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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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Avoids useless check of downstream caps when handling an accept-caps query Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec
808 lines
26 KiB
C
808 lines
26 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2006,2011> Tim-Philipp Müller <tim centricular net>
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* Copyright (C) <2006> Jan Schmidt <thaytan at mad scientist com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-flacdec
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* @see_also: #GstFlacEnc
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*
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* flacdec decodes FLAC streams.
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* <ulink url="http://flac.sourceforge.net/">FLAC</ulink>
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* is a Free Lossless Audio Codec.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch-1.0 filesrc location=media/small/dark.441-16-s.flac ! flacparse ! flacdec ! audioconvert ! audioresample ! autoaudiosink
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* ]|
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* |[
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* gst-launch-1.0 souphttpsrc location=http://gstreamer.freedesktop.org/media/small/dark.441-16-s.flac ! flacparse ! flacdec ! audioconvert ! audioresample ! queue min-threshold-buffers=10 ! autoaudiosink
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* ]|
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "gstflacdec.h"
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#include <gst/gst-i18n-plugin.h>
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#include <gst/tag/tag.h>
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/* Taken from http://flac.sourceforge.net/format.html#frame_header */
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static const GstAudioChannelPosition channel_positions[8][8] = {
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{GST_AUDIO_CHANNEL_POSITION_MONO},
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{GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_LFE1,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
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/* FIXME: 7/8 channel layouts are not defined in the FLAC specs */
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{
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_LFE1,
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GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_LFE1,
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GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}
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};
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GST_DEBUG_CATEGORY_STATIC (flacdec_debug);
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#define GST_CAT_DEFAULT flacdec_debug
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static FLAC__StreamDecoderReadStatus
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gst_flac_dec_read_stream (const FLAC__StreamDecoder * decoder,
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FLAC__byte buffer[], size_t * bytes, void *client_data);
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static FLAC__StreamDecoderWriteStatus
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gst_flac_dec_write_stream (const FLAC__StreamDecoder * decoder,
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const FLAC__Frame * frame,
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const FLAC__int32 * const buffer[], void *client_data);
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static gboolean
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gst_flac_dec_handle_decoder_error (GstFlacDec * dec, gboolean msg);
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static void gst_flac_dec_metadata_cb (const FLAC__StreamDecoder *
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decoder, const FLAC__StreamMetadata * metadata, void *client_data);
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static void gst_flac_dec_error_cb (const FLAC__StreamDecoder *
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decoder, FLAC__StreamDecoderErrorStatus status, void *client_data);
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static void gst_flac_dec_flush (GstAudioDecoder * audio_dec, gboolean hard);
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static gboolean gst_flac_dec_set_format (GstAudioDecoder * dec, GstCaps * caps);
