/* MP3 decoding plugin for GStreamer using the mpg123 library * Copyright (C) 2012 Carlos Rafael Giani * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with this library; if not, write to the Free Software * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA */ /** * SECTION: element-mpg123audiodec * @see_also: lamemp3enc, mad * * Audio decoder for MPEG-1 layer 1/2/3 audio data. * * ## Example pipelines * * |[ * gst-launch-1.0 filesrc location=music.mp3 ! mpegaudioparse ! mpg123audiodec ! audioconvert ! audioresample ! autoaudiosink * ]| Decode and play the mp3 file */ #ifdef HAVE_CONFIG_H #include #endif #include "gstmpg123audiodec.h" #include #include GST_DEBUG_CATEGORY_STATIC (mpg123_debug); #define GST_CAT_DEFAULT mpg123_debug /* Omitted sample formats that mpg123 supports (or at least can support): * - 8bit integer signed * - 8bit integer unsigned * - a-law * - mu-law * - 64bit float * * The first four formats are not supported by the GstAudioDecoder base class. * (The internal gst_audio_format_from_caps_structure() call fails.) * * The 64bit float issue is tricky. mpg123 actually decodes to "real", * not necessarily to "float". * * "real" can be fixed point, 32bit float, 64bit float. There seems to be * no way how to find out which one of them is actually used. * * However, in all known installations, "real" equals 32bit float, so that's * what is used. */ static GstStaticPadTemplate static_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1, " "layer = (int) [ 1, 3 ], " "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, " "channels = (int) [ 1, 2 ], " "parsed = (boolean) true ") ); typedef struct { guint64 clip_start, clip_end; } GstMpg123AudioDecClipInfo; static void gst_mpg123_audio_dec_dispose (GObject * object); static gboolean gst_mpg123_audio_dec_start (GstAudioDecoder * dec); static gboolean gst_mpg123_audio_dec_stop (GstAudioDecoder * dec); static GstFlowReturn gst_mpg123_audio_dec_push_decoded_bytes (GstMpg123AudioDec * mpg123_decoder, unsigned char const *decoded_bytes, size_t num_decoded_bytes, guint64 clip_start, guint64 clip_end); static GstFlowReturn gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec, GstBuffer * input_buffer); static gboolean gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec, GstCaps * input_caps); static void gst_mpg123_audio_dec_flush (GstAudioDecoder * dec, gboolean hard); static void gst_mpg123_audio_dec_push_clip_info (GstMpg123AudioDec * mpg123_decoder, guint64 clip_start, guint64 clip_end); static void gst_mpg123_audio_dec_pop_oldest_clip_info (GstMpg123AudioDec * mpg123_decoder, guint64 * clip_start, guint64 * clip_end); static void gst_mpg123_audio_dec_clear_clip_info_queue (GstMpg123AudioDec * mpg123_decoder); static guint gst_mpg123_audio_dec_get_info_queue_size (GstMpg123AudioDec * mpg123_decoder); G_DEFINE_TYPE (GstMpg123AudioDec, gst_mpg123_audio_dec, GST_TYPE_AUDIO_DECODER); GST_ELEMENT_REGISTER_DEFINE (mpg123audiodec, "mpg123audiodec", GST_RANK_MARGINAL, GST_TYPE_MPG123_AUDIO_DEC); static void gst_mpg123_audio_dec_class_init (GstMpg123AudioDecClass * klass) { GObjectClass *object_class; GstAudioDecoderClass *base_class; GstElementClass *element_class; GstPadTemplate *src_template, *sink_template; int error; GST_DEBUG_CATEGORY_INIT (mpg123_debug, "mpg123", 0, "mpg123 mp3 decoder"); object_class = G_OBJECT_CLASS (klass); base_class = GST_AUDIO_DECODER_CLASS (klass); element_class = GST_ELEMENT_CLASS (klass); gst_element_class_set_static_metadata (element_class, "mpg123 mp3 decoder", "Codec/Decoder/Audio", "Decodes mp3 streams using the mpg123 library", "Carlos Rafael Giani "); /* Not using static pad template for srccaps, since the comma-separated list * of formats needs to be created depending on