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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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d12fdf785c
Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
645 lines
21 KiB
C
645 lines
21 KiB
C
/* MP3 decoding plugin for GStreamer using the mpg123 library
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* Copyright (C) 2012 Carlos Rafael Giani
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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 Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 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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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* SECTION: element-mpg123audiodec
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* @see_also: lamemp3enc, mad
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*
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* Audio decoder for MPEG-1 layer 1/2/3 audio data.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch filesrc location=music.mp3 ! mpegaudioparse ! mpg123audiodec ! audioconvert ! audioresample ! autoaudiosink
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* ]| Decode and play the mp3 file
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* </refsect2>
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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 "gstmpg123audiodec.h"
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#include <stdlib.h>
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#include <string.h>
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GST_DEBUG_CATEGORY_STATIC (mpg123_debug);
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#define GST_CAT_DEFAULT mpg123_debug
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/* Omitted sample formats that mpg123 supports (or at least can support):
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* - 8bit integer signed
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* - 8bit integer unsigned
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* - a-law
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* - mu-law
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* - 64bit float
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*
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* The first four formats are not supported by the GstAudioDecoder base class.
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* (The internal gst_audio_format_from_caps_structure() call fails.)
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*
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* The 64bit float issue is tricky. mpg123 actually decodes to "real",
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* not necessarily to "float".
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*
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* "real" can be fixed point, 32bit float, 64bit float. There seems to be
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* no way how to find out which one of them is actually used.
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*
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* However, in all known installations, "real" equals 32bit float, so that's
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* what is used. */
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static GstStaticPadTemplate static_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/mpeg, "
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"mpegversion = (int) { 1 }, "
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"layer = (int) [ 1, 3 ], "
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"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
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"channels = (int) [ 1, 2 ], " "parsed = (boolean) true ")
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);
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static gboolean gst_mpg123_audio_dec_start (GstAudioDecoder * dec);
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static gboolean gst_mpg123_audio_dec_stop (GstAudioDecoder * dec);
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static GstFlowReturn gst_mpg123_audio_dec_push_decoded_bytes (GstMpg123AudioDec
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* mpg123_decoder, unsigned char const *decoded_bytes,
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size_t const num_decoded_bytes);
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static GstFlowReturn gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * input_buffer);
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static gboolean gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec,
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GstCaps * input_caps);
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static void gst_mpg123_audio_dec_flush (GstAudioDecoder * dec, gboolean hard);
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G_DEFINE_TYPE (GstMpg123AudioDec, gst_mpg123_audio_dec, GST_TYPE_AUDIO_DECODER);
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static void
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gst_mpg123_audio_dec_class_init (GstMpg123AudioDecClass * klass)
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{
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GstAudioDecoderClass *base_class;
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GstElementClass *element_class;
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GstPadTemplate *src_template, *sink_template;
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int error;
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GST_DEBUG_CATEGORY_INIT (mpg123_debug, "mpg123", 0, "mpg123 mp3 decoder");
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base_class = GST_AUDIO_DECODER_CLASS (klass);
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element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_set_static_metadata (element_class,
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"mpg123 mp3 decoder",
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"Codec/Decoder/Audio",
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"Decodes mp3 streams using the mpg123 library",
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"Carlos Rafael Giani <dv@pseudoterminal.org>");
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/* Not using static pad template for srccaps, since the comma-separated list
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* of formats needs to be created depending on whatever mpg123 supports */
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{
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const int *format_list;
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const long *rates_list;
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size_t num, i;
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GString *s;
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GstCaps *src_template_caps;
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s = g_string_new ("audio/x-raw, ");
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mpg123_encodings (&format_list, &num);
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g_string_append (s, "format = { ");
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for (i = 0; i < num; ++i) {
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switch (format_list[i]) {
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case MPG123_ENC_SIGNED_16:
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g_string_append (s, (i > 0) ? ", " : "");
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g_string_append (s, GST_AUDIO_NE (S16));
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break;
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case MPG123_ENC_UNSIGNED_16:
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g_string_append (s, (i > 0) ? ", " : "");
