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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1620 lines
53 KiB
C
1620 lines
53 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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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-flacenc
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* @title: flacenc
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* @see_also: #GstFlacDec
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*
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* flacenc encodes FLAC streams.
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* [FLAC](http://flac.sourceforge.net/) is a Free Lossless Audio Codec.
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* FLAC audio can directly be written into a file, or embedded into containers
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* such as oggmux or matroskamux.
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 audiotestsrc num-buffers=100 ! flacenc ! filesink location=beep.flac
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* ]| Encode a short sine wave into FLAC
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* |[
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* gst-launch-1.0 cdparanoiasrc mode=continuous ! queue ! audioconvert ! flacenc ! filesink location=cd.flac
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* ]| Rip a whole audio CD into a single FLAC file, with the track table saved as a CUE sheet inside the FLAC file
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* |[
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* gst-launch-1.0 cdparanoiasrc track=5 ! queue ! audioconvert ! flacenc ! filesink location=track5.flac
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* ]| Rip track 5 of an audio CD and encode it losslessly to a FLAC file
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*
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*/
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/* TODO: - We currently don't handle discontinuities in the stream in a useful
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* way and instead rely on the developer plugging in audiorate if
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* the stream contains discontinuities.
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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 <stdlib.h>
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#include <string.h>
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#include <gstflacenc.h>
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#include <gst/audio/audio.h>
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#include <gst/tag/tag.h>
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#include <gst/gsttagsetter.h>
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#include "gstflacelements.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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static GstStaticPadTemplate src_factory = 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 ("audio/x-flac")
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);
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enum
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{
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PROP_0,
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PROP_QUALITY,
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PROP_STREAMABLE_SUBSET,
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PROP_MID_SIDE_STEREO,
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PROP_LOOSE_MID_SIDE_STEREO,
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PROP_BLOCKSIZE,
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PROP_MAX_LPC_ORDER,
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PROP_QLP_COEFF_PRECISION,
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PROP_QLP_COEFF_PREC_SEARCH,
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PROP_ESCAPE_CODING,
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PROP_EXHAUSTIVE_MODEL_SEARCH,
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PROP_MIN_RESIDUAL_PARTITION_ORDER,
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PROP_MAX_RESIDUAL_PARTITION_ORDER,
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PROP_RICE_PARAMETER_SEARCH_DIST,
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PROP_PADDING,
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PROP_SEEKPOINTS
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};
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GST_DEBUG_CATEGORY_STATIC (flacenc_debug);
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#define GST_CAT_DEFAULT flacenc_debug
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#define gst_flac_enc_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstFlacEnc, gst_flac_enc, GST_TYPE_AUDIO_ENCODER,
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G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL)
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G_IMPLEMENT_INTERFACE (GST_TYPE_TOC_SETTER, NULL)
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);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (flacenc, "flacenc", GST_RANK_PRIMARY,
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GST_TYPE_FLAC_ENC, flac_element_init (plugin));
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static gboolean gst_flac_enc_start (GstAudioEncoder * enc);
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static gboolean gst_flac_enc_stop (GstAudioEncoder * enc);
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static gboolean gst_flac_enc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_flac_enc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static GstCaps *gst_flac_enc_getcaps (GstAudioEncoder * enc, GstCaps * filter);
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static gboolean gst_flac_enc_sink_event (GstAudioEncoder * enc,
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GstEvent * event);
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static gboolean gst_flac_enc_sink_query (GstAudioEncoder * enc,
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GstQuery * query);
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static void gst_flac_enc_finalize (GObject * object);
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static GstCaps *gst_flac_enc_generate_sink_caps (void);
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static gboolean gst_flac_enc_update_quality (GstFlacEnc * flacenc,
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gint quality);
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static void gst_flac_enc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_flac_enc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static FLAC__StreamEncoderWriteStatus
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gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
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const FLAC__byte buffer[], size_t bytes,
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unsigned samples, unsigned current_frame, void *client_data);
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static FLAC__StreamEncoderSeekStatus
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gst_flac_enc_seek_callback (const FLAC__StreamEncoder * encoder,
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FLAC__uint64 absolute_byte_offset, void *client_data);
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static FLAC__StreamEncoderTellStatus
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gst_flac_enc_tell_callback (const FLAC__StreamEncoder * encoder,
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FLAC__uint64 * absolute_byte_offset, void *client_data);
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typedef struct
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{
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gboolean exhaustive_model_search;
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gboolean escape_coding;
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gboolean mid_side;
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gboolean loose_mid_side;
