mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-29 21:21:12 +00:00
671c89c392
Co-authored-by: Sebastian Dröge <sebastian@centricular.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1028>
809 lines
28 KiB
C
809 lines
28 KiB
C
/* 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_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 <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_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;
|
|
|
|
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_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 (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;
|
|
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)
|