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static gboolean gst_flac_dec_start (GstAudioDecoder * dec);
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static gboolean gst_flac_dec_stop (GstAudioDecoder * dec);
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static GstFlowReturn gst_flac_dec_handle_frame (GstAudioDecoder * audio_dec,
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GstBuffer * buf);
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G_DEFINE_TYPE (GstFlacDec, gst_flac_dec, GST_TYPE_AUDIO_DECODER);
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#if G_BYTE_ORDER == G_LITTLE_ENDIAN
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#define FORMATS "{ S8, S16LE, S24_32LE, S32LE } "
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#else
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#define FORMATS "{ S8, S16BE, S24_32BE, S32BE } "
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#endif
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#define GST_FLAC_DEC_SRC_CAPS \
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"audio/x-raw, " \
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"format = (string) " FORMATS ", " \
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"layout = (string) interleaved, " \
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"rate = (int) [ 1, 655350 ], " \
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"channels = (int) [ 1, 8 ]"
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#define GST_FLAC_DEC_SINK_CAPS \
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"audio/x-flac, " \
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"framed = (boolean) true, " \
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"rate = (int) [ 1, 655350 ], " \
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"channels = (int) [ 1, 8 ]"
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static GstStaticPadTemplate flac_dec_src_factory =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_FLAC_DEC_SRC_CAPS));
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static GstStaticPadTemplate flac_dec_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_FLAC_DEC_SINK_CAPS));
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static void
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gst_flac_dec_class_init (GstFlacDecClass * klass)
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{
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GstAudioDecoderClass *audiodecoder_class;
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GstElementClass *gstelement_class;
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audiodecoder_class = (GstAudioDecoderClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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GST_DEBUG_CATEGORY_INIT (flacdec_debug, "flacdec", 0, "flac decoder");
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audiodecoder_class->stop = GST_DEBUG_FUNCPTR (gst_flac_dec_stop);
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audiodecoder_class->start = GST_DEBUG_FUNCPTR (gst_flac_dec_start);
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audiodecoder_class->flush = GST_DEBUG_FUNCPTR (gst_flac_dec_flush);
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audiodecoder_class->set_format = GST_DEBUG_FUNCPTR (gst_flac_dec_set_format);
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audiodecoder_class->handle_frame =
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GST_DEBUG_FUNCPTR (gst_flac_dec_handle_frame);
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&flac_dec_src_factory));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&flac_dec_sink_factory));
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gst_element_class_set_static_metadata (gstelement_class, "FLAC audio decoder",
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"Codec/Decoder/Audio", "Decodes FLAC lossless audio streams",
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"Tim-Philipp Müller <tim@centricular.net>, "
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"Wim Taymans <wim.taymans@gmail.com>");
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}
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static void
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gst_flac_dec_init (GstFlacDec * flacdec)
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{
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gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (flacdec), TRUE);
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gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