whatever mpg123 supports */ { const int *format_list; const long *rates_list; size_t num, i; GString *s; GstCaps *src_template_caps; s = g_string_new ("audio/x-raw, "); mpg123_encodings (&format_list, &num); g_string_append (s, "format = { "); for (i = 0; i < num; ++i) { switch (format_list[i]) { case MPG123_ENC_SIGNED_16: g_string_append (s, (i > 0) ? ", " : ""); g_string_append (s, GST_AUDIO_NE (S16)); break; case MPG123_ENC_UNSIGNED_16: g_string_append (s, (i > 0) ? ", " : ""); g_string_append (s, GST_AUDIO_NE (U16)); break; case MPG123_ENC_SIGNED_24: g_string_append (s, (i > 0) ? ", " : ""); g_string_append (s, GST_AUDIO_NE (S24)); break; case MPG123_ENC_UNSIGNED_24: g_string_append (s, (i > 0) ? ", " : ""); g_string_append (s, GST_AUDIO_NE (U24)); break; case MPG123_ENC_SIGNED_32: g_string_append (s, (i > 0) ? ", " : ""); g_string_append (s, GST_AUDIO_NE (S32)); break; case MPG123_ENC_UNSIGNED_32: g_string_append (s, (i > 0) ? ", " : ""); g_string_append (s, GST_AUDIO_NE (U32)); break; case MPG123_ENC_FLOAT_32: g_string_append (s, (i > 0) ? ", " : ""); g_string_append (s, GST_AUDIO_NE (F32)); break; default: GST_DEBUG ("Ignoring mpg123 format %d", format_list[i]); break; } } g_string_append (s, " }, "); mpg123_rates (&rates_list, &num); g_string_append (s, "rate = (int) { "); for (i = 0; i < num; ++i) { g_string_append_printf (s, "%s%lu", (i > 0) ? ", " : "", rates_list[i]); } g_string_append (s, "}, "); g_string_append (s, "channels = (int) [ 1, 2 ], "); g_string_append (s, "layout = (string) interleaved"); src_template_caps = gst_caps_from_string (s->str); src_template = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS, src_template_caps); gst_caps_unref (src_template_caps); g_string_free (s, TRUE); } sink_template = gst_static_pad_template_get (&static_sink_template); gst_element_class_add_pad_template (element_class, sink_template); gst_element_class_add_pad_template (element_class, src_template); object_class->dispose = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_dispose); base_class->start = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_start); base_class->stop = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_stop); base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_handle_frame); base_class->set_format = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_set_format); base_class->flush = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_flush); error = mpg123_init (); if (G_UNLIKELY (error != MPG123_OK)) GST_ERROR ("Could not initialize mpg123 library: %s", mpg123_plain_strerror (error)); else GST_INFO ("mpg123 library initialized"); } void gst_mpg123_audio_dec_init (GstMpg123AudioDec * mpg123_decoder) { mpg123_decoder->handle = NULL; mpg123_decoder->audio_clip_info_queue = gst_queue_array_new_for_struct (sizeof (GstMpg123AudioDecClipInfo), 16); gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (mpg123_decoder), TRUE); gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST (mpg123_decoder), TRUE); GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (mpg123_decoder)); } static void gst_mpg123_audio_dec_dispose (GObject * object) { GstMpg123AudioDec *mpg123_decoder = GST_MPG123_AUDIO_DEC (object); if (mpg123_decoder->audio_clip_info_queue != NULL) { gst_queue_array_free (mpg123_decoder->audio_clip_info_queue); mpg123_decoder->audio_clip_info_queue = NULL; } G_OBJECT_CLASS (gst_mpg123_audio_dec_parent_class)->dispose (object); } static gboolean gst_mpg123_audio_dec_start (GstAudioDecoder * dec) { GstMpg123AudioDec *mpg123_decoder; int error; mpg123_decoder = GST_MPG123_AUDIO_DEC (dec); error = 0; mpg123_decoder->handle = mpg123_new (NULL, &error); mpg123_decoder->has_next_audioinfo = FALSE; mpg123_decoder->frame_offset = 