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gstreamer/ext/flac/gstflacenc.c
Vincent Penquerc'h b18d8b085a flacenc: update output segment stop time to match clipped samples 
This will let oggmux generate a granpos on the last page that properly
represents the clipped samples at the end of the stream.
2014-10-30 14:30:30 +00:00

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-flacenc
* @see_also: #GstFlacDec
*
* flacenc encodes FLAC streams.
* <ulink url="http://flac.sourceforge.net/">FLAC</ulink>
* is a Free Lossless Audio Codec. FLAC audio can directly be written into
* a file, or embedded into containers such as oggmux or matroskamux.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 audiotestsrc num-buffers=100 ! flacenc ! filesink location=beep.flac
* ]| Encode a short sine wave into FLAC
* |[
* gst-launch-1.0 cdparanoiasrc mode=continuous ! queue ! audioconvert ! flacenc ! filesink location=cd.flac
* ]| Rip a whole audio CD into a single FLAC file, with the track table saved as a CUE sheet inside the FLAC file
* |[
* gst-launch-1.0 cdparanoiasrc track=5 ! queue ! audioconvert ! flacenc ! filesink location=track5.flac
* ]| Rip track 5 of an audio CD and encode it losslessly to a FLAC file
* </refsect2>
*/
/* TODO: - We currently don't handle discontinuities in the stream in a useful
* way and instead rely on the developer plugging in audiorate if
* the stream contains discontinuities.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gstflacenc.h>
#include <gst/audio/audio.h>
#include <gst/tag/tag.h>
#include <gst/gsttagsetter.h>
/* Taken from http://flac.sourceforge.net/format.html#frame_header */
static const GstAudioChannelPosition channel_positions[8][8] = {
{GST_AUDIO_CHANNEL_POSITION_MONO},
{GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
/* FIXME: 7/8 channel layouts are not defined in the FLAC specs */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}
};
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define FORMATS "{ S24LE, S24_32LE, S16LE, S8 } "
#else
#define FORMATS "{ S24BE, S24_32BE, S16BE, S8 } "
#endif
#define FLAC_SINK_CAPS \
"audio/x-raw, " \
"format = (string) " FORMATS ", " \
"layout = (string) interleaved, " \
"rate = (int) [ 1, 655350 ], " \
"channels = (int) [ 1, 8 ]"
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-flac")
);
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (FLAC_SINK_CAPS)
);
enum
{
PROP_0,
PROP_QUALITY,
PROP_STREAMABLE_SUBSET,
PROP_MID_SIDE_STEREO,
PROP_LOOSE_MID_SIDE_STEREO,
PROP_BLOCKSIZE,
PROP_MAX_LPC_ORDER,
PROP_QLP_COEFF_PRECISION,
PROP_QLP_COEFF_PREC_SEARCH,
PROP_ESCAPE_CODING,
PROP_EXHAUSTIVE_MODEL_SEARCH,
PROP_MIN_RESIDUAL_PARTITION_ORDER,
PROP_MAX_RESIDUAL_PARTITION_ORDER,
PROP_RICE_PARAMETER_SEARCH_DIST,
PROP_PADDING,
PROP_SEEKPOINTS
};
GST_DEBUG_CATEGORY_STATIC (flacenc_debug);
#define GST_CAT_DEFAULT flacenc_debug
#define gst_flac_enc_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstFlacEnc, gst_flac_enc, GST_TYPE_AUDIO_ENCODER,
G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL)
G_IMPLEMENT_INTERFACE (GST_TYPE_TOC_SETTER, NULL)
);
static gboolean gst_flac_enc_start (GstAudioEncoder * enc);
static gboolean gst_flac_enc_stop (GstAudioEncoder * enc);
static gboolean gst_flac_enc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_flac_enc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
static GstCaps *gst_flac_enc_getcaps (GstAudioEncoder * enc, GstCaps * filter);
static gboolean gst_flac_enc_sink_event (GstAudioEncoder * enc,
GstEvent * event);
static void gst_flac_enc_finalize (GObject * object);
static gboolean gst_flac_enc_update_quality (GstFlacEnc * flacenc,
gint quality);
static void gst_flac_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_flac_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
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);
static FLAC__StreamEncoderSeekStatus
gst_flac_enc_seek_callback (const FLAC__StreamEncoder * encoder,
FLAC__uint64 absolute_byte_offset, void *client_data);
static FLAC__StreamEncoderTellStatus
gst_flac_enc_tell_callback (const FLAC__StreamEncoder * encoder,
FLAC__uint64 * absolute_byte_offset, void *client_data);
typedef struct
{
gboolean exhaustive_model_search;
gboolean escape_coding;
gboolean mid_side;
gboolean loose_mid_side;
guint qlp_coeff_precision;
gboolean qlp_coeff_prec_search;
