mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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b0e7e27365
Also filter any CODEC/AUDIO_CODEC tags from incoming tag events. Fixes bug #391543.
1014 lines
31 KiB
C
1014 lines
31 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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 <ulink url="http://en.wikipedia.org/wiki/MP3">MP3</ulink> is not
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* a free format, there are licensing and patent issues to take into
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* consideration. See <ulink url="http://www.vorbis.com/">Ogg/Vorbis</ulink>
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* for a royalty free (and often higher quality) alternative.
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*
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* <refsect2>
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* <title>Output sample rate</title>
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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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* </refsect2>
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch -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 -v alsasrc ! 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 -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 -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 -v audiotestsrc num-buffers=10 ! audio/x-raw-int,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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* </refsect2>
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*
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* Since: 0.10.12
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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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#include <gst/pbutils/descriptions.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-int, "
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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
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"signed = (boolean) true, "
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"width = (int) 16, "
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"depth = (int) 16, "
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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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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 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 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 void gst_lamemp3enc_base_init (gpointer g_class);
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static void gst_lamemp3enc_class_init (GstLameMP3EncClass * klass);
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static void gst_lamemp3enc_init (GstLameMP3Enc * gst_lame);
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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_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_lamemp3enc_chain (GstPad * pad, GstBuffer * buf);
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static gboolean gst_lamemp3enc_setup (GstLameMP3Enc * lame, GstTagList ** tags);
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static GstStateChangeReturn gst_lamemp3enc_change_state (GstElement * element,
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GstStateChange transition);
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static GstElementClass *parent_class = NULL;
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GType
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gst_lamemp3enc_get_type (void)
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{
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static GType gst_lamemp3enc_type = 0;
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if (!gst_lamemp3enc_type) {
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static const GTypeInfo gst_lamemp3enc_info = {
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sizeof (GstLameMP3EncClass),
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gst_lamemp3enc_base_init,
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NULL,
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(GClassInitFunc) gst_lamemp3enc_class_init,
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NULL,
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NULL,
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sizeof (GstLameMP3Enc),
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0,
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(GInstanceInitFunc) gst_lamemp3enc_init,
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};
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static const GInterfaceInfo preset_info = {
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NULL,
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NULL,
