/* GStreamer * Copyright (C) <1999> Erik Walthinsen * Copyright (C) <2004> Wim Taymans * Copyright (C) <2005> Thomas Vander Stichele * Copyright (C) <2009> Sebastian Dröge * * 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 is not * a free format, there are licensing and patent issues to take into * consideration. See Ogg/Vorbis * 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 -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,rate=44100,channels=1 ! lamemp3enc target=bitrate cbr=true bitrate=48 ! filesink location=test.mp3 * ]| Encode to a fixed sample rate * * * Since: 0.10.12 */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include "gstlamemp3enc.h" #include /* 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 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 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_pad_template (gstelement_class, gst_static_pad_template_get (&gst_lamemp3enc_src_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&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 "); 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) { } 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; } } /* three underscores for ___rate is really really really * private as opposed to one underscore */ /* 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; 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; } /* 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, version == 1 ? 1152 : 576); } 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; }