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g_string_append (s, GST_AUDIO_NE (U16));
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break;
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case MPG123_ENC_SIGNED_24:
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g_string_append (s, (i > 0) ? ", " : "");
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g_string_append (s, GST_AUDIO_NE (S24));
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break;
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case MPG123_ENC_UNSIGNED_24:
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g_string_append (s, (i > 0) ? ", " : "");
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g_string_append (s, GST_AUDIO_NE (U24));
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break;
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case MPG123_ENC_SIGNED_32:
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g_string_append (s, (i > 0) ? ", " : "");
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g_string_append (s, GST_AUDIO_NE (S32));
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break;
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case MPG123_ENC_UNSIGNED_32:
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g_string_append (s, (i > 0) ? ", " : "");
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g_string_append (s, GST_AUDIO_NE (U32));
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break;
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case MPG123_ENC_FLOAT_32:
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g_string_append (s, (i > 0) ? ", " : "");
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g_string_append (s, GST_AUDIO_NE (F32));
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break;
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default:
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GST_DEBUG ("Ignoring mpg123 format %d", format_list[i]);
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break;
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}
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}
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g_string_append (s, " }, ");
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mpg123_rates (&rates_list, &num);
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g_string_append (s, "rate = (int) { ");
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for (i = 0; i < num; ++i) {
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g_string_append_printf (s, "%s%lu", (i > 0) ? ", " : "", rates_list[i]);
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}
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g_string_append (s, "}, ");
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g_string_append (s, "channels = (int) [ 1, 2 ], ");
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g_string_append (s, "layout = (string) interleaved");
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src_template_caps = gst_caps_from_string (s->str);
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src_template = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
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src_template_caps);
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g_string_free (s, TRUE);
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}
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sink_template = gst_static_pad_template_get (&static_sink_template);
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gst_element_class_add_pad_template (element_class, sink_template);
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gst_element_class_add_pad_template (element_class, src_template);
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base_class->start = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_stop);
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base_class->handle_frame =
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GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_handle_frame);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_set_format);
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base_class->flush = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_flush);
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error = mpg123_init ();
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if (G_UNLIKELY (error != MPG123_OK))
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GST_ERROR ("Could not initialize mpg123 library: %s",
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mpg123_plain_strerror (error));
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else
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GST_INFO ("mpg123 library initialized");
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}
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void
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gst_mpg123_audio_dec_init (GstMpg123AudioDec * mpg123_decoder)
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{
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mpg123_decoder->handle = NULL;
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gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (mpg123_decoder), TRUE);
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}
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static gboolean
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gst_mpg123_audio_dec_start (GstAudioDecoder * dec)
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{
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GstMpg123AudioDec *mpg123_decoder;
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int error;
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mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
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error = 0;
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mpg123_decoder->handle = mpg123_new (NULL, &error);
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mpg123_decoder->has_next_audioinfo = FALSE;
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mpg123_decoder->frame_offset = 0;
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/* Initially, the mpg123 handle comes with a set of default formats
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* supported. This clears this set. This is necessary, since only one
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* format shall be supported (see set_format for more). */
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mpg123_format_none (mpg123_decoder->handle);
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/* Built-in mpg123 support for gapless decoding is disabled for now,
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* since it does not work well with seeking */
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mpg123_param (mpg123_decoder->handle, MPG123_REMOVE_FLAGS, MPG123_GAPLESS, 0);
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/* Tells mpg123 to use a small read-ahead buffer for better MPEG sync;
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* essential for MP3 radio streams */
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mpg123_param (mpg123_decoder->handle, MPG123_ADD_FLAGS, MPG123_SEEKBUFFER, 0);
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/* Sets the resync limit to the end of the stream (otherwise mpg123 may give
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* up on decoding prematurely, especially with mp3 web radios) */
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mpg123_param (mpg123_decoder->handle, MPG123_RESYNC_LIMIT, -1, 0);
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#if MPG123_API_VERSION >= 36
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/* The precise API version where MPG123_AUTO_RESAMPLE appeared is
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* somewhere between 29 and 36 */
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/* Don't let mpg123 resample output */