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NULL
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};
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gst_lamemp3enc_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstLameMP3Enc",
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&gst_lamemp3enc_info, 0);
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g_type_add_interface_static (gst_lamemp3enc_type, GST_TYPE_PRESET,
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&preset_info);
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}
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return gst_lamemp3enc_type;
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}
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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_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_lamemp3enc_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_lamemp3enc_sink_template));
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gst_element_class_set_details_simple (element_class, "L.A.M.E. mp3 encoder",
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"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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}
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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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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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parent_class = g_type_class_peek_parent (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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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, 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_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_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, 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_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, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_lamemp3enc_change_state);
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}
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static gboolean
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gst_lamemp3enc_src_setcaps (GstPad * pad, GstCaps * caps)
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{
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GST_DEBUG_OBJECT (pad, "caps: %" GST_PTR_FORMAT, caps);
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return TRUE;
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}
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static gboolean
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gst_lamemp3enc_sink_setcaps (GstPad * pad, GstCaps * caps)
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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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GstStructure *structure;
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GstCaps *othercaps;
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GstTagList *tags = NULL;
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lame = GST_LAMEMP3ENC (GST_PAD_PARENT (pad));
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "rate", &lame->samplerate))
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goto no_rate;
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if (!gst_structure_get_int (structure, "channels", &lame->num_channels))
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goto no_channels;
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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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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_pad_set_caps (lame->srcpad, othercaps);
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if (tags) {
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gst_pb_utils_add_codec_description_to_tag_list (tags, GST_TAG_CODEC,
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othercaps);
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gst_pb_utils_add_codec_description_to_tag_list (tags, GST_TAG_AUDIO_CODEC,
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othercaps);
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}
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gst_caps_unref (othercaps);
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if (tags)
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gst_element_found_tags_for_pad (GST_ELEMENT_CAST (lame), lame->srcpad,
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tags);
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return TRUE;
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no_rate:
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{
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GST_ERROR_OBJECT (lame, "input caps have no sample rate field");
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return FALSE;
