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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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934 lines
30 KiB
C
934 lines
30 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2004> Wim Taymans <wim@fluendo.com>
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* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
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* Copyright (C) <2009> Sebastian Dröge <sebastian.droege@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-lamemp3enc
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* @see_also: lame, mad, vorbisenc
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*
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* This element encodes raw integer audio into an MPEG-1 layer 3 (MP3) stream.
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* Note that [MP3](http://en.wikipedia.org/wiki/MP3) is not
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* a free format, there are licensing and patent issues to take into
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* consideration. See [Ogg/Vorbis](http://www.vorbis.com/) for a royalty free
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* (and often higher quality) alternative.
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*
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* ## Output sample rate
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*
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* If no fixed output sample rate is negotiated on the element's src pad,
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* the element will choose an optimal sample rate to resample to internally.
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* For example, a 16-bit 44.1 KHz mono audio stream encoded at 48 kbit will
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* get resampled to 32 KHz. Use filter caps on the src pad to force a
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* particular sample rate.
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*
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* ## Example pipelines
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*
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* |[
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* gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lamemp3enc ! filesink location=sine.mp3
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* ]| Encode a test sine signal to MP3.
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* |[
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* gst-launch-1.0 -v autoaudiosrc ! audioconvert ! lamemp3enc target=bitrate bitrate=192 ! filesink location=alsasrc.mp3
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* ]| Record from a sound card using ALSA and encode to MP3 with an average bitrate of 192kbps
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* |[
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* gst-launch-1.0 -v filesrc location=music.wav ! decodebin ! audioconvert ! audioresample ! lamemp3enc target=quality quality=0 ! id3v2mux ! filesink location=music.mp3
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* ]| Transcode from a .wav file to MP3 (the id3v2mux element is optional) with best VBR quality
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* |[
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* gst-launch-1.0 -v cdda://5 ! audioconvert ! lamemp3enc target=bitrate cbr=true bitrate=192 ! filesink location=track5.mp3
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* ]| Encode Audio CD track 5 to MP3 with a constant bitrate of 192kbps
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* |[
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* 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
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* ]| Encode to a fixed sample rate
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "gstlamemp3enc.h"
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#include <gst/gst-i18n-plugin.h>
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/* lame < 3.98 */
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#ifndef HAVE_LAME_SET_VBR_QUALITY
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#define lame_set_VBR_quality(flags,q) lame_set_VBR_q((flags),(int)(q))
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#endif
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GST_DEBUG_CATEGORY_STATIC (debug);
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#define GST_CAT_DEFAULT debug
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/* elementfactory information */
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/* LAMEMP3ENC can do MPEG-1, MPEG-2, and MPEG-2.5, so it has 9 possible
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* sample rates it supports */
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static GstStaticPadTemplate gst_lamemp3enc_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
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"channels = (int) 1; "
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"audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
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"channels = (int) 2, " "channel-mask = (bitmask) 0x3")
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);
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static GstStaticPadTemplate gst_lamemp3enc_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/mpeg, "
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"mpegversion = (int) 1, "
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"layer = (int) 3, "
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"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
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"channels = (int) [ 1, 2 ]")
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);
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/********** Define useful types for non-programmatic interfaces **********/
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enum
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{
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LAMEMP3ENC_TARGET_QUALITY = 0,
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LAMEMP3ENC_TARGET_BITRATE
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};
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#define GST_TYPE_LAMEMP3ENC_TARGET (gst_lamemp3enc_target_get_type())
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static GType
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gst_lamemp3enc_target_get_type (void)
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{
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static GType lame_target_type = 0;
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static const GEnumValue lame_targets[] = {
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{LAMEMP3ENC_TARGET_QUALITY, "Quality", "quality"},
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{LAMEMP3ENC_TARGET_BITRATE, "Bitrate", "bitrate"},
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{0, NULL, NULL}
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};
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if (!lame_target_type) {
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lame_target_type =
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g_enum_register_static ("GstLameMP3EncTarget", lame_targets);
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}
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return lame_target_type;
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}
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enum
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{
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LAMEMP3ENC_ENCODING_ENGINE_QUALITY_FAST = 0,
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LAMEMP3ENC_ENCODING_ENGINE_QUALITY_STANDARD,
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LAMEMP3ENC_ENCODING_ENGINE_QUALITY_HIGH
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};
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#define GST_TYPE_LAMEMP3ENC_ENCODING_ENGINE_QUALITY (gst_lamemp3enc_encoding_engine_quality_get_type())
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static GType
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gst_lamemp3enc_encoding_engine_quality_get_type (void)
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{
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static GType lame_encoding_engine_quality_type = 0;
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static const GEnumValue lame_encoding_engine_quality[] = {
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{0, "Fast", "fast"},
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{1, "Standard", "standard"},
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{2, "High", "high"},
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{0, NULL, NULL}
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};
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if (!lame_encoding_engine_quality_type) {