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mpg123_param (mpg123_decoder->handle, MPG123_REMOVE_FLAGS,
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MPG123_AUTO_RESAMPLE, 0);
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#endif
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/* Don't let mpg123 print messages to stdout/stderr */
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mpg123_param (mpg123_decoder->handle, MPG123_ADD_FLAGS, MPG123_QUIET, 0);
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/* Open in feed mode (= encoded data is fed manually into the handle). */
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error = mpg123_open_feed (mpg123_decoder->handle);
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if (G_UNLIKELY (error != MPG123_OK)) {
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GST_ELEMENT_ERROR (dec, LIBRARY, INIT, (NULL),
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("%s", mpg123_strerror (mpg123_decoder->handle)));
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mpg123_close (mpg123_decoder->handle);
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mpg123_delete (mpg123_decoder->handle);
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mpg123_decoder->handle = NULL;
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return FALSE;
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}
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GST_INFO_OBJECT (dec, "mpg123 decoder started");
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return TRUE;
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}
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static gboolean
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gst_mpg123_audio_dec_stop (GstAudioDecoder * dec)
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{
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GstMpg123AudioDec *mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
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if (G_LIKELY (mpg123_decoder->handle != NULL)) {
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mpg123_close (mpg123_decoder->handle);
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mpg123_delete (mpg123_decoder->handle);
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mpg123_decoder->handle = NULL;
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}
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GST_INFO_OBJECT (dec, "mpg123 decoder stopped");
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return TRUE;
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}
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static GstFlowReturn
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gst_mpg123_audio_dec_push_decoded_bytes (GstMpg123AudioDec * mpg123_decoder,
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unsigned char const *decoded_bytes, size_t const num_decoded_bytes)
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{
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GstBuffer *output_buffer;
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GstAudioDecoder *dec;
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output_buffer = NULL;
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dec = GST_AUDIO_DECODER (mpg123_decoder);
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if ((num_decoded_bytes == 0) || (decoded_bytes == NULL)) {
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/* This occurs in the first few frames, which do not carry data; once
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* MPG123_AUDIO_DEC_NEW_FORMAT is received, the empty frames stop occurring */
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GST_DEBUG_OBJECT (mpg123_decoder,
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"cannot decode yet, need more data -> no output buffer to push");
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return GST_FLOW_OK;
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}
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output_buffer = gst_buffer_new_allocate (NULL, num_decoded_bytes, NULL);
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if (output_buffer == NULL) {
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/* This is necessary to advance playback in time,
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* even when nothing was decoded. */
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return gst_audio_decoder_finish_frame (dec, NULL, 1);
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} else {
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GstMapInfo info;
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if (gst_buffer_map (output_buffer, &info, GST_MAP_WRITE)) {
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memcpy (info.data, decoded_bytes, num_decoded_bytes);
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gst_buffer_unmap (output_buffer, &info);
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} else {
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GST_ERROR_OBJECT (mpg123_decoder, "gst_buffer_map() returned NULL");
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gst_buffer_unref (output_buffer);
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output_buffer = NULL;
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}
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return gst_audio_decoder_finish_frame (dec, output_buffer, 1);
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}
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}
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static GstFlowReturn
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gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * input_buffer)
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{
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GstMpg123AudioDec *mpg123_decoder;
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int decode_error;
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unsigned char *decoded_bytes;
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size_t num_decoded_bytes;
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GstFlowReturn retval;
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mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
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g_assert (mpg123_decoder->handle != NULL);
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/* The actual decoding */
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{
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/* feed input data (if there is any) */
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if (G_LIKELY (input_buffer != NULL)) {
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GstMapInfo info;
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if (gst_buffer_map (input_buffer, &info, GST_MAP_READ)) {
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mpg123_feed (mpg123_decoder->handle, info.data, info.size);
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gst_buffer_unmap (input_buffer, &info);
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} else {
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GST_AUDIO_DECODER_ERROR (mpg123_decoder, 1, RESOURCE, READ, (NULL),
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("gst_memory_map() failed"), retval);
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return retval;
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}
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}
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/* Try to decode a frame */
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decoded_bytes = NULL;
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num_decoded_bytes = 0;
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decode_error = mpg123_decode_frame (mpg123_decoder->handle,