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(flacdec), TRUE);
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (flacdec));
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}
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static gboolean
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gst_flac_dec_start (GstAudioDecoder * audio_dec)
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{
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FLAC__StreamDecoderInitStatus s;
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GstFlacDec *dec;
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dec = GST_FLAC_DEC (audio_dec);
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dec->adapter = gst_adapter_new ();
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dec->decoder = FLAC__stream_decoder_new ();
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gst_audio_info_init (&dec->info);
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dec->depth = 0;
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/* no point calculating MD5 since it's never checked here */
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FLAC__stream_decoder_set_md5_checking (dec->decoder, false);
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GST_DEBUG_OBJECT (dec, "initializing decoder");
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s = FLAC__stream_decoder_init_stream (dec->decoder,
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gst_flac_dec_read_stream, NULL, NULL, NULL, NULL,
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gst_flac_dec_write_stream, gst_flac_dec_metadata_cb,
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gst_flac_dec_error_cb, dec);
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if (s != FLAC__STREAM_DECODER_INIT_STATUS_OK) {
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GST_ELEMENT_ERROR (GST_ELEMENT (dec), LIBRARY, INIT, (NULL), (NULL));
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return FALSE;
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}
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dec->got_headers = FALSE;
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return TRUE;
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}
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static gboolean
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gst_flac_dec_stop (GstAudioDecoder * dec)
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{
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GstFlacDec *flacdec = GST_FLAC_DEC (dec);
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if (flacdec->decoder) {
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FLAC__stream_decoder_delete (flacdec->decoder);
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flacdec->decoder = NULL;
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}
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if (flacdec->adapter) {
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gst_adapter_clear (flacdec->adapter);
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g_object_unref (flacdec->adapter);
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flacdec->adapter = NULL;
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}
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return TRUE;
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}
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static gboolean
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gst_flac_dec_set_format (GstAudioDecoder * dec, GstCaps * caps)
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{
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const GValue *headers;
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GstFlacDec *flacdec;
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GstStructure *s;
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guint i, num;
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flacdec = GST_FLAC_DEC (dec);
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GST_LOG_OBJECT (dec, "sink caps: %" GST_PTR_FORMAT, caps);
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s = gst_caps_get_structure (caps, 0);
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headers = gst_structure_get_value (s, "streamheader");
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if (headers == NULL || !GST_VALUE_HOLDS_ARRAY (headers)) {
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GST_WARNING_OBJECT (dec, "no 'streamheader' field in input caps, try "
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"adding a flacparse element upstream");