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lame_encoding_engine_quality_type =
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g_enum_register_static ("GstLameMP3EncEncodingEngineQuality",
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lame_encoding_engine_quality);
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}
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return lame_encoding_engine_quality_type;
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}
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/********** Standard stuff for signals and arguments **********/
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enum
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{
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ARG_0,
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ARG_TARGET,
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ARG_BITRATE,
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ARG_CBR,
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ARG_QUALITY,
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ARG_ENCODING_ENGINE_QUALITY,
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ARG_MONO
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};
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#define DEFAULT_TARGET LAMEMP3ENC_TARGET_QUALITY
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#define DEFAULT_BITRATE 128
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#define DEFAULT_CBR FALSE
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#define DEFAULT_QUALITY 4
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#define DEFAULT_ENCODING_ENGINE_QUALITY LAMEMP3ENC_ENCODING_ENGINE_QUALITY_STANDARD
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#define DEFAULT_MONO FALSE
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static gboolean gst_lamemp3enc_start (GstAudioEncoder * enc);
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static gboolean gst_lamemp3enc_stop (GstAudioEncoder * enc);
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static gboolean gst_lamemp3enc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_lamemp3enc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static void gst_lamemp3enc_flush (GstAudioEncoder * enc);
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static void gst_lamemp3enc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_lamemp3enc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_lamemp3enc_setup (GstLameMP3Enc * lame, GstTagList ** tags);
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#define gst_lamemp3enc_parent_class parent_class
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G_DEFINE_TYPE (GstLameMP3Enc, gst_lamemp3enc, GST_TYPE_AUDIO_ENCODER);
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static void
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gst_lamemp3enc_release_memory (GstLameMP3Enc * lame)
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{
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if (lame->lgf) {
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lame_close (lame->lgf);
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lame->lgf = NULL;
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}
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}
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static void
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gst_lamemp3enc_finalize (GObject * obj)
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{
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gst_lamemp3enc_release_memory (GST_LAMEMP3ENC (obj));
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G_OBJECT_CLASS (parent_class)->finalize (obj);
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}
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static void
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gst_lamemp3enc_class_init (GstLameMP3EncClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioEncoderClass *base_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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base_class = (GstAudioEncoderClass *) klass;
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gobject_class->set_property = gst_lamemp3enc_set_property;
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gobject_class->get_property = gst_lamemp3enc_get_property;
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gobject_class->finalize = gst_lamemp3enc_finalize;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_lamemp3enc_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_lamemp3enc_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"L.A.M.E. mp3 encoder", "Codec/Encoder/Audio",
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"High-quality free MP3 encoder",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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base_class->start = GST_DEBUG_FUNCPTR (gst_lamemp3enc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_lamemp3enc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_lamemp3enc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_lamemp3enc_handle_frame);
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base_class->flush = GST_DEBUG_FUNCPTR (gst_lamemp3enc_flush);
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_TARGET,
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g_param_spec_enum ("target", "Target",
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"Optimize for quality or bitrate", GST_TYPE_LAMEMP3ENC_TARGET,
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DEFAULT_TARGET,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BITRATE,
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g_param_spec_int ("bitrate", "Bitrate (kb/s)",
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"Bitrate in kbit/sec (Only valid if target is bitrate, for CBR one "
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"of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, "
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"256 or 320)", 8, 320, DEFAULT_BITRATE,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_CBR,
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g_param_spec_boolean ("cbr", "CBR", "Enforce constant bitrate encoding "
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"(Only valid if target is bitrate)", DEFAULT_CBR,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_QUALITY,
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g_param_spec_float ("quality", "Quality",
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"VBR Quality from 0 to 10, 0 being the best "
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"(Only valid if target is quality)", 0.0, 9.999,
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DEFAULT_QUALITY,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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ARG_ENCODING_ENGINE_QUALITY, g_param_spec_enum ("encoding-engine-quality",
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"Encoding Engine Quality", "Quality/speed of the encoding engine, "
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"this does not affect the bitrate!",
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GST_TYPE_LAMEMP3ENC_ENCODING_ENGINE_QUALITY,
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DEFAULT_ENCODING_ENGINE_QUALITY,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MONO,
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g_param_spec_boolean ("mono", "Mono", "Enforce mono encoding",
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DEFAULT_MONO,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_type_mark_as_plugin_api (GST_TYPE_LAMEMP3ENC_TARGET, 0);
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gst_type_mark_as_plugin_api (GST_TYPE_LAMEMP3ENC_ENCODING_ENGINE_QUALITY, 0);
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}
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static void
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gst_lamemp3enc_init (GstLameMP3Enc * lame)
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{
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (lame));
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}
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static gboolean
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gst_lamemp3enc_start (GstAudioEncoder * enc)
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{
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GstLameMP3Enc *lame = GST_LAMEMP3ENC (enc);
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GST_DEBUG_OBJECT (lame, "start");
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if (!lame->adapter)
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lame->adapter = gst_adapter_new ();
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gst_adapter_clear (lame->adapter);
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return TRUE;
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}
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static gboolean
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gst_lamemp3enc_stop (GstAudioEncoder * enc)