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}
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no_channels:
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{
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GST_ERROR_OBJECT (lame, "input caps have no channels field");
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return FALSE;
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}
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zero_output_rate:
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{
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GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS, (NULL),
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("LAMEMP3ENC decided on a zero sample rate"));
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if (tags)
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gst_tag_list_free (tags);
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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 LAMEMP3ENC encoder. Check your encoding parameters.")), (NULL));
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return FALSE;
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}
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}
|
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|
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static GstCaps *
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gst_lamemp3enc_sink_getcaps (GstPad * pad)
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{
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const GstCaps *templ_caps;
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GstLameMP3Enc *lame;
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GstCaps *allowed = NULL;
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GstCaps *caps, *filter_caps;
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gint i, j;
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lame = GST_LAMEMP3ENC (gst_pad_get_parent (pad));
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|
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/* we want to be able to communicate to upstream elements like audioconvert
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* and audioresample any rate/channel restrictions downstream (e.g. muxer
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* only accepting certain sample rates) */
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templ_caps = gst_pad_get_pad_template_caps (pad);
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allowed = gst_pad_get_allowed_caps (lame->srcpad);
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if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) {
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caps = gst_caps_copy (templ_caps);
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goto done;
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}
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|
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filter_caps = gst_caps_new_empty ();
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|
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for (i = 0; i < gst_caps_get_size (templ_caps); i++) {
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GQuark q_name;
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q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
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|
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/* pick rate + channel fields from allowed caps */
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for (j = 0; j < gst_caps_get_size (allowed); j++) {
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const GstStructure *allowed_s = gst_caps_get_structure (allowed, j);
|
|
const GValue *val;
|
|
GstStructure *s;
|
|
|
|
s = gst_structure_id_empty_new (q_name);
|
|
if ((val = gst_structure_get_value (allowed_s, "rate")))
|
|
gst_structure_set_value (s, "rate", val);
|
|
if ((val = gst_structure_get_value (allowed_s, "channels")))
|
|
gst_structure_set_value (s, "channels", val);
|
|
|
|
gst_caps_merge_structure (filter_caps, s);
|
|
}
|
|
}
|
|
|
|
caps = gst_caps_intersect (filter_caps, templ_caps);
|
|
gst_caps_unref (filter_caps);
|
|
|
|
done:
|
|
|
|
gst_caps_replace (&allowed, NULL);
|
|
gst_object_unref (lame);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static gint64
|
|
gst_lamemp3enc_get_latency (GstLameMP3Enc * lame)
|
|
{
|
|
return gst_util_uint64_scale_int (lame_get_framesize (lame->lgf),
|
|
GST_SECOND, lame->samplerate);
|
|
}
|
|
|
|
static gboolean
|
|
gst_lamemp3enc_src_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
gboolean res = TRUE;
|
|
GstLameMP3Enc *lame;
|
|
GstPad *peerpad;
|
|
|
|
lame = GST_LAMEMP3ENC (gst_pad_get_parent (pad));
|
|
peerpad = gst_pad_get_peer (GST_PAD (lame->sinkpad));
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
if ((res = gst_pad_query (peerpad, query))) {
|
|