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return FALSE;
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}
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if (gst_adapter_available (flacdec->adapter) > 0) {
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GST_WARNING_OBJECT (dec, "unexpected data left in adapter");
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gst_adapter_clear (flacdec->adapter);
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}
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num = gst_value_array_get_size (headers);
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for (i = 0; i < num; ++i) {
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const GValue *header_val;
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GstBuffer *header_buf;
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header_val = gst_value_array_get_value (headers, i);
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if (header_val == NULL || !GST_VALUE_HOLDS_BUFFER (header_val))
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return FALSE;
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header_buf = g_value_dup_boxed (header_val);
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GST_INFO_OBJECT (dec, "pushing header buffer of %" G_GSIZE_FORMAT " bytes "
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"into adapter", gst_buffer_get_size (header_buf));
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gst_adapter_push (flacdec->adapter, header_buf);
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}
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GST_DEBUG_OBJECT (dec, "Processing headers and metadata");
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if (!FLAC__stream_decoder_process_until_end_of_metadata (flacdec->decoder)) {
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GST_WARNING_OBJECT (dec, "process_until_end_of_metadata failed");
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if (FLAC__stream_decoder_get_state (flacdec->decoder) ==
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FLAC__STREAM_DECODER_ABORTED) {
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GST_WARNING_OBJECT (flacdec, "Read callback caused internal abort");
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/* allow recovery */
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gst_adapter_clear (flacdec->adapter);
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FLAC__stream_decoder_flush (flacdec->decoder);
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gst_flac_dec_handle_decoder_error (flacdec, TRUE);
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}
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}
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GST_INFO_OBJECT (dec, "headers and metadata are now processed");
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return TRUE;
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}
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/* CRC-8, poly = x^8 + x^2 + x^1 + x^0, init = 0 */
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static const guint8 crc8_table[256] = {
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0x00, 0x07, 0x0E, 0x09, 0x1C, 0x1B, 0x12, 0x15,
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0x38, 0x3F, 0x36, 0x31, 0x24, 0x23, 0x2A, 0x2D,
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0x70, 0x77, 0x7E, 0x79, 0x6C, 0x6B, 0x62, 0x65,
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0x48, 0x4F, 0x46, 0x41, 0x54, 0x53, 0x5A, 0x5D,
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0xE0, 0xE7, 0xEE, 0xE9, 0xFC, 0xFB, 0xF2, 0xF5,
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0xD8, 0xDF, 0xD6, 0xD1, 0xC4, 0xC3, 0xCA, 0xCD,
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0x90, 0x97, 0x9E, 0x99, 0x8C, 0x8B, 0x82, 0x85,
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0xA8, 0xAF, 0xA6, 0xA1, 0xB4, 0xB3, 0xBA, 0xBD,
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0xC7, 0xC0, 0xC9, 0xCE, 0xDB, 0xDC, 0xD5, 0xD2,
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0xFF, 0xF8, 0xF1, 0xF6, 0xE3, 0xE4, 0xED, 0xEA,
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0xB7, 0xB0, 0xB9, 0xBE, 0xAB, 0xAC, 0xA5, 0xA2,
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0x8F, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9D, 0x9A,
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0x27, 0x20, 0x29, 0x2E, 0x3B, 0x3C, 0x35, 0x32,
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0x1F, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0D, 0x0A,