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&mpg123_decoder->frame_offset, &decoded_bytes, &num_decoded_bytes);
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}
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retval = GST_FLOW_OK;
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switch (decode_error) {
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case MPG123_NEW_FORMAT:
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/* As mentioned in gst_mpg123_audio_dec_set_format(), the next audioinfo
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* is not set immediately; instead, the code waits for mpg123 to take
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* note of the new format, and then sets the audioinfo. This fixes glitches
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* with mp3s containing several format headers (for example, first half
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* using 44.1kHz, second half 32 kHz) */
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GST_LOG_OBJECT (dec,
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"mpg123 reported a new format -> setting next srccaps");
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gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
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num_decoded_bytes);
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/* If there is a next audioinfo, use it, then set has_next_audioinfo to
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* FALSE, to make sure gst_audio_decoder_set_output_format() isn't called
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* again until set_format is called by the base class */
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if (mpg123_decoder->has_next_audioinfo) {
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if (!gst_audio_decoder_set_output_format (dec,
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&(mpg123_decoder->next_audioinfo))) {
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GST_WARNING_OBJECT (dec, "Unable to set output format");
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retval = GST_FLOW_NOT_NEGOTIATED;
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}
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mpg123_decoder->has_next_audioinfo = FALSE;
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}
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break;
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case MPG123_NEED_MORE:
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case MPG123_OK:
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retval = gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder,
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decoded_bytes, num_decoded_bytes);
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break;
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case MPG123_DONE:
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/* If this happens, then the upstream parser somehow missed the ending
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* of the bitstream */
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GST_LOG_OBJECT (dec, "mpg123 is done decoding");
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gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
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num_decoded_bytes);
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retval = GST_FLOW_EOS;
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break;
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default:
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{
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/* Anything else is considered an error */
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int errcode;
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retval = GST_FLOW_ERROR; /* use error by default */
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switch (decode_error) {
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case MPG123_ERR:
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errcode = mpg123_errcode (mpg123_decoder->handle);
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break;
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default:
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errcode = decode_error;
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}
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switch (errcode) {
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case MPG123_BAD_OUTFORMAT:{
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GstCaps *input_caps =
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gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (dec));
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GST_ELEMENT_ERROR (dec, STREAM, FORMAT, (NULL),
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("Output sample format could not be used when trying to decode frame. "
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"This is typically caused when the input caps (often the sample "
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"rate) do not match the actual format of the audio data. "
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"Input caps: %" GST_PTR_FORMAT, input_caps)
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);
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gst_caps_unref (input_caps);
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break;
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}
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default:{
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char const *errmsg = mpg123_plain_strerror (errcode);
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/* GST_AUDIO_DECODER_ERROR sets a new return value according to
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* its estimations */
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GST_AUDIO_DECODER_ERROR (mpg123_decoder, 1, STREAM, DECODE, (NULL),
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("mpg123 decoding error: %s", errmsg), retval);
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}
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}
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}
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}
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return retval;
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}
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static gboolean
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gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec, GstCaps * input_caps)
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{
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/* Using the parsed information upstream, and the list of allowed caps
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* downstream, this code tries to find a suitable audio info. It is important
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* to keep in mind that the rate and number of channels should never deviate
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* from the one the bitstream has, otherwise mpg123 has to mix channels and/or
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* resample (and as its docs say, its internal resampler is very crude). The
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* sample format, however, can be chosen freely, because the MPEG specs do not
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* mandate any special format. Therefore, rate and number of channels are taken
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* from upstream (which parsed the MPEG frames, so the input_caps contain
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* exactly the rate and number of channels the bitstream actually has), while
|
|
* the sample format is chosen by trying out all caps that are allowed by
|
|
* downstream. This way, the output is adjusted to what the downstream prefers.