0; /* Initially, the mpg123 handle comes with a set of default formats * supported. This clears this set. This is necessary, since only one * format shall be supported (see set_format for more). */ mpg123_format_none (mpg123_decoder->handle); /* Built-in mpg123 support for gapless decoding is disabled for now, * since it does not work well with seeking */ mpg123_param (mpg123_decoder->handle, MPG123_REMOVE_FLAGS, MPG123_GAPLESS, 0); /* Tells mpg123 to use a small read-ahead buffer for better MPEG sync; * essential for MP3 radio streams */ mpg123_param (mpg123_decoder->handle, MPG123_ADD_FLAGS, MPG123_SEEKBUFFER, 0); /* Sets the resync limit to the end of the stream (otherwise mpg123 may give * up on decoding prematurely, especially with mp3 web radios) */ mpg123_param (mpg123_decoder->handle, MPG123_RESYNC_LIMIT, -1, 0); #if MPG123_API_VERSION >= 36 /* The precise API version where MPG123_AUTO_RESAMPLE appeared is * somewhere between 29 and 36 */ /* Don't let mpg123 resample output */ mpg123_param (mpg123_decoder->handle, MPG123_REMOVE_FLAGS, MPG123_AUTO_RESAMPLE, 0); #endif /* Don't let mpg123 print messages to stdout/stderr */ mpg123_param (mpg123_decoder->handle, MPG123_ADD_FLAGS, MPG123_QUIET, 0); /* Open in feed mode (= encoded data is fed manually into the handle). */ error = mpg123_open_feed (mpg123_decoder->handle); if (G_UNLIKELY (error != MPG123_OK)) { GST_ELEMENT_ERROR (dec, LIBRARY, INIT, (NULL), ("%s", mpg123_strerror (mpg123_decoder->handle))); mpg123_close (mpg123_decoder->handle); mpg123_delete (mpg123_decoder->handle); mpg123_decoder->handle = NULL; return FALSE; } GST_INFO_OBJECT (dec, "mpg123 decoder started"); return TRUE; } static gboolean gst_mpg123_audio_dec_stop (GstAudioDecoder * dec) { GstMpg123AudioDec *mpg123_decoder = GST_MPG123_AUDIO_DEC (dec); if (G_LIKELY (mpg123_decoder->handle != NULL)) { mpg123_close (mpg123_decoder->handle); mpg123_delete (mpg123_decoder->handle); mpg123_decoder->handle = NULL; } gst_mpg123_audio_dec_clear_clip_info_queue (mpg123_decoder); GST_INFO_OBJECT (dec, "mpg123 decoder stopped"); return TRUE; } static GstFlowReturn gst_mpg123_audio_dec_push_decoded_bytes (GstMpg123AudioDec * mpg123_decoder, unsigned char const *decoded_bytes, size_t num_decoded_bytes, guint64 clip_start, guint64 clip_end) { GstBuffer *output_buffer; GstAudioDecoder *dec; GstMapInfo info; output_buffer = NULL; dec = GST_AUDIO_DECODER (mpg123_decoder); if (G_UNLIKELY ((num_decoded_bytes == 0) || (decoded_bytes == NULL))) { /* This occurs in two cases: * * 1. The first few frames come in. These fill mpg123's buffers, and * do not immediately yield decoded output. This stops once the * mpg123_decode_frame () returns MPG123_NEW_FORMAT. * 2. The decoder is being drained. */ return GST_FLOW_OK; } if (G_UNLIKELY (clip_start + clip_end >= num_decoded_bytes)) { /* Fully-clipped frames still need to be finished, since they got * decoded properly, they are just made of padding samples. */ GST_LOG_OBJECT (mpg123_decoder, "frame is fully clipped; " "not pushing anything downstream"); return gst_audio_decoder_finish_frame (dec, NULL, 1); } /* Apply clipping. */ decoded_bytes += clip_start; num_decoded_bytes -= clip_start + clip_end; output_buffer = gst_audio_decoder_allocate_output_buffer (dec, num_decoded_bytes); if (gst_buffer_map (output_buffer, &info, GST_MAP_WRITE)) { memcpy (info.data, decoded_bytes, num_decoded_bytes); gst_buffer_unmap (output_buffer, &info); } else { GST_ERROR_OBJECT (mpg123_decoder, "gst_buffer_map() returned NULL"); gst_buffer_unref (output_buffer); output_buffer = NULL; } return gst_audio_decoder_finish_frame (dec, output_buffer, 1); } static GstFlowReturn gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec, GstBuffer * input_buffer) { GstMpg123AudioDec *mpg123_decoder; int decode_error; unsigned char *decoded_bytes; size_t num_decoded_bytes; GstFlowReturn retval; gboolean loop = TRUE; mpg123_decoder = GST_MPG123_AUDIO_DEC (dec); g_assert (mpg123_decoder->handle != NULL); /* Feed input data (if there is any) into mpg123. */ if (G_LIKELY (input_buffer != NULL)) { GstMapInfo info; GstAudioClippingMeta *clipping_meta = NULL; /* Drop any Xing/LAME header as marked from the parser. It's not parsed in * this element and would decode to unnecessary silence samples. */ if (GST_BUFFER_FLAG_IS_SET (input_buffer, GST_BUFFER_FLAG_DECODE_ONLY) && GST_BUFFER_FLAG_IS_SET (input_buffer, GST_BUFFER_FLAG_DROPPABLE)) { return gst_audio_decoder_finish_frame (dec, NULL, 1); } else if (gst_buffer_map (input_buffer, &info, GST_MAP_READ)) { GST_LOG_OBJECT (mpg123_decoder, "got new MPEG audio frame with %" G_GSIZE_FORMAT " byte(s); feeding it into mpg123", info.size); mpg123_feed (mpg123_decoder->handle, info.data, info.size); gst_buffer_unmap (input_buffer, &info); } else { GST_AUDIO_DECODER_ERROR (mpg123_decoder, 1, RESOURCE, READ, (NULL), ("gst_memory_map() failed; could not feed MPEG frame into mpg123"), retval); return retval; } clipping_meta = gst_buffer_get_audio_clipping_meta (input_buffer); if (clipping_meta != NULL) { if (clipping_meta->format == GST_FORMAT_DEFAULT) { /* Get clipping info and convert it to bytes. */ gint bpf = GST_AUDIO_INFO_BPF (&(mpg123_decoder->next_audioinfo)); guint64 clip_start = clipping_meta->start * bpf; guint64 clip_end = clipping_meta->end * bpf; /* Push the clipping info into the queue. We cannot use clipping info * directly since mpg123 might not immediately be able to decode this * MPEG frame. In other words, it queues the frames internally. To * make sure we apply clipping properly, we therefore also have to * queue the clipping info accordingly. */ gst_mpg123_audio_dec_push_clip_info (mpg123_decoder, clip_start, clip_end); GST_LOG_OBJECT (dec, "buffer has clipping metadata: start/end %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " samples (= %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " bytes); pushed it into " "audio clip info queue (now has %u item(s))", clipping_meta->start, clipping_meta->end, clip_start, clip_end, gst_mpg123_audio_dec_get_info_queue_size (mpg123_decoder)); } else { gst_mpg123_audio_dec_push_clip_info (mpg123_decoder, 0, 0); GST_WARNING_OBJECT (dec, "buffer has clipping metadata in unsupported format %s", gst_format_get_name (clipping_meta->format)); } } else { gst_mpg123_audio_dec_push_clip_info (mpg123_decoder, 0, 0); } } else { GST_LOG_OBJECT (dec, "got NULL pointer as input; " "will drain mpg123 decoder"); } retval = GST_FLOW_OK; /* Keep trying to decode with mpg123 until it reports that, * it is done, needs more data, or an error occurs. */ while (loop) { guint64 clip_start = 0, clip_end = 0; /* Try to decode a frame */ decoded_bytes = NULL; num_decoded_bytes = 0; decode_error = mpg123_decode_frame (mpg123_decoder->handle, &mpg123_decoder->frame_offset, &decoded_bytes, &num_decoded_bytes); if (G_LIKELY (decoded_bytes != NULL)) { gst_mpg123_audio_dec_pop_oldest_clip_info (mpg123_decoder, &clip_start, &clip_end); if ((clip_start + clip_end) > 0) { GST_LOG_OBJECT (dec, "retrieved clip info from queue; " "will clip %" G_GUINT64_FORMAT " byte(s) at the start and %" G_GUINT64_FORMAT " at the end of the decoded frame; queue now " "has %u item(s)", clip_start, clip_end, gst_mpg123_audio_dec_get_info_queue_size (mpg123_decoder)); } GST_LOG_OBJECT (dec, "decoded %" G_GSIZE_FORMAT " byte(s)", (gsize) num_decoded_bytes); } switch (decode_error) { case MPG123_NEW_FORMAT: /* As mentioned in gst_mpg123_audio_dec_set_format(), the next audioinfo * is not set immediately; instead, the