guint min_residual_partition_order;
guint max_residual_partition_order;
guint rice_parameter_search_dist;
guint max_lpc_order;
guint blocksize;
}
GstFlacEncParams;
static const GstFlacEncParams flacenc_params[] = {
{FALSE, FALSE, FALSE, FALSE, 0, FALSE, 2, 2, 0, 0, 1152},
{FALSE, FALSE, TRUE, TRUE, 0, FALSE, 2, 2, 0, 0, 1152},
{FALSE, FALSE, TRUE, FALSE, 0, FALSE, 0, 3, 0, 0, 1152},
{FALSE, FALSE, FALSE, FALSE, 0, FALSE, 3, 3, 0, 6, 4608},
{FALSE, FALSE, TRUE, TRUE, 0, FALSE, 3, 3, 0, 8, 4608},
{FALSE, FALSE, TRUE, FALSE, 0, FALSE, 3, 3, 0, 8, 4608},
{FALSE, FALSE, TRUE, FALSE, 0, FALSE, 0, 4, 0, 8, 4608},
{TRUE, FALSE, TRUE, FALSE, 0, FALSE, 0, 6, 0, 8, 4608},
{TRUE, FALSE, TRUE, FALSE, 0, FALSE, 0, 6, 0, 12, 4608},
{TRUE, TRUE, TRUE, FALSE, 0, FALSE, 0, 16, 0, 32, 4608},
};
#define DEFAULT_QUALITY 5
#define DEFAULT_PADDING 0
#define DEFAULT_SEEKPOINTS -10
#define GST_TYPE_FLAC_ENC_QUALITY (gst_flac_enc_quality_get_type ())
static GType
gst_flac_enc_quality_get_type (void)
{
static GType qtype = 0;
if (qtype == 0) {
static const GEnumValue values[] = {
{0, "0 - Fastest compression", "0"},
{1, "1", "1"},
{2, "2", "2"},
{3, "3", "3"},
{4, "4", "4"},
{5, "5 - Default", "5"},
{6, "6", "6"},
{7, "7", "7"},
{8, "8 - Highest compression", "8"},
{9, "9 - Insane", "9"},
{0, NULL, NULL}
};
qtype = g_enum_register_static ("GstFlacEncQuality", values);
}
return qtype;
}
static void
gst_flac_enc_class_init (GstFlacEncClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstAudioEncoderClass *base_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
base_class = (GstAudioEncoderClass *) (klass);
GST_DEBUG_CATEGORY_INIT (flacenc_debug, "flacenc", 0,
"Flac encoding element");
gobject_class->set_property = gst_flac_enc_set_property;
gobject_class->get_property = gst_flac_enc_get_property;
gobject_class->finalize = gst_flac_enc_finalize;
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QUALITY,
g_param_spec_enum ("quality",
"Quality",
"Speed versus compression tradeoff",
GST_TYPE_FLAC_ENC_QUALITY, DEFAULT_QUALITY,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_STREAMABLE_SUBSET, g_param_spec_boolean ("streamable-subset",
"Streamable subset",
"true to limit encoder to generating a Subset stream, else false",
TRUE,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MID_SIDE_STEREO,
g_param_spec_boolean ("mid-side-stereo", "Do mid side stereo",
"Do mid side stereo (only for stereo input)",
flacenc_params[DEFAULT_QUALITY].mid_side,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_LOOSE_MID_SIDE_STEREO, g_param_spec_boolean ("loose-mid-side-stereo",
"Loose mid side stereo", "Loose mid side stereo",
flacenc_params[DEFAULT_QUALITY].loose_mid_side,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BLOCKSIZE,
g_param_spec_uint ("blocksize", "Blocksize", "Blocksize in samples",
FLAC__MIN_BLOCK_SIZE, FLAC__MAX_BLOCK_SIZE,
flacenc_params[DEFAULT_QUALITY].blocksize,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_LPC_ORDER,
g_param_spec_uint ("max-lpc-order", "Max LPC order",
"Max LPC order; 0 => use only fixed predictors", 0,
FLAC__MAX_LPC_ORDER, flacenc_params[DEFAULT_QUALITY].max_lpc_order,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_QLP_COEFF_PRECISION, g_param_spec_uint ("qlp-coeff-precision",
"QLP coefficients precision",
"Precision in bits of quantized linear-predictor coefficients; 0 = automatic",
0, 32, flacenc_params[DEFAULT_QUALITY].qlp_coeff_precision,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_QLP_COEFF_PREC_SEARCH, g_param_spec_boolean ("qlp-coeff-prec-search",
"Do QLP coefficients precision search",
"false = use qlp_coeff_precision, "
"true = search around qlp_coeff_precision, take best",
flacenc_params[DEFAULT_QUALITY].qlp_coeff_prec_search,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_ESCAPE_CODING,
g_param_spec_boolean ("escape-coding", "Do Escape coding",
"search for escape codes in the entropy coding stage "
"for slightly better compression",
flacenc_params[DEFAULT_QUALITY].escape_coding,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_EXHAUSTIVE_MODEL_SEARCH,
g_param_spec_boolean ("exhaustive-model-search",
"Do exhaustive model search",
"do exhaustive search of LP coefficient quantization (expensive!)",
flacenc_params[DEFAULT_QUALITY].exhaustive_model_search,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_MIN_RESIDUAL_PARTITION_ORDER,