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guint qlp_coeff_precision;
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gboolean qlp_coeff_prec_search;
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guint min_residual_partition_order;
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guint max_residual_partition_order;
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guint rice_parameter_search_dist;
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guint max_lpc_order;
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guint blocksize;
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}
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GstFlacEncParams;
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static const GstFlacEncParams flacenc_params[] = {
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{FALSE, FALSE, FALSE, FALSE, 0, FALSE, 2, 2, 0, 0, 1152},
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{FALSE, FALSE, TRUE, TRUE, 0, FALSE, 2, 2, 0, 0, 1152},
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{FALSE, FALSE, TRUE, FALSE, 0, FALSE, 0, 3, 0, 0, 1152},
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{FALSE, FALSE, FALSE, FALSE, 0, FALSE, 3, 3, 0, 6, 4608},
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{FALSE, FALSE, TRUE, TRUE, 0, FALSE, 3, 3, 0, 8, 4608},
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{FALSE, FALSE, TRUE, FALSE, 0, FALSE, 3, 3, 0, 8, 4608},
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{FALSE, FALSE, TRUE, FALSE, 0, FALSE, 0, 4, 0, 8, 4608},
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{TRUE, FALSE, TRUE, FALSE, 0, FALSE, 0, 6, 0, 8, 4608},
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{TRUE, FALSE, TRUE, FALSE, 0, FALSE, 0, 6, 0, 12, 4608},
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{TRUE, TRUE, TRUE, FALSE, 0, FALSE, 0, 16, 0, 32, 4608},
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};
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#define DEFAULT_QUALITY 5
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#define DEFAULT_PADDING 0
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#define DEFAULT_SEEKPOINTS -10
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#define GST_TYPE_FLAC_ENC_QUALITY (gst_flac_enc_quality_get_type ())
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static GType
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gst_flac_enc_quality_get_type (void)
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{
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static GType qtype = 0;
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if (qtype == 0) {
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static const GEnumValue values[] = {
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{0, "0 - Fastest compression", "0"},
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{1, "1", "1"},
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{2, "2", "2"},
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{3, "3", "3"},
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{4, "4", "4"},
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{5, "5 - Default", "5"},
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{6, "6", "6"},
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{7, "7", "7"},
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{8, "8 - Highest compression", "8"},
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{9, "9 - Insane", "9"},
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{0, NULL, NULL}
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};
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qtype = g_enum_register_static ("GstFlacEncQuality", values);
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}
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return qtype;
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}
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static void
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gst_flac_enc_class_init (GstFlacEncClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioEncoderClass *base_class;
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GstCaps *sink_caps;
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GstPadTemplate *sink_templ;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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base_class = (GstAudioEncoderClass *) (klass);
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GST_DEBUG_CATEGORY_INIT (flacenc_debug, "flacenc", 0,
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"Flac encoding element");
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gobject_class->set_property = gst_flac_enc_set_property;
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gobject_class->get_property = gst_flac_enc_get_property;
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gobject_class->finalize = gst_flac_enc_finalize;
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QUALITY,
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g_param_spec_enum ("quality",
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"Quality",
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"Speed versus compression tradeoff",
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GST_TYPE_FLAC_ENC_QUALITY, DEFAULT_QUALITY,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_STREAMABLE_SUBSET, g_param_spec_boolean ("streamable-subset",
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"Streamable subset",
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"true to limit encoder to generating a Subset stream, else false",
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TRUE,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MID_SIDE_STEREO,
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g_param_spec_boolean ("mid-side-stereo", "Do mid side stereo",
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"Do mid side stereo (only for stereo input)",
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flacenc_params[DEFAULT_QUALITY].mid_side,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_LOOSE_MID_SIDE_STEREO, g_param_spec_boolean ("loose-mid-side-stereo",
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"Loose mid side stereo", "Loose mid side stereo",
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flacenc_params[DEFAULT_QUALITY].loose_mid_side,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BLOCKSIZE,
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g_param_spec_uint ("blocksize", "Blocksize", "Blocksize in samples",
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FLAC__MIN_BLOCK_SIZE, FLAC__MAX_BLOCK_SIZE,
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flacenc_params[DEFAULT_QUALITY].blocksize,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_LPC_ORDER,
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g_param_spec_uint ("max-lpc-order", "Max LPC order",
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"Max LPC order; 0 => use only fixed predictors", 0,
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FLAC__MAX_LPC_ORDER, flacenc_params[DEFAULT_QUALITY].max_lpc_order,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_QLP_COEFF_PRECISION, g_param_spec_uint ("qlp-coeff-precision",
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"QLP coefficients precision",
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"Precision in bits of quantized linear-predictor coefficients; 0 = automatic",
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0, 32, flacenc_params[DEFAULT_QUALITY].qlp_coeff_precision,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_QLP_COEFF_PREC_SEARCH, g_param_spec_boolean ("qlp-coeff-prec-search",
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"Do QLP coefficients precision search",
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"false = use qlp_coeff_precision, "
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"true = search around qlp_coeff_precision, take best",
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flacenc_params[DEFAULT_QUALITY].qlp_coeff_prec_search,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_ESCAPE_CODING,
