mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-30 05:31:15 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-plugins-good/ext/lame/gstlamemp3enc.c
Xavier Claessens b99ecc78ca Replace gst-i18n-*.h with gi18n-lib.h 
GLib guarantees libintl is always present, using proxy-libintl as
last resort. There is no need to mock gettex API any more.

This fix static build on Windows because G_INTL_STATIC_COMPILATION must
be defined before including libintl.h, and glib does it for us as part
as including glib.h.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
2022-04-19 18:01:06 +00:00

934 lines
30 KiB
C
Raw Blame History

/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, 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 [MP3](http://en.wikipedia.org/wiki/MP3) is not
* a free format, there are licensing and patent issues to take into
* consideration. See [Ogg/Vorbis](http://www.vorbis.com/) for a royalty free
* (and often higher quality) alternative.
*
* ## Output sample rate
*
* 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.
*
* ## Example pipelines
*
* |[
* gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lamemp3enc ! filesink location=sine.mp3
* ]| Encode a test sine signal to MP3.
* |[
* gst-launch-1.0 -v autoaudiosrc ! 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-1.0 -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-1.0 -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-1.0 -v audiotestsrc num-buffers=10 ! audio/x-raw,rate=44100,channels=1 ! lamemp3enc target=bitrate cbr=true bitrate=48 ! filesink location=test.mp3
* ]| Encode to a fixed sample rate
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstlamemp3enc.h"
#include <glib/gi18n-lib.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, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
"channels = (int) 1; "
"audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
"channels = (int) 2, " "channel-mask = (bitmask) 0x3")
);
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 const 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 const 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 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_setup (GstLameMP3Enc * lame, GstTagList ** tags);
#define gst_lamemp3enc_parent_class parent_class
#ifdef ENABLE_NLS
#define _do_init \
GST_DEBUG_CATEGORY_INIT (debug, "lamemp3enc", 0, "lame mp3 encoder");\
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);\
bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
#else /* ENABLE_NLS */
#define _do_init \
GST_DEBUG_CATEGORY_INIT (debug, "lamemp3enc", 0, "lame mp3 encoder");
#endif
G_DEFINE_TYPE_WITH_CODE (GstLameMP3Enc, gst_lamemp3enc, GST_TYPE_AUDIO_ENCODER,
_do_init);
GST_ELEMENT_REGISTER_DEFINE (lamemp3enc, "lamemp3enc", GST_RANK_PRIMARY,
GST_TYPE_LAMEMP3ENC);
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_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;
gobject_class->set_property = gst_lamemp3enc_set_property;
gobject_class->get_property = gst_lamemp3enc_get_property;
gobject_class->finalize = gst_lamemp3enc_finalize;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_lamemp3enc_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_lamemp3enc_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"L.A.M.E. mp3 encoder", "Codec/Encoder/Audio",
"High-quality free MP3 encoder",
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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_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_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_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_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_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_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_type_mark_as_plugin_api (GST_TYPE_LAMEMP3ENC_TARGET, 0);
gst_type_mark_as_plugin_api (GST_TYPE_LAMEMP3ENC_ENCODING_ENGINE_QUALITY, 0);
}
static void
gst_lamemp3enc_init (GstLameMP3Enc * lame)
{
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (lame));
}
static gboolean
gst_lamemp3enc_start (GstAudioEncoder * enc)
{
GstLameMP3Enc *lame = GST_LAMEMP3ENC (enc);
GST_DEBUG_OBJECT (lame, "start");
if (!lame->adapter)
lame->adapter = gst_adapter_new ();
gst_adapter_clear (lame->adapter);
return TRUE;
}
static gboolean
gst_lamemp3enc_stop (GstAudioEncoder * enc)
{
GstLameMP3Enc *lame = GST_LAMEMP3ENC (enc);
GST_DEBUG_OBJECT (lame, "stop");
if (lame->adapter) {
g_object_unref (lame->adapter);
lame->adapter = NULL;
}
gst_lamemp3enc_release_memory (lame);
return TRUE;
}
static gboolean
gst_lamemp3enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstLameMP3Enc *lame;
gint out_samplerate;
gint version;
GstCaps *othercaps;
GstClockTime latency;
GstTagList *tags = NULL;