|
|
*
|
|
* Also, the new output audio info is not set immediately. Instead, it is
|
|
* considered the "next audioinfo". The code waits for mpg123 to notice the new
|
|
* format (= when mpg123_decode_frame() returns MPG123_AUDIO_DEC_NEW_FORMAT),
|
|
* and then sets the next audioinfo. Otherwise, the next audioinfo is set too
|
|
* soon, which may cause problems with mp3s containing several format headers.
|
|
* One example would be an mp3 with the first 30 seconds using 44.1 kHz, then
|
|
* the next 30 seconds using 32 kHz. Rare, but possible.
|
|
*
|
|
* STEPS:
|
|
*
|
|
* 1. get rate and channels from input_caps
|
|
* 2. get allowed caps from src pad
|
|
* 3. for each structure in allowed caps:
|
|
* 3.1. take format
|
|
* 3.2. if the combination of format with rate and channels is unsupported by
|
|
* mpg123, go to (3), or exit with error if there are no more structures
|
|
* to try
|
|
* 3.3. create next audioinfo out of rate,channels,format, and exit
|
|
*/
|
|
|
|
|
|
int rate, channels;
|
|
GstMpg123AudioDec *mpg123_decoder;
|
|
GstCaps *allowed_srccaps;
|
|
guint structure_nr;
|
|
gboolean match_found = FALSE;
|
|
|
|
mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
|
|
|
|
g_assert (mpg123_decoder->handle != NULL);
|
|
|
|
mpg123_decoder->has_next_audioinfo = FALSE;
|
|
|
|
/* Get rate and channels from input_caps */
|
|
{
|
|
GstStructure *structure;
|
|
gboolean err = FALSE;
|
|
|
|
/* Only the first structure is used (multiple
|
|
* input caps structures don't make sense */
|
|
structure = gst_caps_get_structure (input_caps, 0);
|
|
|
|
if (!gst_structure_get_int (structure, "rate", &rate)) {
|
|
err = TRUE;
|
|
GST_ERROR_OBJECT (dec, "Input caps do not have a rate value");
|
|
}
|
|
if (!gst_structure_get_int (structure, "channels", &channels)) {
|
|
err = TRUE;
|
|
GST_ERROR_OBJECT (dec, "Input caps do not have a channel value");
|
|
}
|
|
|
|
if (err)
|
|
return FALSE;
|
|
}
|
|
|
|
/* Get the caps that are allowed by downstream */
|
|
{
|
|
GstCaps *allowed_srccaps_unnorm =
|
|
gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
|
|
if (!allowed_srccaps_unnorm) {
|
|
GST_ERROR_OBJECT (dec, "Allowed src caps are NULL");
|
|
return FALSE;
|
|
}
|
|
allowed_srccaps = gst_caps_normalize (allowed_srccaps_unnorm);
|
|
}
|
|
|
|
/* Go through all allowed caps, pick the first one that matches */
|
|
for (structure_nr = 0; structure_nr < gst_caps_get_size (allowed_srccaps);
|
|
++structure_nr) {
|
|
GstStructure *structure;
|
|
gchar const *format_str;
|
|
GstAudioFormat format;
|
|
int encoding;
|
|
|
|
structure = gst_caps_get_structure (allowed_srccaps, structure_nr);
|
|
|
|
format_str = gst_structure_get_string (structure, "format");
|
|
if (format_str == NULL) {
|
|
GST_DEBUG_OBJECT (dec, "Could not get format from src caps");
|
|
continue;
|
|
}
|
|
|
|
format = gst_audio_format_from_string (format_str);
|
|
if (format == GST_AUDIO_FORMAT_UNKNOWN) {
|
|
GST_DEBUG_OBJECT (dec, "Unknown format %s", format_str);
|
|
continue;
|
|
}
|
|
|
|
