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-rw-r--r--ext/lame/Makefile.am10
-rw-r--r--ext/lame/gstlamemp3enc.c934
-rw-r--r--ext/lame/gstlamemp3enc.h83
-rw-r--r--ext/lame/meson.build19
-rw-r--r--ext/lame/plugin.c44
5 files changed, 1090 insertions, 0 deletions
diff --git a/ext/lame/Makefile.am b/ext/lame/Makefile.am
new file mode 100644
index 000000000..5e5b3a02d
--- /dev/null
+++ b/ext/lame/Makefile.am
@@ -0,0 +1,10 @@
+plugin_LTLIBRARIES = libgstlame.la
+
+libgstlame_la_SOURCES = gstlamemp3enc.c plugin.c
+libgstlame_la_CFLAGS = \
+ $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LAME_CFLAGS)
+libgstlame_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) \
+ $(GST_BASE_LIBS) $(GST_LIBS) $(LAME_LIBS)
+libgstlame_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+noinst_HEADERS = gstlamemp3enc.h
diff --git a/ext/lame/gstlamemp3enc.c b/ext/lame/gstlamemp3enc.c
new file mode 100644
index 000000000..a4a637ad4
--- /dev/null
+++ b/ext/lame/gstlamemp3enc.c
@@ -0,0 +1,934 @@
+/* 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 <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-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
+ * </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, "
+ "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
+G_DEFINE_TYPE (GstLameMP3Enc, gst_lamemp3enc, GST_TYPE_AUDIO_ENCODER);
+
+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));
+}
+
+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
+}
+
+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;
+}
diff --git a/ext/lame/gstlamemp3enc.h b/ext/lame/gstlamemp3enc.h
new file mode 100644
index 000000000..25dbc46f8
--- /dev/null
+++ b/ext/lame/gstlamemp3enc.h
@@ -0,0 +1,83 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * 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.
+ */
+
+
+#ifndef __GST_LAMEMP3ENC_H__
+#define __GST_LAMEMP3ENC_H__
+
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudioencoder.h>
+#include <gst/base/gstadapter.h>
+
+G_BEGIN_DECLS
+
+#include <lame/lame.h>
+
+#define GST_TYPE_LAMEMP3ENC \
+ (gst_lamemp3enc_get_type())
+#define GST_LAMEMP3ENC(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_LAMEMP3ENC,GstLameMP3Enc))
+#define GST_LAMEMP3ENC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_LAMEMP3ENC,GstLameMP3EncClass))
+#define GST_IS_LAMEMP3ENC(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_LAMEMP3ENC))
+#define GST_IS_LAMEMP3ENC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_LAMEMP3ENC))
+
+typedef struct _GstLameMP3Enc GstLameMP3Enc;
+typedef struct _GstLameMP3EncClass GstLameMP3EncClass;
+
+/**
+ * GstLameMP3Enc:
+ *
+ * Opaque data structure.
+ */
+struct _GstLameMP3Enc {
+ GstAudioEncoder element;
+
+ /*< private >*/
+ gint samplerate;
+ gint out_samplerate;
+ gint num_channels;
+
+ /* properties */
+ gint target;
+ gint bitrate;
+ gboolean cbr;
+ gfloat quality;
+ gint encoding_engine_quality;
+ gboolean mono;
+
+ lame_global_flags *lgf;
+
+ GstAdapter *adapter;
+};
+
+struct _GstLameMP3EncClass {
+ GstAudioEncoderClass parent_class;
+};
+
+GType gst_lamemp3enc_get_type(void);
+gboolean gst_lamemp3enc_register (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_LAMEMP3ENC_H__ */
diff --git a/ext/lame/meson.build b/ext/lame/meson.build
new file mode 100644
index 000000000..2812cb930
--- /dev/null
+++ b/ext/lame/meson.build
@@ -0,0 +1,19 @@
+lame_dep = cc.find_library('mp3lame', required : false)
+
+if lame_dep.found() and cc.has_header_symbol('lame/lame.h', 'lame_init')
+ lame_extra_c_args = []
+ if cc.has_header_symbol('lame/lame.h', 'lame_set_VBR_quality')
+ lame_extra_c_args += ['-DHAVE_LAME_SET_VBR_QUALITY']
+ endif
+ if cc.has_header_symbol('lame/lame.h', 'MEDIUM')
+ lame_extra_c_args += ['-DGSTLAME_PRESET']
+ endif
+ lame = library('gstlame',
+ ['gstlamemp3enc.c', 'plugin.c'],
+ c_args : ugly_args + lame_extra_c_args,
+ include_directories : [configinc, libsinc],
+ dependencies : [gstaudio_dep, lame_dep],
+ install : true,
+ install_dir : plugins_install_dir,
+ )
+endif
diff --git a/ext/lame/plugin.c b/ext/lame/plugin.c
new file mode 100644
index 000000000..6aae21ee3
--- /dev/null
+++ b/ext/lame/plugin.c
@@ -0,0 +1,44 @@
+/* GStreamer
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/gst-i18n-plugin.h>
+
+#include "gstlamemp3enc.h"
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+#ifdef ENABLE_NLS
+ bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
+ bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
+#endif /* ENABLE_NLS */
+
+ return gst_lamemp3enc_register (plugin);
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ lame,
+ "Encode MP3s with LAME",
+ plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-18 01:45:40 +0000