy rate and channels restrictions from downstream
The element downstream of mp3enc might only accept certain sample rates or channels,
make sure we relay any restrictions that do exist to upstream when it does a
get_caps() on the sink pad. That way upstream elements like audioresample or
audioconvert can pick a sample rate / channel configuration that will be accepted,
instead of just negotiating to the highest, which might then be rejected.

https://bugzilla.gnome.org/show_bug.cgi?id=641151
2011-02-03 18:27:05 +00:00

979 lines
30 KiB
C

/* 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_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_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_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));
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_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));
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 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
* 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");
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;
}
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;
/* 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);
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;
}