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g_param_spec_boolean ("escape-coding", "Do Escape coding",
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"search for escape codes in the entropy coding stage "
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"for slightly better compression",
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flacenc_params[DEFAULT_QUALITY].escape_coding,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_EXHAUSTIVE_MODEL_SEARCH,
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g_param_spec_boolean ("exhaustive-model-search",
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"Do exhaustive model search",
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"do exhaustive search of LP coefficient quantization (expensive!)",
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flacenc_params[DEFAULT_QUALITY].exhaustive_model_search,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_MIN_RESIDUAL_PARTITION_ORDER,
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g_param_spec_uint ("min-residual-partition-order",
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"Min residual partition order",
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"Min residual partition order (above 4 doesn't usually help much)", 0,
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16, flacenc_params[DEFAULT_QUALITY].min_residual_partition_order,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_MAX_RESIDUAL_PARTITION_ORDER,
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g_param_spec_uint ("max-residual-partition-order",
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"Max residual partition order",
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"Max residual partition order (above 4 doesn't usually help much)", 0,
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16, flacenc_params[DEFAULT_QUALITY].max_residual_partition_order,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_RICE_PARAMETER_SEARCH_DIST,
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g_param_spec_uint ("rice-parameter-search-dist",
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"rice_parameter_search_dist",
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"0 = try only calc'd parameter k; else try all [k-dist..k+dist] "
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"parameters, use best", 0, FLAC__MAX_RICE_PARTITION_ORDER,
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flacenc_params[DEFAULT_QUALITY].rice_parameter_search_dist,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_PADDING,
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g_param_spec_uint ("padding",
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"Padding",
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"Write a PADDING block with this length in bytes", 0, G_MAXUINT,
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DEFAULT_PADDING,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_SEEKPOINTS,
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g_param_spec_int ("seekpoints",
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"Seekpoints",
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"Add SEEKTABLE metadata (if > 0, number of entries, if < 0, interval in sec)",
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-G_MAXINT, G_MAXINT,
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DEFAULT_SEEKPOINTS,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
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sink_caps = gst_flac_enc_generate_sink_caps ();
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sink_templ = gst_pad_template_new ("sink",
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GST_PAD_SINK, GST_PAD_ALWAYS, sink_caps);
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gst_element_class_add_pad_template (gstelement_class, sink_templ);
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gst_caps_unref (sink_caps);
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gst_element_class_set_static_metadata (gstelement_class, "FLAC audio encoder",
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"Codec/Encoder/Audio",
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"Encodes audio with the FLAC lossless audio encoder",
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"Wim Taymans <wim.taymans@chello.be>");
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base_class->start = GST_DEBUG_FUNCPTR (gst_flac_enc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_flac_enc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_flac_enc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_flac_enc_handle_frame);
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base_class->getcaps = GST_DEBUG_FUNCPTR (gst_flac_enc_getcaps);
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base_class->sink_event = GST_DEBUG_FUNCPTR (gst_flac_enc_sink_event);
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base_class->sink_query = GST_DEBUG_FUNCPTR (gst_flac_enc_sink_query);
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gst_type_mark_as_plugin_api (GST_TYPE_FLAC_ENC_QUALITY, 0);
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}
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static void
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gst_flac_enc_init (GstFlacEnc * flacenc)
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{
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GstAudioEncoder *enc = GST_AUDIO_ENCODER (flacenc);
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flacenc->encoder = FLAC__stream_encoder_new ();
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gst_flac_enc_update_quality (flacenc, DEFAULT_QUALITY);
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/* arrange granulepos marking (and required perfect ts) */
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gst_audio_encoder_set_mark_granule (enc, TRUE);
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gst_audio_encoder_set_perfect_timestamp (enc, TRUE);
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}
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static void
|
|
gst_flac_enc_finalize (GObject * object)
|
|
{
|
|
GstFlacEnc *flacenc = GST_FLAC_ENC (object);
|
|
|
|
FLAC__stream_encoder_delete (flacenc->encoder);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static gboolean
|
|
gst_flac_enc_start (GstAudioEncoder * enc)
|
|
{
|
|
GstFlacEnc *flacenc = GST_FLAC_ENC (enc);
|
|
|
|
GST_DEBUG_OBJECT (enc, "start");
|
|
flacenc->stopped = TRUE;
|
|
flacenc->got_headers = FALSE;
|
|
flacenc->last_flow = GST_FLOW_OK;
|
|
flacenc->offset = 0;
|
|
flacenc->eos = FALSE;
|
|
flacenc->tags = gst_tag_list_new_empty ();
|
|
flacenc->toc = NULL;
|
|
flacenc->samples_in = 0;
|
|
flacenc->samples_out = 0;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_flac_enc_stop (GstAudioEncoder * enc)
|
|
{
|
|
GstFlacEnc *flacenc = GST_FLAC_ENC (enc);
|
|
|
|
GST_DEBUG_OBJECT (enc, "stop");
|
|
gst_tag_list_unref (flacenc->tags);
|
|
flacenc->tags = NULL;
|
|
if (flacenc->toc)
|
|
gst_toc_unref (flacenc->toc);
|
|
flacenc->toc = NULL;
|
|
if (FLAC__stream_encoder_get_state (flacenc->encoder) !=
|
|
FLAC__STREAM_ENCODER_UNINITIALIZED) {
|
|
flacenc->stopped = TRUE;
|
|
FLAC__stream_encoder_finish (flacenc->encoder);
|
|
}
|
|
if (flacenc->meta) {
|
|
FLAC__metadata_object_delete (flacenc->meta[0]);
|
|
|
|
if (flacenc->meta[1])
|
|
FLAC__metadata_object_delete (flacenc->meta[1]);
|
|
|
|
if (flacenc->meta[2])
|
|
FLAC__metadata_object_delete (flacenc->meta[2]);
|
|
|
|
if (flacenc->meta[3])
|
|
FLAC__metadata_object_delete (flacenc->meta[3]);
|
|
|
|
g_free (flacenc->meta);
|
|
flacenc->meta = NULL;
|
|
}
|
|
g_list_foreach (flacenc->headers, (GFunc) gst_mini_object_unref, NULL);
|
|
g_list_free (flacenc->headers);
|
|
flacenc->headers = NULL;
|
|
|
|
gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
|
|
gst_toc_setter_reset (GST_TOC_SETTER (enc));
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
add_one_tag (const GstTagList * list, const gchar * tag, gpointer user_data)
|
|
{
|
|
GList *comments;
|
|
GList *it;
|
|
GstFlacEnc *flacenc = GST_FLAC_ENC (user_data);
|
|
|
|
/* IMAGE and PREVIEW_IMAGE tags are already written