lame = GST_LAMEMP3ENC (enc);
/* 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))
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);
}
lame->out_samplerate = out_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_audio_encoder_set_output_format (GST_AUDIO_ENCODER (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_audio_encoder_merge_tags (enc, tags, GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (tags);
}
return TRUE;
zero_output_rate:
{
if (tags)
gst_tag_list_unref (tags);
GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS, (NULL),
("LAME mp3 audio decided on a zero sample rate"));
return FALSE;
}
setup_failed:
{
GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS,
(_("Failed to configure LAME mp3 audio encoder. Check your encoding parameters.")), (NULL));
return FALSE;
}
}
/* <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;
}
}
/* **** credits go to mpegaudioparse **** */
static const guint mp3types_bitrates[2][3][16] = {
{
{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}
},
{
{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}
},
};
static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
{22050, 24000, 16000},
{11025, 12000, 8000}
};
static inline guint
mp3_type_frame_length_from_header (GstLameMP3Enc * lame, guint32 header,
guint * put_version, guint * put_layer, guint * put_channels,
guint * put_bitrate, guint * put_samplerate, guint * put_mode,
guint * put_crc)
{
guint length;
gulong mode, samplerate, bitrate, layer, channels, padding, crc;
gulong version;
gint lsf, mpg25;
if (header & (1 << 20)) {
lsf = (header & (1 << 19)) ? 0 : 1;
mpg25 = 0;
} else {
lsf = 1;
mpg25 = 1;
}
version = 1 + lsf + mpg25;
layer = 4 - ((header >> 17) & 0x3);
crc = (header >> 16) & 0x1;
bitrate = (header >> 12) & 0xF;
bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
/* The caller has ensured we have a valid header, so bitrate can't be
zero here. */
g_assert (bitrate != 0);
samplerate = (header >> 10) & 0x3;
samplerate = mp3types_freqs[lsf + mpg25][samplerate];
padding = (header >> 9) & 0x1;
mode = (header >> 6) & 0x3;
channels = (mode == 3) ? 1 : 2;
switch (layer) {
case 1:
length = 4 * ((bitrate * 12) / samplerate + padding);
break;
case 2:
length = (bitrate * 144) / samplerate + padding;
break;
default:
case 3:
length = (bitrate * 144) / (samplerate << lsf) + padding;
break;
}
GST_DEBUG_OBJECT (lame, "Calculated mp3 frame length of %u bytes", length);
GST_DEBUG_OBJECT (lame, "samplerate = %lu, bitrate = %lu, version = %lu, "
"layer = %lu, channels = %lu", samplerate, bitrate, version,
layer, channels);
if (put_version)
*put_version = version;
if (put_layer)
*put_layer = layer;
if (put_channels)
*put_channels = channels;
if (put_bitrate)
*put_bitrate = bitrate;
if (put_samplerate)
*put_samplerate = samplerate;
if (put_mode)
*put_mode = mode;
if (put_crc)
*put_crc = crc;
return length;
}
static gboolean
mp3_sync_check (GstLameMP3Enc * lame, unsigned long head)
{
GST_DEBUG_OBJECT (lame, "checking mp3 header 0x%08lx", head);
/* if it's not a valid sync */
if ((head & 0xffe00000) != 0xffe00000) {
GST_WARNING_OBJECT (lame, "invalid sync");
return FALSE;
}
/* if it's an invalid MPEG version */
if (((head >> 19) & 3) == 0x1) {
GST_WARNING_OBJECT (lame, "invalid MPEG version: 0x%lx", (head >> 19) & 3);
return FALSE;
}
/* if it's an invalid layer */
if (!((head >> 17) & 3)) {
GST_WARNING_OBJECT (lame, "invalid layer: 0x%lx", (head >> 17) & 3);
return FALSE;
}
/* if it's an invalid bitrate */
if (((head >> 12) & 0xf) == 0x0) {
GST_WARNING_OBJECT (lame, "invalid bitrate: 0x%lx."
"Free format files are not supported yet", (head >> 12) & 0xf);
return FALSE;
}
if (((head >> 12) & 0xf) == 0xf) {
GST_WARNING_OBJECT (lame, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
return FALSE;
}
/* if it's an invalid samplerate */
if (((head >> 10) & 0x3) == 0x3) {
GST_WARNING_OBJECT (lame, "invalid samplerate: 0x%lx", (head >> 10) & 0x3);
return FALSE;
}
if ((head & 0x3) == 0x2) {
/* Ignore this as there are some files with emphasis 0x2 that can
* be played fine. See BGO #537235 */
GST_WARNING_OBJECT (lame, "invalid emphasis: 0x%lx", head & 0x3);
}
return TRUE;
}
/* **** end mpegaudioparse **** */
static GstFlowReturn
gst_lamemp3enc_finish_frames (GstLameMP3Enc * lame)
{
gint av;
guint header;
GstFlowReturn result = GST_FLOW_OK;
/* limited parsing, we don't expect to lose sync here */
while ((result == GST_FLOW_OK) &&
((av = gst_adapter_available (lame->adapter)) > 4)) {
guint rate, version, layer, size;
GstBuffer *mp3_buf;
const guint8 *data;
guint samples_per_frame;
data = gst_adapter_map (lame->adapter, 4);
header = GST_READ_UINT32_BE (data);
gst_adapter_unmap (lame->adapter);
if (!mp3_sync_check (lame, header))
goto invalid_header;