code waits for mpg123 to take * note of the new format, and then sets the audioinfo. This fixes glitches * with mp3s containing several format headers (for example, first half * using 44.1kHz, second half 32 kHz) */ gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes, num_decoded_bytes, clip_start, clip_end); GST_LOG_OBJECT (dec, "mpg123 reported a new format -> setting next srccaps"); /* If there is a next audioinfo, use it, then set has_next_audioinfo to * FALSE, to make sure gst_audio_decoder_set_output_format() isn't called * again until set_format is called by the base class */ if (mpg123_decoder->has_next_audioinfo) { if (!gst_audio_decoder_set_output_format (dec, &(mpg123_decoder->next_audioinfo))) { GST_WARNING_OBJECT (dec, "Unable to set output format"); retval = GST_FLOW_NOT_NEGOTIATED; loop = FALSE; } mpg123_decoder->has_next_audioinfo = FALSE; } break; case MPG123_NEED_MORE: loop = FALSE; GST_LOG_OBJECT (dec, "mpg123 needs more data to continue decoding"); retval = gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes, num_decoded_bytes, clip_start, clip_end); break; case MPG123_OK: retval = gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes, num_decoded_bytes, clip_start, clip_end); break; case MPG123_DONE: /* If this happens, then the upstream parser somehow missed the ending * of the bitstream */ gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes, num_decoded_bytes, clip_start, clip_end); GST_LOG_OBJECT (dec, "mpg123 is done decoding"); retval = GST_FLOW_EOS; loop = FALSE; break; default: { /* Anything else is considered an error */ int errcode; /* use error by default */ retval = GST_FLOW_ERROR; loop = FALSE; switch (decode_error) { case MPG123_ERR: errcode = mpg123_errcode (mpg123_decoder->handle); break; default: errcode = decode_error; } switch (errcode) { case MPG123_BAD_OUTFORMAT:{ GstCaps *input_caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (dec)); GST_ELEMENT_ERROR (dec, STREAM, FORMAT, (NULL), ("Output sample format could not be used when trying to decode frame. " "This is typically caused when the input caps (often the sample " "rate) do not match the actual format of the audio data. " "Input caps: %" GST_PTR_FORMAT, (gpointer) input_caps) ); gst_caps_unref (input_caps); break; } default:{ char const *errmsg = mpg123_plain_strerror (errcode); /* GST_AUDIO_DECODER_ERROR sets a new return value according to * its estimations */ GST_AUDIO_DECODER_ERROR (mpg123_decoder, 1, STREAM, DECODE, (NULL), ("mpg123 decoding error: %s", errmsg), retval); } } } } } GST_LOG_OBJECT (mpg123_decoder, "done handling frame"); return retval; } static gboolean gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec, GstCaps * input_caps) { /* "encoding" is the sample format specifier for mpg123 */ int encoding; int sample_rate, num_channels; GstAudioFormat format; GstMpg123AudioDec *mpg123_decoder; gboolean retval = FALSE; mpg123_decoder = GST_MPG123_AUDIO_DEC (dec); g_assert (mpg123_decoder->handle != NULL); mpg123_decoder->has_next_audioinfo = FALSE; /* Get sample rate and number of 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", &sample_rate)) { err = TRUE; GST_ERROR_OBJECT (dec, "Input caps do not have a rate value"); } if (!gst_structure_get_int (structure, "channels", &num_channels)) { err = TRUE; GST_ERROR_OBJECT (dec, "Input caps do not have a channel value"); } if (G_UNLIKELY (err)) goto done; } /* Get sample format from the allowed src caps */ { GstCaps *allowed_srccaps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec)); if (allowed_srccaps == NULL) { /* srcpad is not linked (yet), so no peer information is available; * just use the default sample format (16 bit signed integer) */ GST_DEBUG_OBJECT (mpg123_decoder, "srcpad