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0x57, 0x50, 0x59, 0x5E, 0x4B, 0x4C, 0x45, 0x42,
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0x6F, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7D, 0x7A,
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0x89, 0x8E, 0x87, 0x80, 0x95, 0x92, 0x9B, 0x9C,
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0xB1, 0xB6, 0xBF, 0xB8, 0xAD, 0xAA, 0xA3, 0xA4,
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0xF9, 0xFE, 0xF7, 0xF0, 0xE5, 0xE2, 0xEB, 0xEC,
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0xC1, 0xC6, 0xCF, 0xC8, 0xDD, 0xDA, 0xD3, 0xD4,
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0x69, 0x6E, 0x67, 0x60, 0x75, 0x72, 0x7B, 0x7C,
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0x51, 0x56, 0x5F, 0x58, 0x4D, 0x4A, 0x43, 0x44,
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0x19, 0x1E, 0x17, 0x10, 0x05, 0x02, 0x0B, 0x0C,
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0x21, 0x26, 0x2F, 0x28, 0x3D, 0x3A, 0x33, 0x34,
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0x4E, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5C, 0x5B,
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0x76, 0x71, 0x78, 0x7F, 0x6A, 0x6D, 0x64, 0x63,
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0x3E, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2C, 0x2B,
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0x06, 0x01, 0x08, 0x0F, 0x1A, 0x1D, 0x14, 0x13,
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0xAE, 0xA9, 0xA0, 0xA7, 0xB2, 0xB5, 0xBC, 0xBB,
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0x96, 0x91, 0x98, 0x9F, 0x8A, 0x8D, 0x84, 0x83,
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0xDE, 0xD9, 0xD0, 0xD7, 0xC2, 0xC5, 0xCC, 0xCB,
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0xE6, 0xE1, 0xE8, 0xEF, 0xFA, 0xFD, 0xF4, 0xF3
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};
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static guint8
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gst_flac_calculate_crc8 (const guint8 * data, guint length)
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{
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guint8 crc = 0;
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while (length--) {
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crc = crc8_table[crc ^ *data];
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++data;
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}
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return crc;
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}
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/* FIXME: for our purposes it's probably enough to just check for the sync
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* marker - we just want to know if it's a header frame or not */
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static gboolean
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gst_flac_dec_scan_got_frame (GstFlacDec * flacdec, const guint8 * data,
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guint size)
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{
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guint headerlen;
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guint sr_from_end = 0; /* can be 0, 8 or 16 */
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guint bs_from_end = 0; /* can be 0, 8 or 16 */
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guint32 val = 0;
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guint8 bs, sr, ca, ss, pb;
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gboolean vbs;
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if (size < 10)
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return FALSE;
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/* sync */
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if (data[0] != 0xFF || (data[1] & 0xFC) != 0xF8)
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return FALSE;
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vbs = ! !(data[1] & 1); /* variable blocksize */
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bs = (data[2] & 0xF0) >> 4; /* blocksize marker */
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sr = (data[2] & 0x0F); /* samplerate marker */
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ca = (data[3] & 0xF0) >> 4; /* channel assignment */
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ss = (data[3] & 0x0F) >> 1; /* sample size marker */
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pb = (data[3] & 0x01); /* padding bit */
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GST_LOG_OBJECT (flacdec,
