| /* 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; |
| } |