gboolean live;
|
|
GstClockTime min_latency, max_latency;
|
|
gint64 latency;
|
|
|
|
if (lame->lgf == NULL)
|
|
break;
|
|
|
|
gst_query_parse_latency (query, &live, &min_latency, &max_latency);
|
|
|
|
latency = gst_lamemp3enc_get_latency (lame);
|
|
|
|
/* add our latency */
|
|
min_latency += latency;
|
|
if (max_latency != -1)
|
|
max_latency += latency;
|
|
|
|
gst_query_set_latency (query, live, min_latency, max_latency);
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query (peerpad, query);
|
|
break;
|
|
}
|
|
|
|
gst_object_unref (peerpad);
|
|
gst_object_unref (lame);
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_lamemp3enc_init (GstLameMP3Enc * lame)
|
|
{
|
|
GST_DEBUG_OBJECT (lame, "starting initialization");
|
|
|
|
lame->sinkpad =
|
|
gst_pad_new_from_static_template (&gst_lamemp3enc_sink_template, "sink");
|
|
gst_pad_set_event_function (lame->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_lamemp3enc_sink_event));
|
|
gst_pad_set_chain_function (lame->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_lamemp3enc_chain));
|
|
gst_pad_set_setcaps_function (lame->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_lamemp3enc_sink_setcaps));
|
|
gst_pad_set_getcaps_function (lame->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_lamemp3enc_sink_getcaps));
|
|
gst_element_add_pad (GST_ELEMENT (lame), lame->sinkpad);
|
|
|
|
lame->srcpad =
|
|
gst_pad_new_from_static_template (&gst_lamemp3enc_src_template, "src");
|
|
gst_pad_set_query_function (lame->srcpad,
|
|
GST_DEBUG_FUNCPTR (gst_lamemp3enc_src_query));
|
|
gst_pad_set_setcaps_function (lame->srcpad,
|
|
GST_DEBUG_FUNCPTR (gst_lamemp3enc_src_setcaps));
|
|
gst_element_add_pad (GST_ELEMENT (lame), lame->srcpad);
|
|
|
|
lame->samplerate = 44100;
|
|
lame->num_channels = 2;
|
|
lame->setup = FALSE;
|
|
|
|
/* Set default settings */
|
|
lame->target = DEFAULT_TARGET;
|
|
lame->bitrate = DEFAULT_BITRATE;
|
|
lame->cbr = DEFAULT_CBR;
|
|
lame->quality = DEFAULT_QUALITY;
|
|
lame->encoding_engine_quality = DEFAULT_ENCODING_ENGINE_QUALITY;
|
|
lame->mono = DEFAULT_MONO;
|
|
|
|
GST_DEBUG_OBJECT (lame, "done initializing");
|
|
}
|
|
|
|
/* <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;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_lamemp3enc_sink_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
gboolean ret;
|
|
GstLameMP3Enc *lame;
|
|
|
|
lame = GST_LAMEMP3ENC (gst_pad_get_parent (pad));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:{
|
|
GST_DEBUG_OBJECT (lame, "handling EOS event");
|
|
|
|
if (lame->lgf != NULL) {
|
|
GstBuffer *buf;
|
|
gint size;
|
|
|
|
buf = gst_buffer_new_and_alloc (7200);
|
|
size = lame_encode_flush (lame->lgf, GST_BUFFER_DATA (buf), 7200);
|
|
|
|
if (size > 0 && lame->last_flow == GST_FLOW_OK) {
|
|
gint64 duration;
|
|
|
|
duration = gst_util_uint64_scale (size, 8 * GST_SECOND,
|
|
1000 * lame->bitrate);
|
|
|
|
if (lame->last_ts == GST_CLOCK_TIME_NONE) {
|
|
lame->last_ts = lame->eos_ts;
|
|
lame->last_duration = duration;
|
|
} else {
|
|
lame->last_duration += duration;
|
|
}
|
|
|
|
GST_BUFFER_TIMESTAMP (buf) = lame->last_ts;
|
|
GST_BUFFER_DURATION (buf) = lame->last_duration;
|
|
lame->last_ts = GST_CLOCK_TIME_NONE;
|
|
GST_BUFFER_SIZE (buf) = size;
|
|
GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
|
|
gst_buffer_set_caps (buf, GST_PAD_CAPS (lame->srcpad));
|
|
gst_pad_push (lame->srcpad, buf);
|
|
} else {
|
|
GST_DEBUG_OBJECT (lame, "no final packet (size=%d, last_flow=%s)",
|
|
size, gst_flow_get_name (lame->last_flow));
|
|
gst_buffer_unref (buf);
|
|
}
|
|
}
|
|
|
|
ret = gst_pad_event_default (pad, event);
|
|
break;
|
|
}
|
|
case GST_EVENT_FLUSH_START:
|
|
GST_DEBUG_OBJECT (lame, "handling FLUSH start event");
|
|
/* forward event */
|
|
ret = gst_pad_push_event (lame->srcpad, event);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
{
|
|
guchar *mp3_data = NULL;
|
|
gint mp3_buffer_size;
|
|
|
|
GST_DEBUG_OBJECT (lame, "handling FLUSH stop event");
|
|
|
|
if (lame->lgf) {
|
|
/* clear buffers if we already have lame set up */
|
|
mp3_buffer_size = 7200;
|
|
mp3_data = g_malloc (mp3_buffer_size);
|
|
lame_encode_flush (lame->lgf, mp3_data, mp3_buffer_size);
|
|
g_free (mp3_data);
|
|
}
|
|
|
|
ret = gst_pad_push_event (lame->srcpad, event);
|
|
break;
|
|
}
|
|
case GST_EVENT_TAG:{
|
|
GstTagList *tags;
|
|
|
|
gst_event_parse_tag (event, &tags);
|
|
|
|
tags = gst_tag_list_copy (tags);
|
|
gst_event_unref (event);
|
|
|
|
gst_tag_list_remove_tag (tags, GST_TAG_CODEC);
|
|
gst_tag_list_remove_tag (tags, GST_TAG_AUDIO_CODEC);
|
|
event = gst_event_new_tag (tags);
|
|
|
|
ret = gst_pad_push_event (lame->srcpad, event);
|
|
break;
|
|
}
|
|
default:
|
|
ret = gst_pad_event_default (pad, event);
|
|
break;
|
|
}
|
|
gst_object_unref (lame);
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_lamemp3enc_chain (GstPad * pad, GstBuffer * buf)
|
|
{
|
|
GstLameMP3Enc *lame;
|
|
guchar *mp3_data;
|
|
gint mp3_buffer_size, mp3_size;