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"got sync, vbs=%d,bs=%x,sr=%x,ca=%x,ss=%x,pb=%x", vbs, bs, sr, ca, ss,
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pb);
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if (bs == 0 || sr == 0x0F || ca >= 0x0B || ss == 0x03 || ss == 0x07) {
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return FALSE;
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}
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/* read block size from end of header? */
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if (bs == 6)
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bs_from_end = 8;
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else if (bs == 7)
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bs_from_end = 16;
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/* read sample rate from end of header? */
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if (sr == 0x0C)
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sr_from_end = 8;
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else if (sr == 0x0D || sr == 0x0E)
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sr_from_end = 16;
|
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|
|
val = data[4];
|
|
/* This is slightly faster than a loop */
|
|
if (!(val & 0x80)) {
|
|
val = 0;
|
|
} else if ((val & 0xc0) && !(val & 0x20)) {
|
|
val = 1;
|
|
} else if ((val & 0xe0) && !(val & 0x10)) {
|
|
val = 2;
|
|
} else if ((val & 0xf0) && !(val & 0x08)) {
|
|
val = 3;
|
|
} else if ((val & 0xf8) && !(val & 0x04)) {
|
|
val = 4;
|
|
} else if ((val & 0xfc) && !(val & 0x02)) {
|
|
val = 5;
|
|
} else if ((val & 0xfe) && !(val & 0x01)) {
|
|
val = 6;
|
|
} else {
|
|
GST_LOG_OBJECT (flacdec, "failed to read sample/frame");
|
|
return FALSE;
|
|
}
|
|
|
|
val++;
|
|
headerlen = 4 + val + (bs_from_end / 8) + (sr_from_end / 8);
|
|
|
|
if (gst_flac_calculate_crc8 (data, headerlen) != data[headerlen]) {
|
|
GST_LOG_OBJECT (flacdec, "invalid checksum");
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_flac_dec_handle_decoder_error (GstFlacDec * dec, gboolean msg)
|
|
{
|
|
gboolean ret;
|
|
|
|
dec->error_count++;
|
|
if (dec->error_count > 10) {
|
|
if (msg)
|
|
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), (NULL));
|
|
dec->last_flow = GST_FLOW_ERROR;
|
|
ret = TRUE;
|
|
} else {
|
|
GST_DEBUG_OBJECT (dec, "ignoring error for now at count %d",
|
|
dec->error_count);
|
|
ret = FALSE;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_flac_dec_metadata_cb (const FLAC__StreamDecoder * decoder,
|
|
const FLAC__StreamMetadata * metadata, void *client_data)
|
|
{
|
|
GstFlacDec *flacdec = GST_FLAC_DEC (client_data);
|
|
GstAudioChannelPosition position[8];
|
|
|
|
GST_LOG_OBJECT (flacdec, "metadata type: %d", metadata->type);
|
|
|
|
switch (metadata->type) {
|
|
case FLAC__METADATA_TYPE_STREAMINFO:{
|
|
gint64 samples;
|
|
guint depth, width, gdepth, channels;
|
|
|
|
samples = metadata->data.stream_info.total_samples;
|
|
|
|
flacdec->min_blocksize = metadata->data.stream_info.min_blocksize;
|
|
flacdec->max_blocksize = metadata->data.stream_info.max_blocksize;
|
|
flacdec->depth = depth = metadata->data.stream_info.bits_per_sample;
|
|
|
|
if (depth < 9) {
|
|
gdepth = width = 8;
|
|
} else if (depth < 17) {
|
|
gdepth = width = 16;
|
|
} else if (depth < 25) {
|
|
gdepth = 24;
|
|
width = 32;
|
|
} else {
|
|
gdepth = width = 32;
|
|
}
|
|
|
|
channels = metadata->data.stream_info.channels;
|
|
memcpy (position, channel_positions[channels - 1], sizeof (position));
|
|
gst_audio_channel_positions_to_valid_order (position, channels);
|
|
/* Note: we create the inverse reordering map here */
|
|
gst_audio_get_channel_reorder_map (channels,
|
|
position, channel_positions[channels - 1],
|
|
flacdec->channel_reorder_map);
|
|
|
|
gst_audio_info_set_format (&flacdec->info,
|
|
gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth),
|
|
metadata->data.stream_info.sample_rate,
|
|
metadata->data.stream_info.channels, position);
|
|
|
|
GST_DEBUG_OBJECT (flacdec, "blocksize: min=%u, max=%u",
|
|
flacdec->min_blocksize, flacdec->max_blocksize);
|
|
GST_DEBUG_OBJECT (flacdec, "sample rate: %u, channels: %u",
|
|
flacdec->info.rate, flacdec->info.channels);
|
|
GST_DEBUG_OBJECT (flacdec, "depth: %u, width: %u", flacdec->depth,
|
|
flacdec->info.finfo->width);
|
|
|
|
GST_DEBUG_OBJECT (flacdec, "total samples = %" G_GINT64_FORMAT, samples);
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_flac_dec_error_cb (const FLAC__StreamDecoder * d,
|
|
FLAC__StreamDecoderErrorStatus status, void *client_data)
|
|
{
|