size = mp3_type_frame_length_from_header (lame, header, &version, &layer,
NULL, NULL, &rate, NULL, NULL);
if (G_UNLIKELY (layer != 3 || rate != lame->out_samplerate)) {
GST_DEBUG_OBJECT (lame,
"unexpected mp3 header with rate %u, version %u, layer %u",
rate, version, layer);
goto invalid_header;
}
if (size > av) {
/* pretty likely to occur when lame is holding back on us */
GST_LOG_OBJECT (lame, "frame size %u (> %d)", size, av);
break;
}
/* Account for the internal resampling, finish frame really wants to
* know about the number of incoming samples
*/
samples_per_frame = (version == 1) ? 1152 : 576;
samples_per_frame *= lame->samplerate;
samples_per_frame /= lame->out_samplerate;
/* should be ok now */
mp3_buf = gst_adapter_take_buffer (lame->adapter, size);
/* number of samples for MPEG-1, layer 3 */
result = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame),
mp3_buf, samples_per_frame);
}
exit:
return result;
/* ERRORS */
invalid_header:
{
GST_ELEMENT_ERROR (lame, STREAM, ENCODE,
("invalid lame mp3 sync header %08X", header), (NULL));
result = GST_FLOW_ERROR;
goto exit;
}
}
static GstFlowReturn
gst_lamemp3enc_flush_full (GstLameMP3Enc * lame, gboolean push)
{
GstBuffer *buf;
GstMapInfo map;
gint size;
GstFlowReturn result = GST_FLOW_OK;
gint av;
if (!lame->lgf)
return GST_FLOW_OK;
buf = gst_buffer_new_and_alloc (7200);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
size = lame_encode_flush (lame->lgf, map.data, 7200);
if (size > 0) {
gst_buffer_unmap (buf, &map);
gst_buffer_resize (buf, 0, size);
GST_DEBUG_OBJECT (lame, "collecting final %d bytes", size);
gst_adapter_push (lame->adapter, buf);
} else {
gst_buffer_unmap (buf, &map);
GST_DEBUG_OBJECT (lame, "no final packet (size=%d, push=%d)", size, push);
gst_buffer_unref (buf);
result = GST_FLOW_OK;
}
if (push) {
result = gst_lamemp3enc_finish_frames (lame);
} else {
/* never mind */
gst_adapter_clear (lame->adapter);
}
/* either way, we expect nothing left */
if ((av = gst_adapter_available (lame->adapter))) {
/* should this be more fatal ?? */
GST_WARNING_OBJECT (lame, "unparsed %d bytes left after flushing", av);
/* clean up anyway */
gst_adapter_clear (lame->adapter);
}
return result;
}
static void
gst_lamemp3enc_flush (GstAudioEncoder * enc)
{
gst_lamemp3enc_flush_full (GST_LAMEMP3ENC (enc), FALSE);
}
static GstFlowReturn
gst_lamemp3enc_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
{
GstLameMP3Enc *lame;
gint mp3_buffer_size, mp3_size;
GstBuffer *mp3_buf;
GstFlowReturn result;
gint num_samples;
GstMapInfo in_map, mp3_map;
lame = GST_LAMEMP3ENC (enc);
/* squeeze remaining and push */
if (G_UNLIKELY (in_buf == NULL))
return gst_lamemp3enc_flush_full (lame, TRUE);
gst_buffer_map (in_buf, &in_map, GST_MAP_READ);
num_samples = in_map.size / 2;
/* allocate space for output */
mp3_buffer_size = 1.25 * num_samples + 7200;
mp3_buf = gst_buffer_new_allocate (NULL, mp3_buffer_size, NULL);
gst_buffer_map (mp3_buf, &mp3_map, GST_MAP_WRITE);
/* 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 *) in_map.data,
(short int *) in_map.data, num_samples, mp3_map.data, mp3_buffer_size);
} else {
mp3_size = lame_encode_buffer_interleaved (lame->lgf,
(short int *) in_map.data,
num_samples / lame->num_channels, mp3_map.data, mp3_buffer_size);
}
gst_buffer_unmap (in_buf, &in_map);
GST_LOG_OBJECT (lame, "encoded %" G_GSIZE_FORMAT " bytes of audio "
"to %d bytes of mp3", in_map.size, mp3_size);
if (G_LIKELY (mp3_size > 0)) {
/* unfortunately lame does not provide frame delineated output,
* so collect output and parse into frames ... */
gst_buffer_unmap (mp3_buf, &mp3_map);
gst_buffer_resize (mp3_buf, 0, mp3_size);
gst_adapter_push (lame->adapter, mp3_buf);
result = gst_lamemp3enc_finish_frames (lame);
} else {
gst_buffer_unmap (mp3_buf, &mp3_map);
if (mp3_size < 0) {
/* eat error ? */
g_warning ("error %d", mp3_size);
}
gst_buffer_unref (mp3_buf);
result = GST_FLOW_OK;
}
return result;
}
/* 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) { \
GST_ERROR_OBJECT (lame, "setup failed: " G_STRINGIFY (command)); \
if (*tags) { \
gst_tag_list_unref (*tags); \
*tags = NULL; \
} \
return FALSE; \
} \
}G_STMT_END
int retval;
GstCaps *allowed_caps;
GST_DEBUG_OBJECT (lame, "starting setup");
lame->lgf = lame_init ();
if (lame->lgf == NULL)
return FALSE;
*tags = gst_tag_list_new_empty ();
/* 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 (GST_AUDIO_ENCODER_SRC_PAD (lame));
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));
}
gst_tag_list_add (*tags, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
lame->bitrate * 1000, NULL);
}
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) {
/* 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 res;
#undef CHECK_ERROR
}
Reference in a new issue View git blame Copy permalink