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{
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GstLameMP3Enc *lame = GST_LAMEMP3ENC (enc);
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GST_DEBUG_OBJECT (lame, "stop");
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if (lame->adapter) {
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g_object_unref (lame->adapter);
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lame->adapter = NULL;
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}
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gst_lamemp3enc_release_memory (lame);
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return TRUE;
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}
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static gboolean
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gst_lamemp3enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
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{
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GstLameMP3Enc *lame;
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gint out_samplerate;
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gint version;
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GstCaps *othercaps;
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GstClockTime latency;
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GstTagList *tags = NULL;
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lame = GST_LAMEMP3ENC (enc);
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/* parameters already parsed for us */
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lame->samplerate = GST_AUDIO_INFO_RATE (info);
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lame->num_channels = GST_AUDIO_INFO_CHANNELS (info);
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/* but we might be asked to reconfigure, so reset */
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gst_lamemp3enc_release_memory (lame);
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GST_DEBUG_OBJECT (lame, "setting up lame");
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if (!gst_lamemp3enc_setup (lame, &tags))
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goto setup_failed;
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out_samplerate = lame_get_out_samplerate (lame->lgf);
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if (out_samplerate == 0)
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goto zero_output_rate;
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if (out_samplerate != lame->samplerate) {
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GST_WARNING_OBJECT (lame,
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"output samplerate %d is different from incoming samplerate %d",
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out_samplerate, lame->samplerate);
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}
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lame->out_samplerate = out_samplerate;
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version = lame_get_version (lame->lgf);
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if (version == 0)
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version = 2;
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else if (version == 1)
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version = 1;
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else if (version == 2)
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version = 3;
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othercaps =
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gst_caps_new_simple ("audio/mpeg",
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"mpegversion", G_TYPE_INT, 1,
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"mpegaudioversion", G_TYPE_INT, version,
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"layer", G_TYPE_INT, 3,
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"channels", G_TYPE_INT, lame->mono ? 1 : lame->num_channels,
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"rate", G_TYPE_INT, out_samplerate, NULL);
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/* and use these caps */
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gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), othercaps);
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gst_caps_unref (othercaps);
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/* base class feedback:
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* - we will handle buffers, just hand us all available
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* - report latency */
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latency = gst_util_uint64_scale_int (lame_get_framesize (lame->lgf),
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GST_SECOND, lame->samplerate);
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gst_audio_encoder_set_latency (enc, latency, latency);
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if (tags) {
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gst_audio_encoder_merge_tags (enc, tags, GST_TAG_MERGE_REPLACE);
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gst_tag_list_unref (tags);
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}
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return TRUE;
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zero_output_rate:
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{
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if (tags)
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gst_tag_list_unref (tags);
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GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS, (NULL),
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("LAME mp3 audio decided on a zero sample rate"));
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return FALSE;
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}
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setup_failed:
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{
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GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS,
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(_("Failed to configure LAME mp3 audio encoder. Check your encoding parameters.")), (NULL));
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return FALSE;
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}
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}
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/* <php-emulation-mode>three underscores for ___rate is really really really
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* private as opposed to one underscore<php-emulation-mode> */
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/* call this MACRO outside of the NULL state so that we have a higher chance
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* of actually having a pipeline and bus to get the message through */
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#define CHECK_AND_FIXUP_BITRATE(obj,param,rate) \
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G_STMT_START { \
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gint ___rate = rate; \
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gint maxrate = 320; \
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gint multiplier = 64; \
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if (rate == 0) { \
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___rate = rate; \
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} else if (rate <= 64) { \
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maxrate = 64; multiplier = 8; \
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if ((rate % 8) != 0) ___rate = GST_ROUND_UP_8 (rate); \
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} else if (rate <= 128) { \
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maxrate = 128; multiplier = 16; \
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if ((rate % 16) != 0) ___rate = GST_ROUND_UP_16 (rate); \
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} else if (rate <= 256) { \
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maxrate = 256; multiplier = 32; \
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if ((rate % 32) != 0) ___rate = GST_ROUND_UP_32 (rate); \
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} else if (rate <= 320) { \
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maxrate = 320; multiplier = 64; \
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if ((rate % 64) != 0) ___rate = GST_ROUND_UP_64 (rate); \
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} \
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if (___rate != rate) { \
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GST_ELEMENT_WARNING (obj, LIBRARY, SETTINGS, \
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(_("The requested bitrate %d kbit/s for property '%s' " \
|
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"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;
|
|
}
|