|
|
* differently, no need to store them inside the
|
|
* vorbiscomments too */
|
|
if (strcmp (tag, GST_TAG_IMAGE) == 0
|
|
|| strcmp (tag, GST_TAG_PREVIEW_IMAGE) == 0)
|
|
return;
|
|
|
|
comments = gst_tag_to_vorbis_comments (list, tag);
|
|
for (it = comments; it != NULL; it = it->next) {
|
|
FLAC__StreamMetadata_VorbisComment_Entry commment_entry;
|
|
|
|
commment_entry.length = strlen (it->data);
|
|
commment_entry.entry = it->data;
|
|
FLAC__metadata_object_vorbiscomment_insert_comment (flacenc->meta[0],
|
|
flacenc->meta[0]->data.vorbis_comment.num_comments,
|
|
commment_entry, TRUE);
|
|
g_free (it->data);
|
|
}
|
|
g_list_free (comments);
|
|
}
|
|
|
|
static gboolean
|
|
add_cuesheet (const GstToc * toc, guint sample_rate,
|
|
FLAC__StreamMetadata * cuesheet)
|
|
{
|
|
gint8 track_num = 0;
|
|
gint64 start, stop;
|
|
gchar *isrc = NULL;
|
|
const gchar *is_legal;
|
|
GList *list;
|
|
GstTagList *tags;
|
|
GstTocEntry *entry, *subentry = NULL;
|
|
FLAC__StreamMetadata_CueSheet *cs;
|
|
FLAC__StreamMetadata_CueSheet_Track *track;
|
|
|
|
cs = &cuesheet->data.cue_sheet;
|
|
if (!cs)
|
|
return FALSE;
|
|
|
|
/* check if the TOC entries is valid */
|
|
list = gst_toc_get_entries (toc);
|
|
entry = list->data;
|
|
if (gst_toc_entry_is_alternative (entry)) {
|
|
list = gst_toc_entry_get_sub_entries (entry);
|
|
while (list) {
|
|
subentry = list->data;
|
|
if (!gst_toc_entry_is_sequence (subentry))
|
|
return FALSE;
|
|
list = g_list_next (list);
|
|
}
|
|
list = gst_toc_entry_get_sub_entries (entry);
|
|
}
|
|
if (gst_toc_entry_is_sequence (entry)) {
|
|
while (list) {
|
|
entry = list->data;
|
|
if (!gst_toc_entry_is_sequence (entry))
|
|
return FALSE;
|
|
list = g_list_next (list);
|
|
}
|
|
list = gst_toc_get_entries (toc);
|
|
}
|
|
|
|
/* add tracks in cuesheet */
|
|
while (list) {
|
|
entry = list->data;
|
|
gst_toc_entry_get_start_stop_times (entry, &start, &stop);
|
|
tags = gst_toc_entry_get_tags (entry);
|
|
if (tags)
|
|
gst_tag_list_get_string (tags, GST_TAG_ISRC, &isrc);
|
|
track = FLAC__metadata_object_cuesheet_track_new ();
|
|
track->offset =
|
|
(FLAC__uint64) gst_util_uint64_scale_round (start, sample_rate,
|
|
GST_SECOND);
|
|
track->number = (FLAC__byte) track_num + 1;
|
|
if (isrc != NULL && strlen (isrc) <= 12)
|
|
g_strlcpy (track->isrc, isrc, 13);
|
|
if (track->number <= 0)
|
|
return FALSE;
|
|
if (!FLAC__metadata_object_cuesheet_insert_track (cuesheet, track_num,
|
|
track, FALSE))
|
|
return FALSE;
|
|
if (!FLAC__metadata_object_cuesheet_track_insert_blank_index (cuesheet,
|
|
track_num, 0))
|
|
return FALSE;
|
|
track_num++;
|
|
list = g_list_next (list);
|
|
}
|
|
|
|
if (cs->num_tracks <= 0)
|
|
return FALSE;
|
|
|
|
/* add lead-out track in cuesheet */
|
|
track = FLAC__metadata_object_cuesheet_track_new ();
|
|
track->offset =
|
|
(FLAC__uint64) gst_util_uint64_scale_round (stop, sample_rate,
|
|
GST_SECOND);
|
|
track->number = 255;
|
|
if (!FLAC__metadata_object_cuesheet_insert_track (cuesheet, cs->num_tracks,
|
|
track, FALSE))
|
|
return FALSE;
|
|
|
|
/* check if the cuesheet is valid */
|
|
if (!FLAC__metadata_object_cuesheet_is_legal (cuesheet, FALSE, &is_legal)) {
|
|
g_warning ("%s\n", is_legal);
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_flac_enc_set_metadata (GstFlacEnc * flacenc, GstAudioInfo * info,
|
|
guint64 total_samples)
|
|
{
|
|
const GstTagList *user_tags;
|
|
GstTagList *copy;
|
|
gint entries = 1;
|
|
gint n_images, n_preview_images;
|
|
FLAC__StreamMetadata *cuesheet;
|
|
|
|
g_return_if_fail (flacenc != NULL);
|
|
|
|
user_tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (flacenc));
|
|
if ((flacenc->tags == NULL) && (user_tags == NULL)) {
|
|
return;
|
|
}
|
|
copy = gst_tag_list_merge (user_tags, flacenc->tags,
|
|
gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (flacenc)));
|
|
n_images = gst_tag_list_get_tag_size (copy, GST_TAG_IMAGE);
|
|
n_preview_images = gst_tag_list_get_tag_size (copy, GST_TAG_PREVIEW_IMAGE);
|
|
|
|
flacenc->meta =
|
|
g_new0 (FLAC__StreamMetadata *, 4 + n_images + n_preview_images);
|
|
|
|
flacenc->meta[0] =
|
|
FLAC__metadata_object_new (FLAC__METADATA_TYPE_VORBIS_COMMENT);
|
|
gst_tag_list_foreach (copy, add_one_tag, flacenc);
|
|
|
|
if (!flacenc->toc)
|
|
flacenc->toc = gst_toc_setter_get_toc (GST_TOC_SETTER (flacenc));
|
|
|
|
if (flacenc->toc) {
|
|
cuesheet = FLAC__metadata_object_new (FLAC__METADATA_TYPE_CUESHEET);
|
|
if (add_cuesheet (flacenc->toc, GST_AUDIO_INFO_RATE (info), cuesheet)) {
|
|
flacenc->meta[entries] = cuesheet;
|
|
entries++;
|
|
} else {
|
|
FLAC__metadata_object_delete (cuesheet);
|
|
flacenc->meta[entries] = NULL;
|
|
}
|
|
}
|
|
|
|
if (n_images + n_preview_images > 0) {
|
|
GstSample *sample;
|
|
GstBuffer *buffer;
|
|
GstCaps *caps;
|
|
const GstStructure *structure;
|
|
GstTagImageType image_type = GST_TAG_IMAGE_TYPE_NONE;
|
|
gint i, width = 0, height = 0, png_icon_count = 0, other_icon_count = 0;
|
|
GstMapInfo map;
|
|
|
|
for (i = 0; i < n_images + n_preview_images; i++) {
|
|
gboolean is_preview_image = (i >= n_images);
|
|
|
|
if (i < n_images) {
|
|
if (!gst_tag_list_get_sample_index (copy, GST_TAG_IMAGE, i, &sample))
|
|
continue;
|
|
} else {
|
|
if (!gst_tag_list_get_sample_index (copy, GST_TAG_PREVIEW_IMAGE,
|
|
i - n_images, &sample))
|
|
continue;
|
|
}
|
|
|
|
structure = gst_sample_get_info (sample);
|
|
caps = gst_sample_get_caps (sample);
|
|
if (!caps) {
|
|
GST_FIXME_OBJECT (flacenc, "Image tag without caps");
|
|
gst_sample_unref (sample);
|
|
continue;
|
|
}
|
|
|
|
flacenc->meta[entries] =
|
|
FLAC__metadata_object_new (FLAC__METADATA_TYPE_PICTURE);
|
|
|
|
GST_LOG_OBJECT (flacenc, "image info: %" GST_PTR_FORMAT, structure);
|
|
|
|
if (structure)
|
|
gst_structure_get (structure, "image-type", GST_TYPE_TAG_IMAGE_TYPE,
|
|
&image_type, NULL);
|
|
else
|
|
image_type = GST_TAG_IMAGE_TYPE_NONE;
|
|
|
|
GST_LOG_OBJECT (flacenc, "image caps: %" GST_PTR_FORMAT, caps);
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
gst_structure_get (structure, "width", G_TYPE_INT, &width,
|
|
"height", G_TYPE_INT, &height, NULL);
|
|
|
|
/* Convert to ID3v2 APIC image type */
|
|
if (image_type == GST_TAG_IMAGE_TYPE_NONE) {
|
|
if (is_preview_image) {
|
|
/* 1 - 32x32 pixels 'file icon' (PNG only)
|
|
* 2 - Other file icon
|
|
* There may only be one each of picture type 1 and 2 in a file. */
|
|
if (width == 32 && height == 32
|
|
&& gst_structure_has_name (structure, "image/png")
|
|
&& png_icon_count++ == 0) {
|
|
image_type = 1;
|
|
} else if (width <= 32 && height <= 32 && other_icon_count++ == 0) {
|
|
image_type = 2;
|
|
} else {
|
|
image_type = 0; /* Other */
|
|
}
|
|
} else {
|
|
image_type = 0; /* Other */
|
|
}
|
|
} else {
|
|
/* GStreamer enum is the same but without the two icon types 1+2 */
|
|
image_type = image_type + 2;
|
|
}
|
|
|
|
buffer = gst_sample_get_buffer (sample);
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
FLAC__metadata_object_picture_set_data (flacenc->meta[entries],
|
|
map.data, map.size, TRUE);
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
GST_LOG_OBJECT (flacenc, "Setting picture type %d", image_type);
|
|
flacenc->meta[entries]->data.picture.type = image_type;
|
|
|
|
if (width > 0 && height > 0) {
|
|
flacenc->meta[entries]->data.picture.width = width;
|
|
flacenc->meta[entries]->data.picture.height = height;
|
|
}
|
|
|
|
FLAC__metadata_object_picture_set_mime_type (flacenc->meta[entries],
|
|
(char *) gst_structure_get_name (structure), TRUE);
|
|
|
|
gst_sample_unref (sample);
|
|
entries++;
|
|
}
|
|
}
|
|
|
|
if (flacenc->seekpoints && total_samples != GST_CLOCK_TIME_NONE) {
|
|
gboolean res;
|
|
guint samples;
|
|
|
|
flacenc->meta[entries] =
|
|
FLAC__metadata_object_new (FLAC__METADATA_TYPE_SEEKTABLE);
|
|
if (flacenc->seekpoints > 0) {
|
|
res =
|
|
FLAC__metadata_object_seektable_template_append_spaced_points
|
|
(flacenc->meta[entries], flacenc->seekpoints, total_samples);
|
|
} else {
|
|
samples = -flacenc->seekpoints * GST_AUDIO_INFO_RATE (info);
|
|
res =
|
|
FLAC__metadata_object_seektable_template_append_spaced_points_by_samples
|
|
(flacenc->meta[entries], samples, total_samples);
|
|
}
|
|
if (!res) {
|
|
GST_DEBUG_OBJECT (flacenc, "adding seekpoint template %d failed",
|
|
flacenc->seekpoints);