|
|
gint64 duration;
|
|
GstFlowReturn result;
|
|
gint num_samples;
|
|
guint8 *data;
|
|
guint size;
|
|
|
|
lame = GST_LAMEMP3ENC (GST_PAD_PARENT (pad));
|
|
|
|
GST_LOG_OBJECT (lame, "entered chain");
|
|
|
|
if (!lame->setup)
|
|
goto not_setup;
|
|
|
|
data = GST_BUFFER_DATA (buf);
|
|
size = GST_BUFFER_SIZE (buf);
|
|
|
|
num_samples = size / 2;
|
|
|
|
/* allocate space for output */
|
|
mp3_buffer_size = 1.25 * num_samples + 7200;
|
|
mp3_data = g_malloc (mp3_buffer_size);
|
|
|
|
/* 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 *) data,
|
|
(short int *) data, num_samples, mp3_data, mp3_buffer_size);
|
|
} else {
|
|
mp3_size = lame_encode_buffer_interleaved (lame->lgf,
|
|
(short int *) data,
|
|
num_samples / lame->num_channels, mp3_data, mp3_buffer_size);
|
|
}
|
|
|
|
GST_LOG_OBJECT (lame, "encoded %d bytes of audio to %d bytes of mp3",
|
|
size, mp3_size);
|
|
|
|
duration = gst_util_uint64_scale_int (size, GST_SECOND,
|
|
2 * lame->samplerate * lame->num_channels);
|
|
|
|
if (GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE &&
|
|
GST_BUFFER_DURATION (buf) != duration) {
|
|
GST_DEBUG_OBJECT (lame, "incoming buffer had incorrect duration %"
|
|
GST_TIME_FORMAT ", outgoing buffer will have correct duration %"
|
|
GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_TIME_ARGS (duration));
|
|
}
|
|
|
|
if (lame->last_ts == GST_CLOCK_TIME_NONE) {
|
|
lame->last_ts = GST_BUFFER_TIMESTAMP (buf);
|
|
lame->last_offs = GST_BUFFER_OFFSET (buf);
|
|
lame->last_duration = duration;
|
|
} else {
|
|
lame->last_duration += duration;
|
|
}
|
|
|
|
gst_buffer_unref (buf);
|
|
|
|
if (mp3_size < 0) {
|
|
g_warning ("error %d", mp3_size);
|
|
}
|
|
|
|
if (mp3_size > 0) {
|
|
GstBuffer *outbuf;
|
|
|
|
outbuf = gst_buffer_new ();
|
|
GST_BUFFER_DATA (outbuf) = mp3_data;
|
|
GST_BUFFER_MALLOCDATA (outbuf) = mp3_data;
|
|
GST_BUFFER_SIZE (outbuf) = mp3_size;
|
|
GST_BUFFER_TIMESTAMP (outbuf) = lame->last_ts;
|
|
GST_BUFFER_OFFSET (outbuf) = lame->last_offs;
|
|
GST_BUFFER_DURATION (outbuf) = lame->last_duration;
|
|
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (lame->srcpad));
|
|
|
|
result = gst_pad_push (lame->srcpad, outbuf);
|
|
lame->last_flow = result;
|
|
if (result != GST_FLOW_OK) {
|
|
GST_DEBUG_OBJECT (lame, "flow return: %s", gst_flow_get_name (result));
|
|
}
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (lame->last_ts))
|
|
lame->eos_ts = lame->last_ts + lame->last_duration;
|
|
else
|
|
lame->eos_ts = GST_CLOCK_TIME_NONE;
|
|
lame->last_ts = GST_CLOCK_TIME_NONE;
|
|
} else {
|
|
g_free (mp3_data);
|
|
result = GST_FLOW_OK;
|
|
}
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
not_setup:
|
|
{
|
|
gst_buffer_unref (buf);
|
|
GST_ELEMENT_ERROR (lame, CORE, NEGOTIATION, (NULL),
|
|
("encoder not initialized (input is not audio?)"));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
/* set up the encoder state */
|
|
static gboolean
|
|
gst_lamemp3enc_setup (GstLameMP3Enc * lame, GstTagList ** tags)
|
|
{
|
|
|
|
#define CHECK_ERROR(command) G_STMT_START {\
|
|
if ((command) < 0) { \
|
|
GST_ERROR_OBJECT (lame, "setup failed: " G_STRINGIFY (command)); \
|
|
if (*tags) { \
|
|
gst_tag_list_free (*tags); \
|
|
*tags = NULL; \
|
|
} \
|
|
return FALSE; \
|
|
} \
|
|
}G_STMT_END
|
|
|
|
int retval;
|
|
GstCaps *allowed_caps;
|
|
|
|
GST_DEBUG_OBJECT (lame, "starting setup");
|
|
|
|
/* check if we're already setup; if we are, we might want to check
|
|
* if this initialization is compatible with the previous one */
|
|
/* FIXME: do this */
|
|
if (lame->setup) {
|
|
GST_WARNING_OBJECT (lame, "already setup");
|
|
lame->setup = FALSE;
|
|
}
|
|
|
|
lame->lgf = lame_init ();
|
|
|
|
if (lame->lgf == NULL)
|
|
return FALSE;
|
|
|
|
*tags = gst_tag_list_new ();
|
|
|
|
/* post latency message on the bus */
|
|
gst_element_post_message (GST_ELEMENT (lame),
|
|
gst_message_new_latency (GST_OBJECT (lame)));
|
|
|
|
/* 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 (lame->srcpad);
|
|
|
|
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, 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) {
|
|
lame->setup = TRUE;
|
|
/* 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);
|
|
} else {
|
|
GST_ERROR_OBJECT (lame, "lame_init_params returned %d", retval);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (lame, "done with setup");
|
|
|
|
return lame->setup;
|
|
#undef CHECK_ERROR
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_lamemp3enc_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstLameMP3Enc *lame;
|
|
GstStateChangeReturn result;
|
|
|
|
lame = GST_LAMEMP3ENC (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
lame->last_flow = GST_FLOW_OK;
|
|
lame->last_ts = GST_CLOCK_TIME_NONE;
|
|
lame->eos_ts = GST_CLOCK_TIME_NONE;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
gst_lamemp3enc_release_memory (lame);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
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;
|
|
}
|