g_param_spec_uint ("min-residual-partition-order",
"Min residual partition order",
"Min residual partition order (above 4 doesn't usually help much)", 0,
16, flacenc_params[DEFAULT_QUALITY].min_residual_partition_order,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_MAX_RESIDUAL_PARTITION_ORDER,
g_param_spec_uint ("max-residual-partition-order",
"Max residual partition order",
"Max residual partition order (above 4 doesn't usually help much)", 0,
16, flacenc_params[DEFAULT_QUALITY].max_residual_partition_order,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_RICE_PARAMETER_SEARCH_DIST,
g_param_spec_uint ("rice-parameter-search-dist",
"rice_parameter_search_dist",
"0 = try only calc'd parameter k; else try all [k-dist..k+dist] "
"parameters, use best", 0, FLAC__MAX_RICE_PARTITION_ORDER,
flacenc_params[DEFAULT_QUALITY].rice_parameter_search_dist,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_PADDING,
g_param_spec_uint ("padding",
"Padding",
"Write a PADDING block with this length in bytes", 0, G_MAXUINT,
DEFAULT_PADDING,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_SEEKPOINTS,
g_param_spec_int ("seekpoints",
"Seekpoints",
"Add SEEKTABLE metadata (if > 0, number of entries, if < 0, interval in sec)",
-G_MAXINT, G_MAXINT,
DEFAULT_SEEKPOINTS,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_set_static_metadata (gstelement_class, "FLAC audio encoder",
"Codec/Encoder/Audio",
"Encodes audio with the FLAC lossless audio encoder",
"Wim Taymans <wim.taymans@chello.be>");
base_class->start = GST_DEBUG_FUNCPTR (gst_flac_enc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_flac_enc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_flac_enc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_flac_enc_handle_frame);
base_class->getcaps = GST_DEBUG_FUNCPTR (gst_flac_enc_getcaps);
base_class->sink_event = GST_DEBUG_FUNCPTR (gst_flac_enc_sink_event);
}
static void
gst_flac_enc_init (GstFlacEnc * flacenc)
{
GstAudioEncoder *enc = GST_AUDIO_ENCODER (flacenc);
flacenc->encoder = FLAC__stream_encoder_new ();
gst_flac_enc_update_quality (flacenc, DEFAULT_QUALITY);
/* arrange granulepos marking (and required perfect ts) */
gst_audio_encoder_set_mark_granule (enc, TRUE);
gst_audio_encoder_set_perfect_timestamp (enc, TRUE);
}
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)
strcpy (track->isrc, isrc);
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;
GstMapInfo map;
for (i = 0; i < n_images + n_preview_images; i++) {
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);
if (structure)
gst_structure_get (structure, "image-type", GST_TYPE_TAG_IMAGE_TYPE,
&image_type, NULL);
else
image_type = GST_TAG_IMAGE_TYPE_NONE;
/* Convert to ID3v2 APIC image type */
if (image_type == GST_TAG_IMAGE_TYPE_NONE)
image_type = (i < n_images) ? 0x00 : 0x01;
else
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);
/* FIXME: There's no way to set the picture type in libFLAC */
flacenc->meta[entries]->data.picture.type = image_type;
structure = gst_caps_get_structure (caps, 0);
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_getcaps (GstAudioEncoder * enc, GstCaps * filter)
{
GstCaps *ret = NULL, *caps = NULL;
GstPad *pad;
pad = GST_AUDIO_ENCODER_SINK_PAD (enc);
if (gst_pad_has_current_caps (pad)) {
ret = gst_pad_get_current_caps (pad);
} else {
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);
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 ();
for (i = 1; i <= 8; i++) {
s2 = gst_structure_copy (s);
if (i == 1) {
gst_structure_set (s2, "channels", G_TYPE_INT, 1, NULL);
} else {
guint64 channel_mask;
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);
}
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)
{
gint64 duration;
GstAudioInfo *info =
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (flacenc));
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 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));
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;
/* no special feedback to base class; should provide all available samples */
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;
}
#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);
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);
}
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);
}
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