|
|
FLAC__metadata_object_delete (flacenc->meta[1]);
|
|
flacenc->meta[entries] = NULL;
|
|
} else {
|
|
entries++;
|
|
}
|
|
} else if (flacenc->seekpoints && total_samples == GST_CLOCK_TIME_NONE) {
|
|
GST_WARNING_OBJECT (flacenc, "total time unknown; can not add seekpoints");
|
|
}
|
|
|
|
if (flacenc->padding > 0) {
|
|
flacenc->meta[entries] =
|
|
FLAC__metadata_object_new (FLAC__METADATA_TYPE_PADDING);
|
|
flacenc->meta[entries]->length = flacenc->padding;
|
|
entries++;
|
|
}
|
|
|
|
if (FLAC__stream_encoder_set_metadata (flacenc->encoder,
|
|
flacenc->meta, entries) != true)
|
|
g_warning ("Dude, i'm already initialized!");
|
|
|
|
gst_tag_list_unref (copy);
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_flac_enc_generate_sink_caps (void)
|
|
{
|
|
GstCaps *ret;
|
|
gint i;
|
|
GValue v_list = { 0, };
|
|
GValue v = { 0, };
|
|
GstStructure *s, *s2;
|
|
|
|
g_value_init (&v_list, GST_TYPE_LIST);
|
|
g_value_init (&v, G_TYPE_STRING);
|
|
|
|
/* Use system's endianness */
|
|
g_value_set_static_string (&v, "S8");
|
|
gst_value_list_append_value (&v_list, &v);
|
|
g_value_set_static_string (&v, GST_AUDIO_NE (S16));
|
|
gst_value_list_append_value (&v_list, &v);
|
|
g_value_set_static_string (&v, GST_AUDIO_NE (S24));
|
|
gst_value_list_append_value (&v_list, &v);
|
|
g_value_set_static_string (&v, GST_AUDIO_NE (S24_32));
|
|
gst_value_list_append_value (&v_list, &v);
|
|
g_value_unset (&v);
|
|
|
|
s = gst_structure_new_empty ("audio/x-raw");
|
|
gst_structure_take_value (s, "format", &v_list);
|
|
|
|
gst_structure_set (s, "layout", G_TYPE_STRING, "interleaved",
|
|
"rate", GST_TYPE_INT_RANGE, 1, 655350, NULL);
|
|
|
|
ret = gst_caps_new_empty ();
|
|
s2 = gst_structure_copy (s);
|
|
gst_structure_set (s2, "channels", G_TYPE_INT, 1, NULL);
|
|
gst_caps_append_structure (ret, s2);
|
|
for (i = 2; i <= 8; i++) {
|
|
guint64 channel_mask;
|
|
|
|
s2 = gst_structure_copy (s);
|
|
gst_audio_channel_positions_to_mask (channel_positions[i - 1], i,
|
|
FALSE, &channel_mask);
|
|
gst_structure_set (s2, "channels", G_TYPE_INT, i, "channel-mask",
|
|
GST_TYPE_BITMASK, channel_mask, NULL);
|
|
|
|
gst_caps_append_structure (ret, s2);
|
|
}
|
|
gst_structure_free (s);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_flac_enc_getcaps (GstAudioEncoder * enc, GstCaps * filter)
|
|
{
|
|
GstCaps *ret = NULL, *caps = NULL;
|
|
GstPad *pad;
|
|
|
|
pad = GST_AUDIO_ENCODER_SINK_PAD (enc);
|
|
|
|
ret = gst_pad_get_current_caps (pad);
|
|
if (ret == NULL) {
|
|
ret = gst_pad_get_pad_template_caps (pad);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (pad, "Return caps %" GST_PTR_FORMAT, ret);
|
|
|
|
caps = gst_audio_encoder_proxy_getcaps (enc, ret, filter);
|
|
gst_caps_unref (ret);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static guint64
|
|
gst_flac_enc_peer_query_total_samples (GstFlacEnc * flacenc, GstPad * pad,
|
|
GstAudioInfo * info)
|
|
{
|
|
gint64 duration;
|
|
|
|
GST_DEBUG_OBJECT (flacenc, "querying peer for DEFAULT format duration");
|
|
if (gst_pad_peer_query_duration (pad, GST_FORMAT_DEFAULT, &duration)
|
|
&& duration != GST_CLOCK_TIME_NONE)
|
|
goto done;
|
|
|
|
GST_DEBUG_OBJECT (flacenc, "querying peer for TIME format duration");
|
|
|
|
if (gst_pad_peer_query_duration (pad, GST_FORMAT_TIME, &duration)
|
|
&& duration != GST_CLOCK_TIME_NONE) {
|
|
GST_DEBUG_OBJECT (flacenc, "peer reported duration %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (duration));
|
|
duration = GST_CLOCK_TIME_TO_FRAMES (duration, GST_AUDIO_INFO_RATE (info));
|
|
|
|
goto done;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (flacenc, "Upstream reported no total samples");
|
|
return GST_CLOCK_TIME_NONE;
|
|
|
|
done:
|
|
GST_DEBUG_OBJECT (flacenc,
|
|
"Upstream reported %" G_GUINT64_FORMAT " total samples", duration);
|
|
|
|
return duration;
|
|
}
|
|
|
|
static gint64
|
|
gst_flac_enc_get_latency (GstFlacEnc * flacenc)
|
|
{
|
|
/* The FLAC specification states that the data is processed in blocks,
|
|
* regardless of the number of channels. Thus, The latency can be calculated
|
|
* using the blocksize and rate. For example a 1 second block sampled at
|
|
* 44.1KHz has a blocksize of 44100 */
|
|
|
|
/* Get the blocksize */
|
|
const guint blocksize = FLAC__stream_encoder_get_blocksize (flacenc->encoder);
|
|
|
|
/* Get the sample rate in KHz */
|
|
const guint rate = FLAC__stream_encoder_get_sample_rate (flacenc->encoder);
|
|
if (!rate)
|
|
return 0;
|
|
|
|
/* Calculate the latecy */
|
|
return (blocksize * GST_SECOND) / rate;
|
|
}
|
|
|
|
static gboolean
|
|
gst_flac_enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
|
|
{
|
|
GstFlacEnc *flacenc;
|
|
guint64 total_samples = GST_CLOCK_TIME_NONE;
|
|
FLAC__StreamEncoderInitStatus init_status;
|
|
|
|
flacenc = GST_FLAC_ENC (enc);
|
|
|
|
/* if configured again, means something changed, can't handle that */
|
|
if (FLAC__stream_encoder_get_state (flacenc->encoder) !=
|
|
FLAC__STREAM_ENCODER_UNINITIALIZED)
|
|
goto encoder_already_initialized;
|
|
|
|
/* delay setting output caps/format until we have all headers */
|
|
|
|
gst_audio_get_channel_reorder_map (GST_AUDIO_INFO_CHANNELS (info),
|
|
channel_positions[GST_AUDIO_INFO_CHANNELS (info) - 1], info->position,
|
|
flacenc->channel_reorder_map);
|
|
|
|
total_samples = gst_flac_enc_peer_query_total_samples (flacenc,
|
|
GST_AUDIO_ENCODER_SINK_PAD (enc), info);
|
|
|
|
FLAC__stream_encoder_set_bits_per_sample (flacenc->encoder,
|
|
GST_AUDIO_INFO_DEPTH (info));
|
|
FLAC__stream_encoder_set_sample_rate (flacenc->encoder,
|
|
GST_AUDIO_INFO_RATE (info));
|
|
FLAC__stream_encoder_set_channels (flacenc->encoder,
|
|
GST_AUDIO_INFO_CHANNELS (info));
|
|
|
|
if (total_samples != GST_CLOCK_TIME_NONE)
|
|
FLAC__stream_encoder_set_total_samples_estimate (flacenc->encoder,
|
|
MIN (total_samples, G_GUINT64_CONSTANT (0x0FFFFFFFFF)));
|
|
|
|
gst_flac_enc_set_metadata (flacenc, info, total_samples);
|
|
|
|
/* callbacks clear to go now;
|
|
* write callbacks receives headers during init */
|
|
flacenc->stopped = FALSE;
|
|
|
|
init_status = FLAC__stream_encoder_init_stream (flacenc->encoder,
|
|
gst_flac_enc_write_callback, gst_flac_enc_seek_callback,
|
|
gst_flac_enc_tell_callback, NULL, flacenc);
|
|
if (init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK)
|
|
goto failed_to_initialize;
|
|
|
|
/* feedback to base class */
|
|
gst_audio_encoder_set_latency (enc,
|
|
gst_flac_enc_get_latency (flacenc), gst_flac_enc_get_latency (flacenc));
|
|
|
|
return TRUE;
|
|
|
|
encoder_already_initialized:
|
|
{
|
|
g_warning ("flac already initialized -- fixme allow this");
|
|
return FALSE;
|
|
}
|
|
failed_to_initialize:
|
|
{
|
|
GST_ELEMENT_ERROR (flacenc, LIBRARY, INIT, (NULL),
|
|
("could not initialize encoder (wrong parameters?) %d", init_status));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_flac_enc_update_quality (GstFlacEnc * flacenc, gint quality)
|
|
{
|
|
GstAudioInfo *info =
|
|
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (flacenc));
|
|
|
|
flacenc->quality = quality;
|
|
|
|
#define DO_UPDATE(name, val, str) \
|
|
G_STMT_START { \
|
|
if (FLAC__stream_encoder_get_##name (flacenc->encoder) != \
|
|
flacenc_params[quality].val) { \
|
|
FLAC__stream_encoder_set_##name (flacenc->encoder, \
|
|
flacenc_params[quality].val); \
|
|
g_object_notify (G_OBJECT (flacenc), str); \
|
|
} \
|
|
} G_STMT_END
|
|
|
|
g_object_freeze_notify (G_OBJECT (flacenc));
|
|
|
|
if (GST_AUDIO_INFO_CHANNELS (info) == 2
|
|
|| GST_AUDIO_INFO_CHANNELS (info) == 0) {
|
|
DO_UPDATE (do_mid_side_stereo, mid_side, "mid_side_stereo");
|
|
DO_UPDATE (loose_mid_side_stereo, loose_mid_side, "loose_mid_side");
|
|
}
|
|
|
|
DO_UPDATE (blocksize, blocksize, "blocksize");
|
|
DO_UPDATE (max_lpc_order, max_lpc_order, "max_lpc_order");
|
|
DO_UPDATE (qlp_coeff_precision, qlp_coeff_precision, "qlp_coeff_precision");
|
|
DO_UPDATE (do_qlp_coeff_prec_search, qlp_coeff_prec_search,
|
|
"qlp_coeff_prec_search");
|
|
DO_UPDATE (do_escape_coding, escape_coding, "escape_coding");
|
|
DO_UPDATE (do_exhaustive_model_search, exhaustive_model_search,
|
|
"exhaustive_model_search");
|
|
DO_UPDATE (min_residual_partition_order, min_residual_partition_order,
|
|
"min_residual_partition_order");
|
|