is not linked (yet) -> using S16 sample format"); format = GST_AUDIO_FORMAT_S16; encoding = MPG123_ENC_SIGNED_16; } else if (gst_caps_is_empty (allowed_srccaps)) { gst_caps_unref (allowed_srccaps); goto done; } else { gchar const *format_str; GValue const *format_value; /* Look at the sample format values from the first structure */ GstStructure *structure = gst_caps_get_structure (allowed_srccaps, 0); format_value = gst_structure_get_value (structure, "format"); if (format_value == NULL) { gst_caps_unref (allowed_srccaps); goto done; } else if (GST_VALUE_HOLDS_LIST (format_value)) { /* if value is a format list, pick the first entry */ GValue const *fmt_list_value = gst_value_list_get_value (format_value, 0); format_str = g_value_get_string (fmt_list_value); } else if (G_VALUE_HOLDS_STRING (format_value)) { /* if value is a string, use it directly */ format_str = g_value_get_string (format_value); } else { GST_ERROR_OBJECT (mpg123_decoder, "unexpected type for 'format' field " "in caps structure %" GST_PTR_FORMAT, (gpointer) structure); gst_caps_unref (allowed_srccaps); goto done; } /* get the format value from the string */ format = gst_audio_format_from_string (format_str); gst_caps_unref (allowed_srccaps); g_assert (format != GST_AUDIO_FORMAT_UNKNOWN); /* convert format to mpg123 encoding */ 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: g_assert_not_reached (); goto done; } } } /* Sample rate, number of channels, and sample format are known at this point. * Set the audioinfo structure's values and the mpg123 format. */ { int err; /* clear all existing format settings from the mpg123 instance */ mpg123_format_none (mpg123_decoder->handle); /* set the chosen format */ err = mpg123_format (mpg123_decoder->handle, sample_rate, num_channels, encoding); if (err != MPG123_OK) { GST_WARNING_OBJECT (dec, "mpg123_format() failed: %s", mpg123_strerror (mpg123_decoder->handle)); } else { gst_audio_info_init (&(mpg123_decoder->next_audioinfo)); gst_audio_info_set_format (&(mpg123_decoder->next_audioinfo), format, sample_rate, num_channels, NULL); GST_LOG_OBJECT (dec, "The next audio format is: %s, %u Hz, %u channels", gst_audio_format_to_string (format), sample_rate, num_channels); mpg123_decoder->has_next_audioinfo = TRUE; retval = TRUE; } } done: return retval; } 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; } if (hard) mpg123_decoder->has_next_audioinfo = FALSE; gst_mpg123_audio_dec_clear_clip_info_queue (mpg123_decoder); /* 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 void gst_mpg123_audio_dec_push_clip_info (GstMpg123AudioDec * mpg123_decoder, guint64 clip_start, guint64 clip_end) { GstMpg123AudioDecClipInfo clip_info = { clip_start, clip_end }; gst_queue_array_push_tail_struct (mpg123_decoder->audio_clip_info_queue, &clip_info); } static void gst_mpg123_audio_dec_pop_oldest_clip_info (GstMpg123AudioDec * mpg123_decoder, guint64 * clip_start, guint64 * clip_end) { guint queue_length; GstMpg123AudioDecClipInfo *clip_info; queue_length = gst_mpg123_audio_dec_get_info_queue_size (mpg123_decoder); if (queue_length == 0) return; clip_info = gst_queue_array_pop_head_struct (mpg123_decoder->audio_clip_info_queue); *clip_start = clip_info->clip_start; *clip_end = clip_info->clip_end; } static void gst_mpg123_audio_dec_clear_clip_info_queue (GstMpg123AudioDec * mpg123_decoder) { gst_queue_array_clear (mpg123_decoder->audio_clip_info_queue); } static guint gst_mpg123_audio_dec_get_info_queue_size (GstMpg123AudioDec * mpg123_decoder) { return gst_queue_array_get_length (mpg123_decoder->audio_clip_info_queue); } static gboolean plugin_init (GstPlugin * plugin) { return GST_ELEMENT_REGISTER (mpg123audiodec, plugin); } 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)