|
const gchar *error;
|
|
GstFlacDec *dec;
|
|
|
|
dec = GST_FLAC_DEC (client_data);
|
|
|
|
switch (status) {
|
|
case FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC:
|
|
/* Ignore this error and keep processing */
|
|
return;
|
|
case FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER:
|
|
error = "bad header";
|
|
break;
|
|
case FLAC__STREAM_DECODER_ERROR_STATUS_FRAME_CRC_MISMATCH:
|
|
error = "CRC mismatch";
|
|
break;
|
|
default:
|
|
error = "unknown error";
|
|
break;
|
|
}
|
|
|
|
if (gst_flac_dec_handle_decoder_error (dec, FALSE))
|
|
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("%s (%d)", error, status));
|
|
}
|
|
|
|
static FLAC__StreamDecoderReadStatus
|
|
gst_flac_dec_read_stream (const FLAC__StreamDecoder * decoder,
|
|
FLAC__byte buffer[], size_t * bytes, void *client_data)
|
|
{
|
|
GstFlacDec *dec = GST_FLAC_DEC (client_data);
|
|
guint len;
|
|
|
|
len = MIN (gst_adapter_available (dec->adapter), *bytes);
|
|
|
|
if (len == 0) {
|
|
GST_LOG_OBJECT (dec, "0 bytes available at the moment");
|
|
return FLAC__STREAM_DECODER_READ_STATUS_ABORT;
|
|
}
|
|
|
|
GST_LOG_OBJECT (dec, "feeding %u bytes to decoder "
|
|
"(available=%" G_GSIZE_FORMAT ", bytes=%u)",
|
|
len, gst_adapter_available (dec->adapter), (guint) * bytes);
|
|
gst_adapter_copy (dec->adapter, buffer, 0, len);
|
|
*bytes = len;
|
|
|
|
gst_adapter_flush (dec->adapter, len);
|
|
|
|
return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
|
|
}
|
|
|
|
static FLAC__StreamDecoderWriteStatus
|
|
gst_flac_dec_write (GstFlacDec * flacdec, const FLAC__Frame * frame,
|
|
const FLAC__int32 * const buffer[])
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstBuffer *outbuf;
|
|
guint depth = frame->header.bits_per_sample;
|
|
guint width, gdepth;
|
|
guint sample_rate = frame->header.sample_rate;
|
|
guint channels = frame->header.channels;
|
|
guint samples = frame->header.blocksize;
|
|
guint j, i;
|
|
GstMapInfo map;
|
|
gboolean caps_changed;
|
|
GstAudioChannelPosition chanpos[8];
|
|
|
|
GST_LOG_OBJECT (flacdec, "samples in frame header: %d", samples);
|
|
|
|
if (depth == 0) {
|
|
if (flacdec->depth < 4 || flacdec->depth > 32) {
|
|
GST_ERROR_OBJECT (flacdec, "unsupported depth %d from STREAMINFO",
|
|
flacdec->depth);
|
|
ret = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
|
|
depth = flacdec->depth;
|
|
}
|
|
|
|
switch (depth) {
|
|
case 8:
|
|
gdepth = width = 8;
|
|
break;
|
|
case 12:
|
|
case 16:
|
|
gdepth = width = 16;
|
|
break;
|
|
case 20:
|
|
case 24:
|
|
gdepth = 24;
|
|
width = 32;
|
|
break;
|
|
case 32:
|
|
gdepth = width = 32;
|
|
break;
|
|
default:
|
|
GST_ERROR_OBJECT (flacdec, "unsupported depth %d", depth);
|
|
ret = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
|
|
if (sample_rate == 0) {
|
|
if (flacdec->info.rate != 0) {
|
|
sample_rate = flacdec->info.rate;
|
|
} else {
|
|
GST_ERROR_OBJECT (flacdec, "unknown sample rate");
|
|
ret = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
caps_changed = (sample_rate != GST_AUDIO_INFO_RATE (&flacdec->info))
|
|
|| (width != GST_AUDIO_INFO_WIDTH (&flacdec->info))
|
|
|| (gdepth != GST_AUDIO_INFO_DEPTH (&flacdec->info))
|
|
|| (channels != GST_AUDIO_INFO_CHANNELS (&flacdec->info));
|
|
|
|
if (caps_changed
|
|
|| !gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (flacdec))) {
|
|
GST_DEBUG_OBJECT (flacdec, "Negotiating %d Hz @ %d channels", sample_rate,
|
|
channels);
|
|
|
|
memcpy (chanpos, channel_positions[flacdec->info.channels - 1],
|
|
sizeof (chanpos));
|
|
gst_audio_channel_positions_to_valid_order (chanpos,
|
|
flacdec->info.channels);
|
|
gst_audio_info_set_format (&flacdec->info,
|
|
gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth),
|
|
sample_rate, channels, chanpos);
|
|
|
|
/* Note: we create the inverse reordering map here */
|
|
gst_audio_get_channel_reorder_map (flacdec->info.channels,
|
|
flacdec->info.position, channel_positions[flacdec->info.channels - 1],
|
|
flacdec->channel_reorder_map);
|
|
|
|
flacdec->depth = depth;
|
|
|
|
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (flacdec),
|
|
&flacdec->info);
|
|
}
|
|
|
|
outbuf =
|
|
gst_buffer_new_allocate (NULL, samples * channels * (width / 8), NULL);
|
|
|
|
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
|
|
if (width == 8) {
|
|
gint8 *outbuffer = (gint8 *) map.data;
|
|
gint *reorder_map = flacdec->channel_reorder_map;
|
|
|
|
g_assert (gdepth == 8 && depth == 8);
|
|
for (i = 0; i < samples; i++) {
|
|
for (j = 0; j < channels; j++) {
|
|
*outbuffer++ = (gint8) buffer[reorder_map[j]][i];
|
|
}
|
|
}
|
|
} else if (width == 16) {
|
|
gint16 *outbuffer = (gint16 *) map.data;
|
|
