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gstreamer/ext/flac/gstflacenc.c
Julian Bouzas 2ebdd70c21 flac: report latency in flacenc and flacdec 
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
2019-03-25 15:14:32 +01:00

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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}
};
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-flac")
);
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 gboolean gst_flac_enc_sink_query (GstAudioEncoder * enc,
GstQuery * query);
static void gst_flac_enc_finalize (GObject * object);
static GstCaps *gst_flac_enc_generate_sink_caps (void);
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;
GstCaps *sink_caps;
GstPadTemplate *sink_templ;
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_static_pad_template (gstelement_class, &src_factory);
sink_caps = gst_flac_enc_generate_sink_caps ();
sink_templ = gst_pad_template_new ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, sink_caps);
gst_element_class_add_pad_template (gstelement_class, sink_templ);
gst_caps_unref (sink_caps);
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);
base_class->sink_query = GST_DEBUG_FUNCPTR (gst_flac_enc_sink_query);
}
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 != 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)
{
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 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));
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);
}
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