DO_UPDATE (max_residual_partition_order, max_residual_partition_order,
|
|
"max_residual_partition_order");
|
|
DO_UPDATE (rice_parameter_search_dist, rice_parameter_search_dist,
|
|
"rice_parameter_search_dist");
|
|
|
|
#undef DO_UPDATE
|
|
|
|
g_object_thaw_notify (G_OBJECT (flacenc));
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static FLAC__StreamEncoderSeekStatus
|
|
gst_flac_enc_seek_callback (const FLAC__StreamEncoder * encoder,
|
|
FLAC__uint64 absolute_byte_offset, void *client_data)
|
|
{
|
|
GstFlacEnc *flacenc;
|
|
GstPad *peerpad;
|
|
GstSegment seg;
|
|
|
|
flacenc = GST_FLAC_ENC (client_data);
|
|
|
|
if (flacenc->stopped)
|
|
return FLAC__STREAM_ENCODER_SEEK_STATUS_OK;
|
|
|
|
if ((peerpad = gst_pad_get_peer (GST_AUDIO_ENCODER_SRC_PAD (flacenc)))) {
|
|
GstEvent *event;
|
|
gboolean ret;
|
|
GstQuery *query;
|
|
gboolean seekable = FALSE;
|
|
|
|
/* try to seek to the beginning of the output */
|
|
query = gst_query_new_seeking (GST_FORMAT_BYTES);
|
|
if (gst_pad_query (peerpad, query)) {
|
|
GstFormat format;
|
|
|
|
gst_query_parse_seeking (query, &format, &seekable, NULL, NULL);
|
|
if (format != GST_FORMAT_BYTES)
|
|
seekable = FALSE;
|
|
} else {
|
|
GST_LOG_OBJECT (flacenc, "SEEKING query not handled");
|
|
}
|
|
gst_query_unref (query);
|
|
|
|
if (!seekable) {
|
|
GST_DEBUG_OBJECT (flacenc, "downstream not seekable; not rewriting");
|
|
gst_object_unref (peerpad);
|
|
return FLAC__STREAM_ENCODER_SEEK_STATUS_UNSUPPORTED;
|
|
}
|
|
|
|
gst_segment_init (&seg, GST_FORMAT_BYTES);
|
|
seg.start = absolute_byte_offset;
|
|
seg.stop = GST_BUFFER_OFFSET_NONE;
|
|
seg.time = 0;
|
|
event = gst_event_new_segment (&seg);
|
|
|
|
ret = gst_pad_send_event (peerpad, event);
|
|
gst_object_unref (peerpad);
|
|
|
|
if (ret) {
|
|
GST_DEBUG ("Seek to %" G_GUINT64_FORMAT " %s",
|
|
(guint64) absolute_byte_offset, "succeeded");
|
|
} else {
|
|
GST_DEBUG ("Seek to %" G_GUINT64_FORMAT " %s",
|
|
(guint64) absolute_byte_offset, "failed");
|
|
return FLAC__STREAM_ENCODER_SEEK_STATUS_UNSUPPORTED;
|
|
}
|
|
} else {
|
|
GST_DEBUG ("Seek to %" G_GUINT64_FORMAT " failed (no peer pad)",
|
|
(guint64) absolute_byte_offset);
|
|
}
|
|
|
|
flacenc->offset = absolute_byte_offset;
|
|
return FLAC__STREAM_ENCODER_SEEK_STATUS_OK;
|
|
}
|
|
|
|
static void
|
|
notgst_value_array_append_buffer (GValue * array_val, GstBuffer * buf)
|
|
{
|
|
GValue value = { 0, };
|
|
|
|
g_value_init (&value, GST_TYPE_BUFFER);
|
|
/* copy buffer to avoid problems with circular refcounts */
|
|
buf = gst_buffer_copy (buf);
|
|
/* again, for good measure */
|
|
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
|
|
gst_value_set_buffer (&value, buf);
|
|
gst_buffer_unref (buf);
|
|
gst_value_array_append_value (array_val, &value);
|
|
g_value_unset (&value);
|
|
}
|
|
|
|
#define HDR_TYPE_STREAMINFO 0
|
|
#define HDR_TYPE_VORBISCOMMENT 4
|
|
|
|
static GstFlowReturn
|
|
gst_flac_enc_process_stream_headers (GstFlacEnc * enc)
|
|
{
|
|
GstBuffer *vorbiscomment = NULL;
|
|
GstBuffer *streaminfo = NULL;
|
|
GstBuffer *marker = NULL;
|
|
GValue array = { 0, };
|
|
GstCaps *caps;
|
|
GList *l;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstAudioInfo *info =
|
|
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (enc));
|
|
|
|
caps = gst_caps_new_simple ("audio/x-flac",
|
|
"channels", G_TYPE_INT, GST_AUDIO_INFO_CHANNELS (info),
|
|
"rate", G_TYPE_INT, GST_AUDIO_INFO_RATE (info), NULL);
|
|
|
|
for (l = enc->headers; l != NULL; l = l->next) {
|
|
GstBuffer *buf;
|
|
GstMapInfo map;
|
|
guint8 *data;
|
|
gsize size;
|
|
|
|
/* mark buffers so oggmux will ignore them if it already muxed the
|
|
* header buffers from the streamheaders field in the caps */
|
|
l->data = gst_buffer_make_writable (GST_BUFFER_CAST (l->data));
|
|
|
|
buf = GST_BUFFER_CAST (l->data);
|
|
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
|
|
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
data = map.data;
|
|
size = map.size;
|
|
|
|
/* find initial 4-byte marker which we need to skip later on */
|
|
if (size == 4 && memcmp (data, "fLaC", 4) == 0) {
|
|
marker = buf;
|
|
} else if (size > 1 && (data[0] & 0x7f) == HDR_TYPE_STREAMINFO) {
|
|
streaminfo = buf;
|
|
} else if (size > 1 && (data[0] & 0x7f) == HDR_TYPE_VORBISCOMMENT) {
|
|
vorbiscomment = buf;
|
|
}
|
|
|
|
gst_buffer_unmap (buf, &map);
|
|
}
|
|
|
|
if (marker == NULL || streaminfo == NULL || vorbiscomment == NULL) {
|
|
GST_WARNING_OBJECT (enc, "missing header %p %p %p, muxing into container "
|
|
"formats may be broken", marker, streaminfo, vorbiscomment);
|
|
goto push_headers;
|
|
}
|
|
|
|
g_value_init (&array, GST_TYPE_ARRAY);
|
|
|
|
/* add marker including STREAMINFO header */
|
|
{
|
|
GstBuffer *buf;
|
|
guint16 num;
|
|
GstMapInfo map;
|
|
guint8 *bdata;
|
|
gsize slen;
|
|
|
|
/* minus one for the marker that is merged with streaminfo here */
|
|
num = g_list_length (enc->headers) - 1;
|
|
|
|
slen = gst_buffer_get_size (streaminfo);
|
|
buf = gst_buffer_new_and_alloc (13 + slen);
|
|
|
|
gst_buffer_map (buf, &map, GST_MAP_WRITE);
|
|
bdata = map.data;
|
|
bdata[0] = 0x7f;
|
|
memcpy (bdata + 1, "FLAC", 4);
|
|
bdata[5] = 0x01; /* mapping version major */
|
|
bdata[6] = 0x00; /* mapping version minor */
|
|
bdata[7] = (num & 0xFF00) >> 8;
|
|
bdata[8] = (num & 0x00FF) >> 0;
|
|
memcpy (bdata + 9, "fLaC", 4);
|
|
gst_buffer_extract (streaminfo, 0, bdata + 13, slen);
|
|
gst_buffer_unmap (buf, &map);
|
|
|
|
notgst_value_array_append_buffer (&array, buf);
|
|
gst_buffer_unref (buf);
|
|
}
|
|
|
|
/* add VORBISCOMMENT header */
|
|
notgst_value_array_append_buffer (&array, vorbiscomment);
|
|
|
|
/* add other headers, if there are any */
|
|
for (l = enc->headers; l != NULL; l = l->next) {
|
|
GstBuffer *buf = GST_BUFFER_CAST (l->data);
|
|
|
|
if (buf != marker && buf != streaminfo && buf != vorbiscomment) {
|
|
notgst_value_array_append_buffer (&array, buf);
|
|
}
|
|
}
|
|
|
|
gst_structure_set_value (gst_caps_get_structure (caps, 0),
|
|
"streamheader", &array);
|
|
g_value_unset (&array);
|
|
|
|
push_headers:
|
|
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), caps);
|
|
|
|
gst_audio_encoder_set_headers (GST_AUDIO_ENCODER (enc), enc->headers);
|
|
enc->headers = NULL;
|
|
|
|
gst_caps_unref (caps);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static FLAC__StreamEncoderWriteStatus
|
|
gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
|
|
const FLAC__byte buffer[], size_t bytes,
|
|
unsigned samples, unsigned current_frame, void *client_data)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstFlacEnc *flacenc;
|
|
GstBuffer *outbuf;
|
|
GstSegment *segment;
|
|
GstClockTime duration;
|
|
|
|
flacenc = GST_FLAC_ENC (client_data);
|
|
|
|
if (flacenc->stopped)
|
|
return FLAC__STREAM_ENCODER_WRITE_STATUS_OK;
|
|
|
|
outbuf = gst_buffer_new_and_alloc (bytes);
|
|
gst_buffer_fill (outbuf, 0, buffer, bytes);
|
|
|
|
/* we assume libflac passes us stuff neatly framed */
|
|
if (!flacenc->got_headers) {
|
|
if (samples == 0) {
|
|
GST_DEBUG_OBJECT (flacenc, "Got header, queueing (%u bytes)",
|
|
(guint) bytes);
|
|
flacenc->headers = g_list_append (flacenc->headers, outbuf);
|
|
/* note: it's important that we increase our byte offset */
|
|
goto out;
|
|
} else {
|
|
GST_INFO_OBJECT (flacenc, "Non-header packet, we have all headers now");
|
|
ret = gst_flac_enc_process_stream_headers (flacenc);
|
|
flacenc->got_headers = TRUE;
|
|
}
|
|
}
|
|
|
|
if (flacenc->got_headers && samples == 0) {
|
|
/* header fixup, push downstream directly */
|
|
GST_DEBUG_OBJECT (flacenc, "Fixing up headers at pos=%" G_GUINT64_FORMAT
|
|
", size=%u", flacenc->offset, (guint) bytes);
|
|
#if 0
|
|
GST_MEMDUMP_OBJECT (flacenc, "Presumed header fragment",
|
|
GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf));
|
|
#endif
|
|
ret = gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (flacenc), outbuf);
|
|
} else {
|
|
/* regular frame data, pass to base class */
|
|
if (flacenc->eos && flacenc->samples_in == flacenc->samples_out + samples) {
|
|
/* If encoding part of a frame, and we have no set stop time on
|
|
* the output segment, we update the segment stop time to reflect
|
|
* the last sample. This will let oggmux set the last page's
|
|
* granpos to tell a decoder the dummy samples should be clipped.