switch (format) {
|
|
case GST_AUDIO_FORMAT_S16:
|
|
encoding = MPG123_ENC_SIGNED_16;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S24:
|
|
encoding = MPG123_ENC_SIGNED_24;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S32:
|
|
encoding = MPG123_ENC_SIGNED_32;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U16:
|
|
encoding = MPG123_ENC_UNSIGNED_16;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U24:
|
|
encoding = MPG123_ENC_UNSIGNED_24;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U32:
|
|
encoding = MPG123_ENC_UNSIGNED_32;
|
|
break;
|
|
case GST_AUDIO_FORMAT_F32:
|
|
encoding = MPG123_ENC_FLOAT_32;
|
|
break;
|
|
default:
|
|
GST_DEBUG_OBJECT (dec,
|
|
"Format %s in srccaps is not supported", format_str);
|
|
continue;
|
|
}
|
|
|
|
{
|
|
int err;
|
|
|
|
/* Cleanup old formats & set new one */
|
|
mpg123_format_none (mpg123_decoder->handle);
|
|
err = mpg123_format (mpg123_decoder->handle, rate, channels, encoding);
|
|
if (err != MPG123_OK) {
|
|
GST_DEBUG_OBJECT (dec,
|
|
"mpg123 cannot use caps %" GST_PTR_FORMAT
|
|
" because mpg123_format() failed: %s", structure,
|
|
mpg123_strerror (mpg123_decoder->handle));
|
|
continue;
|
|
}
|
|
}
|
|
|
|
gst_audio_info_init (&(mpg123_decoder->next_audioinfo));
|
|
gst_audio_info_set_format (&(mpg123_decoder->next_audioinfo), format, rate,
|
|
channels, NULL);
|
|
GST_LOG_OBJECT (dec, "The next audio format is: %s, %u Hz, %u channels",
|
|
format_str, rate, channels);
|
|
mpg123_decoder->has_next_audioinfo = TRUE;
|
|
|
|
match_found = TRUE;
|
|
|
|
break;
|
|
}
|
|
|
|
gst_caps_unref (allowed_srccaps);
|
|
|
|
return match_found;
|
|
}
|
|
|
|
|
|
static void
|
|
gst_mpg123_audio_dec_flush (GstAudioDecoder * dec, gboolean hard)
|
|
{
|
|
int error;
|
|
GstMpg123AudioDec *mpg123_decoder;
|
|
|
|
GST_LOG_OBJECT (dec, "Flushing decoder");
|
|
|
|
mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
|
|
|
|
g_assert (mpg123_decoder->handle != NULL);
|
|
|
|
/* Flush by reopening the feed */
|
|
mpg123_close (mpg123_decoder->handle);
|
|
error = mpg123_open_feed (mpg123_decoder->handle);
|
|
|
|
if (G_UNLIKELY (error != MPG123_OK)) {
|
|
GST_ELEMENT_ERROR (dec, LIBRARY, INIT, (NULL),
|
|
("Error while reopening mpg123 feed: %s",
|
|
mpg123_plain_strerror (error)));
|
|
mpg123_close (mpg123_decoder->handle);
|
|
mpg123_delete (mpg123_decoder->handle);
|
|
mpg123_decoder->handle = NULL;
|
|
}
|
|
|
|
mpg123_decoder->has_next_audioinfo = FALSE;
|
|
|
|
/* opening/closing feeds do not affect the format defined by the
|
|
* mpg123_format() call that was made in gst_mpg123_audio_dec_set_format(),
|
|
* and since the up/downstream caps are not expected to change here, no
|
|
* mpg123_format() calls are done */
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "mpg123audiodec",
|
|
GST_RANK_MARGINAL, gst_mpg123_audio_dec_get_type ());
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
mpg123, "mp3 decoding based on the mpg123 library",
|
|
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|