gint *reorder_map = flacdec->channel_reorder_map;
|
|
|
|
if (gdepth != depth) {
|
|
for (i = 0; i < samples; i++) {
|
|
for (j = 0; j < channels; j++) {
|
|
*outbuffer++ =
|
|
(gint16) (buffer[reorder_map[j]][i] << (gdepth - depth));
|
|
}
|
|
}
|
|
} else {
|
|
for (i = 0; i < samples; i++) {
|
|
for (j = 0; j < channels; j++) {
|
|
*outbuffer++ = (gint16) buffer[reorder_map[j]][i];
|
|
}
|
|
}
|
|
}
|
|
} else if (width == 32) {
|
|
gint32 *outbuffer = (gint32 *) map.data;
|
|
gint *reorder_map = flacdec->channel_reorder_map;
|
|
|
|
if (gdepth != depth) {
|
|
for (i = 0; i < samples; i++) {
|
|
for (j = 0; j < channels; j++) {
|
|
*outbuffer++ =
|
|
(gint32) (buffer[reorder_map[j]][i] << (gdepth - depth));
|
|
}
|
|
}
|
|
} else {
|
|
for (i = 0; i < samples; i++) {
|
|
for (j = 0; j < channels; j++) {
|
|
*outbuffer++ = (gint32) buffer[reorder_map[j]][i];
|
|
}
|
|
}
|
|
}
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
gst_buffer_unmap (outbuf, &map);
|
|
|
|
GST_DEBUG_OBJECT (flacdec, "pushing %d samples", samples);
|
|
if (flacdec->error_count)
|
|
flacdec->error_count--;
|
|
|
|
ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (flacdec), outbuf, 1);
|
|
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
|
|
GST_DEBUG_OBJECT (flacdec, "finish_frame flow %s", gst_flow_get_name (ret));
|
|
}
|
|
|
|
done:
|
|
|
|
/* we act on the flow return value later in the handle_frame function, as we
|
|
* don't want to mess up the internal decoder state by returning ABORT when
|
|
* the error is in fact non-fatal (like a pad in flushing mode) and we want
|
|
* to continue later. So just pretend everything's dandy and act later. */
|
|
flacdec->last_flow = ret;
|
|
|
|
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
|
|
}
|
|
|
|
static FLAC__StreamDecoderWriteStatus
|
|
gst_flac_dec_write_stream (const FLAC__StreamDecoder * decoder,
|
|
const FLAC__Frame * frame,
|
|
const FLAC__int32 * const buffer[], void *client_data)
|
|
{
|
|
return gst_flac_dec_write (GST_FLAC_DEC (client_data), frame, buffer);
|
|
}
|
|
|
|
static void
|
|
gst_flac_dec_flush (GstAudioDecoder * audio_dec, gboolean hard)
|
|
{
|
|
GstFlacDec *dec = GST_FLAC_DEC (audio_dec);
|
|
|
|
if (!hard) {
|
|
guint available = gst_adapter_available (dec->adapter);
|
|
|
|
if (available > 0) {
|
|
GST_INFO_OBJECT (dec, "draining, %u bytes left in adapter", available);
|
|
FLAC__stream_decoder_process_until_end_of_stream (dec->decoder);
|
|
}
|
|
}
|
|
|
|
FLAC__stream_decoder_flush (dec->decoder);
|
|
gst_adapter_clear (dec->adapter);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_flac_dec_handle_frame (GstAudioDecoder * audio_dec, GstBuffer * buf)
|
|
{
|
|
GstFlacDec *dec;
|
|
|
|
dec = GST_FLAC_DEC (audio_dec);
|
|
|
|
/* drain remaining data? */
|
|
if (G_UNLIKELY (buf == NULL)) {
|
|
gst_flac_dec_flush (audio_dec, FALSE);
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
GST_LOG_OBJECT (dec, "frame: ts %" GST_TIME_FORMAT ", flags 0x%04x, "
|
|
"%" G_GSIZE_FORMAT " bytes", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
|
GST_BUFFER_FLAGS (buf), gst_buffer_get_size (buf));
|
|
|
|
/* drop any in-stream headers, we've processed those in set_format already */
|
|
if (G_UNLIKELY (!dec->got_headers)) {
|
|
gboolean got_audio_frame;
|
|
GstMapInfo map;
|
|
|
|
/* check if this is a flac audio frame (rather than a header or junk) */
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
got_audio_frame = gst_flac_dec_scan_got_frame (dec, map.data, map.size);
|
|
gst_buffer_unmap (buf, &map);
|
|
|
|
if (!got_audio_frame) {
|
|
GST_INFO_OBJECT (dec, "dropping in-stream header, %" G_GSIZE_FORMAT " "
|
|
"bytes", map.size);
|
|
gst_audio_decoder_finish_frame (audio_dec, NULL, 1);
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
GST_INFO_OBJECT (dec, "first audio frame, got all in-stream headers now");
|
|
dec->got_headers = TRUE;
|
|
}
|
|
|
|
gst_adapter_push (dec->adapter, gst_buffer_ref (buf));
|
|
buf = NULL;
|
|
|
|
dec->last_flow = GST_FLOW_OK;
|
|
|
|
/* framed - there should always be enough data to decode something */
|
|
GST_LOG_OBJECT (dec, "%" G_GSIZE_FORMAT " bytes available",
|
|
gst_adapter_available (dec->adapter));
|
|
|
|
if (!FLAC__stream_decoder_process_single (dec->decoder)) {
|
|
GST_INFO_OBJECT (dec, "process_single failed");
|
|
if (FLAC__stream_decoder_get_state (dec->decoder) ==
|
|
FLAC__STREAM_DECODER_ABORTED) {
|
|
GST_WARNING_OBJECT (dec, "Read callback caused internal abort");
|
|
/* allow recovery */
|
|
gst_adapter_clear (dec->adapter);
|
|
FLAC__stream_decoder_flush (dec->decoder);
|
|
gst_flac_dec_handle_decoder_error (dec, TRUE);
|
|
}
|
|
}
|
|
|
|
return dec->last_flow;
|
|
}
|