|
|
*/
|
|
segment = &GST_AUDIO_ENCODER_OUTPUT_SEGMENT (flacenc);
|
|
if (!GST_CLOCK_TIME_IS_VALID (segment->stop)) {
|
|
GST_DEBUG_OBJECT (flacenc,
|
|
"No stop time and partial frame, updating segment");
|
|
duration =
|
|
gst_util_uint64_scale (flacenc->samples_out + samples,
|
|
GST_SECOND,
|
|
FLAC__stream_encoder_get_sample_rate (flacenc->encoder));
|
|
segment->stop = segment->start + duration;
|
|
GST_DEBUG_OBJECT (flacenc, "new output segment %" GST_SEGMENT_FORMAT,
|
|
segment);
|
|
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (flacenc),
|
|
gst_event_new_segment (segment));
|
|
}
|
|
}
|
|
|
|
GST_LOG ("Pushing buffer: samples=%u, size=%u, pos=%" G_GUINT64_FORMAT,
|
|
samples, (guint) bytes, flacenc->offset);
|
|
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (flacenc),
|
|
outbuf, samples);
|
|
}
|
|
|
|
if (ret != GST_FLOW_OK)
|
|
GST_DEBUG_OBJECT (flacenc, "flow: %s", gst_flow_get_name (ret));
|
|
|
|
flacenc->last_flow = ret;
|
|
|
|
out:
|
|
flacenc->offset += bytes;
|
|
|
|
if (ret != GST_FLOW_OK)
|
|
return FLAC__STREAM_ENCODER_WRITE_STATUS_FATAL_ERROR;
|
|
|
|
return FLAC__STREAM_ENCODER_WRITE_STATUS_OK;
|
|
}
|
|
|
|
static FLAC__StreamEncoderTellStatus
|
|
gst_flac_enc_tell_callback (const FLAC__StreamEncoder * encoder,
|
|
FLAC__uint64 * absolute_byte_offset, void *client_data)
|
|
{
|
|
GstFlacEnc *flacenc = GST_FLAC_ENC (client_data);
|
|
|
|
*absolute_byte_offset = flacenc->offset;
|
|
|
|
return FLAC__STREAM_ENCODER_TELL_STATUS_OK;
|
|
}
|
|
|
|
static gboolean
|
|
gst_flac_enc_sink_event (GstAudioEncoder * enc, GstEvent * event)
|
|
{
|
|
GstFlacEnc *flacenc;
|
|
GstTagList *taglist;
|
|
GstToc *toc;
|
|
gboolean ret = FALSE;
|
|
|
|
flacenc = GST_FLAC_ENC (enc);
|
|
|
|
GST_DEBUG ("Received %s event on sinkpad, %" GST_PTR_FORMAT,
|
|
GST_EVENT_TYPE_NAME (event), event);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
flacenc->eos = TRUE;
|
|
ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event);
|
|
break;
|
|
case GST_EVENT_TAG:
|
|
if (flacenc->tags) {
|
|
gst_event_parse_tag (event, &taglist);
|
|
gst_tag_list_insert (flacenc->tags, taglist,
|
|
gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (flacenc)));
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event);
|
|
break;
|
|
case GST_EVENT_TOC:
|
|
gst_event_parse_toc (event, &toc, NULL);
|
|
if (toc) {
|
|
if (flacenc->toc != toc) {
|
|
if (flacenc->toc)
|
|
gst_toc_unref (flacenc->toc);
|
|
flacenc->toc = toc;
|
|
}
|
|
}
|
|
ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event);
|
|
break;
|
|
case GST_EVENT_SEGMENT:
|
|
flacenc->samples_in = 0;
|
|
flacenc->samples_out = 0;
|
|
ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event);
|
|
break;
|
|
default:
|
|
ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_flac_enc_sink_query (GstAudioEncoder * enc, GstQuery * query)
|
|
{
|
|
GstPad *pad = GST_AUDIO_ENCODER_SINK_PAD (enc);
|
|
gboolean ret = FALSE;
|
|
|
|
GST_DEBUG ("Received %s query on sinkpad, %" GST_PTR_FORMAT,
|
|
GST_QUERY_TYPE_NAME (query), query);
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_ACCEPT_CAPS:{
|
|
GstCaps *acceptable, *caps;
|
|
|
|
acceptable = gst_pad_get_current_caps (pad);
|
|
if (acceptable == NULL) {
|
|
acceptable = gst_pad_get_pad_template_caps (pad);
|
|
}
|
|
|
|
gst_query_parse_accept_caps (query, &caps);
|
|
|
|
gst_query_set_accept_caps_result (query,
|
|
gst_caps_is_subset (caps, acceptable));
|
|
gst_caps_unref (acceptable);
|
|
ret = TRUE;
|
|
}
|
|
break;
|
|
default:
|
|
ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_query (enc, query);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
|
|
#define READ_INT24 GST_READ_UINT24_LE
|
|
#else
|
|
#define READ_INT24 GST_READ_UINT24_BE
|
|
#endif
|
|
|
|
static GstFlowReturn
|
|
gst_flac_enc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
|
|
{
|
|
GstFlacEnc *flacenc;
|
|
FLAC__int32 *data;
|
|
gint samples, width, channels;
|
|
gulong i;
|
|
gint j;
|
|
FLAC__bool res;
|
|
GstMapInfo map;
|
|
GstAudioInfo *info =
|
|
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (enc));
|
|
gint *reorder_map;
|
|
|
|
flacenc = GST_FLAC_ENC (enc);
|
|
|
|
/* base class ensures configuration */
|
|
g_return_val_if_fail (GST_AUDIO_INFO_WIDTH (info) != 0,
|
|
GST_FLOW_NOT_NEGOTIATED);
|
|
|
|
width = GST_AUDIO_INFO_WIDTH (info);
|
|
channels = GST_AUDIO_INFO_CHANNELS (info);
|
|
reorder_map = flacenc->channel_reorder_map;
|
|
|
|
if (G_UNLIKELY (!buffer)) {
|
|
if (flacenc->eos) {
|
|
GST_DEBUG_OBJECT (flacenc, "finish encoding");
|
|
FLAC__stream_encoder_finish (flacenc->encoder);
|
|
} else {
|
|
/* can't handle intermittent draining/resyncing */
|
|
GST_ELEMENT_WARNING (flacenc, STREAM, FORMAT, (NULL),
|
|
("Stream discontinuity detected. "
|
|
"The output may have wrong timestamps, "
|
|
"consider using audiorate to handle discontinuities"));
|
|
}
|
|
return flacenc->last_flow;
|
|
}
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
samples = map.size / (width >> 3);
|
|
|
|
data = g_malloc (samples * sizeof (FLAC__int32));
|
|
|
|
samples /= channels;
|
|
GST_LOG_OBJECT (flacenc, "processing %d samples, %d channels", samples,
|
|
channels);
|
|
if (width == 8) {
|
|
gint8 *indata = (gint8 *) map.data;
|
|
|
|
for (i = 0; i < samples; i++)
|
|
for (j = 0; j < channels; j++)
|
|
data[i * channels + reorder_map[j]] =
|
|
(FLAC__int32) indata[i * channels + j];
|
|
} else if (width == 16) {
|
|
gint16 *indata = (gint16 *) map.data;
|
|
|
|
for (i = 0; i < samples; i++)
|
|
for (j = 0; j < channels; j++)
|
|
data[i * channels + reorder_map[j]] =
|
|
(FLAC__int32) indata[i * channels + j];
|
|
} else if (width == 24) {
|
|
guint8 *indata = (guint8 *) map.data;
|
|
guint32 val;
|
|
|
|
