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gstreamer/subprojects/gst-plugins-good/ext/mpg123/gstmpg123audiodec.c
Sebastian Dröge 0ef396359c gst: Move GstQueueArray as GstVecDeque to core 
And change lengths and indices from guint to gsize for a more correct type.

Also deprecate GstQueueArray and implement it in terms of GstVecDeque.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-05-06 18:25:42 +00:00

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/* 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 <config.h>
#endif
#include "gstmpg123audiodec.h"
#include <stdlib.h>
#include <string.h>
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_PRIMARY, 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 <dv@pseudoterminal.org>");
/* 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_vec_deque_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_vec_deque_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_vec_deque_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_vec_deque_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_vec_deque_clear (mpg123_decoder->audio_clip_info_queue);
}
static guint
gst_mpg123_audio_dec_get_info_queue_size (GstMpg123AudioDec * mpg123_decoder)
{
return gst_vec_deque_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)
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