/* 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 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 ")
);
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 const num_decoded_bytes);
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);
G_DEFINE_TYPE (GstMpg123AudioDec, gst_mpg123_audio_dec, GST_TYPE_AUDIO_DECODER);
static void
gst_mpg123_audio_dec_class_init (GstMpg123AudioDecClass * klass)
{
GstAudioDecoderClass *base_class;
GstElementClass *element_class;
GstPadTemplate *src_template, *sink_template;
int error;
GST_DEBUG_CATEGORY_INIT (mpg123_debug, "mpg123", 0, "mpg123 mp3 decoder");
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);
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);
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;
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (mpg123_decoder), TRUE);
}
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_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 const num_decoded_bytes)
{
GstBuffer *output_buffer;
GstAudioDecoder *dec;
output_buffer = NULL;
dec = GST_AUDIO_DECODER (mpg123_decoder);
if ((num_decoded_bytes == 0) || (decoded_bytes == NULL)) {
/* This occurs in the first few frames, which do not carry data; once
* MPG123_AUDIO_DEC_NEW_FORMAT is received, the empty frames stop occurring */
GST_DEBUG_OBJECT (mpg123_decoder,
"cannot decode yet, need more data -> no output buffer to push");
return GST_FLOW_OK;
}
output_buffer = gst_buffer_new_allocate (NULL, num_decoded_bytes, NULL);
if (output_buffer == NULL) {
/* This is necessary to advance playback in time,
* even when nothing was decoded. */
return gst_audio_decoder_finish_frame (dec, NULL, 1);
} else {
GstMapInfo info;
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;
mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
g_assert (mpg123_decoder->handle != NULL);
/* The actual decoding */
{
/* feed input data (if there is any) */
if (G_LIKELY (input_buffer != NULL)) {
GstMapInfo info;
if (gst_buffer_map (input_buffer, &info, GST_MAP_READ)) {
mpg123_feed (mpg123_decoder->handle, info.data, info.size);
gst_buffer_unmap (input_buffer, &info);
} else {
GST_ERROR_OBJECT (mpg123_decoder, "gst_memory_map() failed");
return GST_FLOW_ERROR;
}
}
/* 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);
}
retval = GST_FLOW_OK;
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_LOG_OBJECT (dec,
"mpg123 reported a new format -> setting next srccaps");
gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
num_decoded_bytes);
/* 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;
}
mpg123_decoder->has_next_audioinfo = FALSE;
}
break;
case MPG123_NEED_MORE:
case MPG123_OK:
retval = gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder,
decoded_bytes, num_decoded_bytes);
break;
case MPG123_DONE:
/* If this happens, then the upstream parser somehow missed the ending
* of the bitstream */
GST_LOG_OBJECT (dec, "mpg123 is done decoding");
gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
num_decoded_bytes);
retval = GST_FLOW_EOS;
break;
default:
{
/* Anything else is considered an error */
int errcode;
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, input_caps)
);
gst_caps_unref (input_caps);
break;
}
default:{
char const *errmsg = mpg123_plain_strerror (errcode);
GST_ERROR_OBJECT (dec, "Reported error: %s", errmsg);
}
}
retval = GST_FLOW_ERROR;
}
}
return retval;
}
static gboolean
gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec, GstCaps * input_caps)
{
/* Using the parsed information upstream, and the list of allowed caps
* downstream, this code tries to find a suitable audio info. It is important
* to keep in mind that the rate and number of channels should never deviate
* from the one the bitstream has, otherwise mpg123 has to mix channels and/or
* resample (and as its docs say, its internal resampler is very crude). The
* sample format, however, can be chosen freely, because the MPEG specs do not
* mandate any special format. Therefore, rate and number of channels are taken
* from upstream (which parsed the MPEG frames, so the input_caps contain
* 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));
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)