for (i = 0; i < samples; i++)
|
|
for (j = 0; j < channels; j++) {
|
|
val = READ_INT24 (&indata[3 * (i * channels + j)]);
|
|
if (val & 0x00800000)
|
|
val |= 0xff000000;
|
|
data[i * channels + reorder_map[j]] = (FLAC__int32) val;
|
|
}
|
|
} else if (width == 32) {
|
|
gint32 *indata = (gint32 *) map.data;
|
|
|
|
for (i = 0; i < samples; i++)
|
|
for (j = 0; j < channels; j++)
|
|
data[i * channels + reorder_map[j]] =
|
|
(FLAC__int32) indata[i * channels + j];
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
res = FLAC__stream_encoder_process_interleaved (flacenc->encoder,
|
|
(const FLAC__int32 *) data, samples);
|
|
flacenc->samples_in += samples;
|
|
|
|
g_free (data);
|
|
|
|
if (!res) {
|
|
if (flacenc->last_flow == GST_FLOW_OK)
|
|
return GST_FLOW_ERROR;
|
|
else
|
|
return flacenc->last_flow;
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static void
|
|
gst_flac_enc_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstFlacEnc *this = GST_FLAC_ENC (object);
|
|
GstAudioEncoder *enc = GST_AUDIO_ENCODER (object);
|
|
guint64 curr_latency = 0, old_latency = gst_flac_enc_get_latency (this);
|
|
|
|
GST_OBJECT_LOCK (this);
|
|
|
|
switch (prop_id) {
|
|
case PROP_QUALITY:
|
|
gst_flac_enc_update_quality (this, g_value_get_enum (value));
|
|
break;
|
|
case PROP_STREAMABLE_SUBSET:
|
|
FLAC__stream_encoder_set_streamable_subset (this->encoder,
|
|
g_value_get_boolean (value));
|
|
break;
|
|
case PROP_MID_SIDE_STEREO:
|
|
FLAC__stream_encoder_set_do_mid_side_stereo (this->encoder,
|
|
g_value_get_boolean (value));
|
|
break;
|
|
case PROP_LOOSE_MID_SIDE_STEREO:
|
|
FLAC__stream_encoder_set_loose_mid_side_stereo (this->encoder,
|
|
g_value_get_boolean (value));
|
|
break;
|
|
case PROP_BLOCKSIZE:
|
|
FLAC__stream_encoder_set_blocksize (this->encoder,
|
|
g_value_get_uint (value));
|
|
break;
|
|
case PROP_MAX_LPC_ORDER:
|
|
FLAC__stream_encoder_set_max_lpc_order (this->encoder,
|
|
g_value_get_uint (value));
|
|
break;
|
|
case PROP_QLP_COEFF_PRECISION:
|
|
FLAC__stream_encoder_set_qlp_coeff_precision (this->encoder,
|
|
g_value_get_uint (value));
|
|
break;
|
|
case PROP_QLP_COEFF_PREC_SEARCH:
|
|
FLAC__stream_encoder_set_do_qlp_coeff_prec_search (this->encoder,
|
|
g_value_get_boolean (value));
|
|
break;
|
|
case PROP_ESCAPE_CODING:
|
|
FLAC__stream_encoder_set_do_escape_coding (this->encoder,
|
|
g_value_get_boolean (value));
|
|
break;
|
|
case PROP_EXHAUSTIVE_MODEL_SEARCH:
|
|
FLAC__stream_encoder_set_do_exhaustive_model_search (this->encoder,
|
|
g_value_get_boolean (value));
|
|
break;
|
|
case PROP_MIN_RESIDUAL_PARTITION_ORDER:
|
|
FLAC__stream_encoder_set_min_residual_partition_order (this->encoder,
|
|
g_value_get_uint (value));
|
|
break;
|
|
case PROP_MAX_RESIDUAL_PARTITION_ORDER:
|
|
FLAC__stream_encoder_set_max_residual_partition_order (this->encoder,
|
|
g_value_get_uint (value));
|
|
break;
|
|
case PROP_RICE_PARAMETER_SEARCH_DIST:
|
|
FLAC__stream_encoder_set_rice_parameter_search_dist (this->encoder,
|
|
g_value_get_uint (value));
|
|
break;
|
|
case PROP_PADDING:
|
|
this->padding = g_value_get_uint (value);
|
|
break;
|
|
case PROP_SEEKPOINTS:
|
|
this->seekpoints = g_value_get_int (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
|
|
GST_OBJECT_UNLOCK (this);
|
|
|
|
/* Update latency if it has changed */
|
|
curr_latency = gst_flac_enc_get_latency (this);
|
|
if (old_latency != curr_latency)
|
|
gst_audio_encoder_set_latency (enc, curr_latency, curr_latency);
|
|
}
|
|
|
|
static void
|
|
gst_flac_enc_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstFlacEnc *this = GST_FLAC_ENC (object);
|
|
|
|
GST_OBJECT_LOCK (this);
|
|
|
|
switch (prop_id) {
|
|
case PROP_QUALITY:
|
|
g_value_set_enum (value, this->quality);
|
|
break;
|
|
case PROP_STREAMABLE_SUBSET:
|
|
g_value_set_boolean (value,
|
|
FLAC__stream_encoder_get_streamable_subset (this->encoder));
|
|
break;
|
|
case PROP_MID_SIDE_STEREO:
|
|
g_value_set_boolean (value,
|
|
FLAC__stream_encoder_get_do_mid_side_stereo (this->encoder));
|
|
break;
|
|
case PROP_LOOSE_MID_SIDE_STEREO:
|
|
g_value_set_boolean (value,
|
|
FLAC__stream_encoder_get_loose_mid_side_stereo (this->encoder));
|
|
break;
|
|
case PROP_BLOCKSIZE:
|
|
g_value_set_uint (value,
|
|
FLAC__stream_encoder_get_blocksize (this->encoder));
|
|
break;
|
|
case PROP_MAX_LPC_ORDER:
|
|
g_value_set_uint (value,
|
|
FLAC__stream_encoder_get_max_lpc_order (this->encoder));
|
|
break;
|
|
case PROP_QLP_COEFF_PRECISION:
|
|
g_value_set_uint (value,
|
|
FLAC__stream_encoder_get_qlp_coeff_precision (this->encoder));
|
|
break;
|
|
case PROP_QLP_COEFF_PREC_SEARCH:
|
|
g_value_set_boolean (value,
|
|
FLAC__stream_encoder_get_do_qlp_coeff_prec_search (this->encoder));
|
|
break;
|
|
case PROP_ESCAPE_CODING:
|
|
g_value_set_boolean (value,
|
|
FLAC__stream_encoder_get_do_escape_coding (this->encoder));
|
|
break;
|
|
case PROP_EXHAUSTIVE_MODEL_SEARCH:
|
|
g_value_set_boolean (value,
|
|
FLAC__stream_encoder_get_do_exhaustive_model_search (this->encoder));
|
|
break;
|
|
case PROP_MIN_RESIDUAL_PARTITION_ORDER:
|
|
g_value_set_uint (value,
|
|
FLAC__stream_encoder_get_min_residual_partition_order
|
|
(this->encoder));
|
|
break;
|
|
case PROP_MAX_RESIDUAL_PARTITION_ORDER:
|
|
g_value_set_uint (value,
|
|
FLAC__stream_encoder_get_max_residual_partition_order
|
|
(this->encoder));
|
|
break;
|
|
case PROP_RICE_PARAMETER_SEARCH_DIST:
|
|
g_value_set_uint (value,
|
|
FLAC__stream_encoder_get_rice_parameter_search_dist (this->encoder));
|
|
break;
|
|
case PROP_PADDING:
|
|
g_value_set_uint (value, this->padding);
|
|
break;
|
|
case PROP_SEEKPOINTS:
|
|
g_value_set_int (value, this->seekpoints);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
|
|
GST_OBJECT_UNLOCK (this);
|
|
}
|