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flacenc: port to audioencoder

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This commit is contained in:
Mark Nauwelaerts 2011-09-28 16:09:58 +02:00
parent 671b56f9da
commit e8bcd41d73
3 changed files with 142 additions and 240 deletions
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3
ext/flac/Makefile.am
View file

@ -1,7 +1,8 @@
plugin_LTLIBRARIES = libgstflac.la
libgstflac_la_SOURCES = gstflac.c gstflacdec.c gstflacenc.c gstflactag.c
libgstflac_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(FLAC_CFLAGS)
libgstflac_la_CFLAGS = -DGST_USE_UNSTABLE_API \
$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(FLAC_CFLAGS)
libgstflac_la_LIBADD = \
$(GST_PLUGINS_BASE_LIBS) -lgsttag-$(GST_MAJORMINOR) \
-lgstaudio-$(GST_MAJORMINOR) \

364
ext/flac/gstflacenc.c
View file

@ -157,35 +157,31 @@ GST_DEBUG_CATEGORY_STATIC (flacenc_debug);
NULL, \
NULL \
}; \
static const GInterfaceInfo preset_info = { \
NULL, \
NULL, \
NULL \
}; \
g_type_add_interface_static (type, GST_TYPE_TAG_SETTER, \
&tag_setter_info); \
g_type_add_interface_static (type, GST_TYPE_PRESET, \
&preset_info); \
}G_STMT_END
GST_BOILERPLATE_FULL (GstFlacEnc, gst_flac_enc, GstElement, GST_TYPE_ELEMENT,
_do_init);
GST_BOILERPLATE_FULL (GstFlacEnc, gst_flac_enc, GstAudioEncoder,
GST_TYPE_AUDIO_ENCODER, _do_init);
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);
static gboolean gst_flac_enc_sink_event (GstAudioEncoder * enc,
GstEvent * event);
static void gst_flac_enc_finalize (GObject * object);
static gboolean gst_flac_enc_sink_setcaps (GstPad * pad, GstCaps * caps);
static GstCaps *gst_flac_enc_sink_getcaps (GstPad * pad);
static gboolean gst_flac_enc_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_flac_enc_chain (GstPad * pad, GstBuffer * buffer);
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 GstStateChangeReturn gst_flac_enc_change_state (GstElement * element,
GstStateChange transition);
static FLAC__StreamEncoderWriteStatus
gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
@ -280,15 +276,22 @@ 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);
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;
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->event = GST_DEBUG_FUNCPTR (gst_flac_enc_sink_event);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QUALITY,
g_param_spec_enum ("quality",
"Quality",
@ -401,38 +404,19 @@ gst_flac_enc_class_init (GstFlacEncClass * klass)
-G_MAXINT, G_MAXINT,
DEFAULT_SEEKPOINTS,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state = gst_flac_enc_change_state;
}
static void
gst_flac_enc_init (GstFlacEnc * flacenc, GstFlacEncClass * klass)
{
flacenc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
gst_pad_set_chain_function (flacenc->sinkpad,
GST_DEBUG_FUNCPTR (gst_flac_enc_chain));
gst_pad_set_event_function (flacenc->sinkpad,
GST_DEBUG_FUNCPTR (gst_flac_enc_sink_event));
gst_pad_set_getcaps_function (flacenc->sinkpad,
GST_DEBUG_FUNCPTR (gst_flac_enc_sink_getcaps));
gst_pad_set_setcaps_function (flacenc->sinkpad,
GST_DEBUG_FUNCPTR (gst_flac_enc_sink_setcaps));
gst_element_add_pad (GST_ELEMENT (flacenc), flacenc->sinkpad);
flacenc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_pad_use_fixed_caps (flacenc->srcpad);
gst_element_add_pad (GST_ELEMENT (flacenc), flacenc->srcpad);
GstAudioEncoder *enc = GST_AUDIO_ENCODER (flacenc);
flacenc->encoder = FLAC__stream_encoder_new ();
flacenc->offset = 0;
flacenc->samples_written = 0;
flacenc->channels = 0;
gst_flac_enc_update_quality (flacenc, DEFAULT_QUALITY);
flacenc->tags = gst_tag_list_new ();
flacenc->got_headers = FALSE;
flacenc->headers = NULL;
flacenc->last_flow = GST_FLOW_OK;
/* 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
@ -440,12 +424,62 @@ gst_flac_enc_finalize (GObject * object)
{
GstFlacEnc *flacenc = GST_FLAC_ENC (object);
gst_tag_list_free (flacenc->tags);
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->channels = 0;
flacenc->depth = 0;
flacenc->sample_rate = 0;
flacenc->eos = FALSE;
flacenc->tags = gst_tag_list_new ();
return TRUE;
}
static gboolean
gst_flac_enc_stop (GstAudioEncoder * enc)
{
GstFlacEnc *flacenc = GST_FLAC_ENC (enc);
GST_DEBUG_OBJECT (enc, "stop");
gst_tag_list_free (flacenc->tags);
flacenc->tags = 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]);
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;
return TRUE;
}
static void
add_one_tag (const GstTagList * list, const gchar * tag, gpointer user_data)
{
@ -629,9 +663,12 @@ gst_flac_enc_caps_append_structure_with_widths (GstCaps * caps,
}
static GstCaps *
gst_flac_enc_sink_getcaps (GstPad * pad)
gst_flac_enc_getcaps (GstAudioEncoder * enc)
{
GstCaps *ret = NULL;
GstCaps *ret = NULL, *caps = NULL;
GstPad *pad;
pad = GST_AUDIO_ENCODER_SINK_PAD (enc);
GST_OBJECT_LOCK (pad);
@ -679,7 +716,10 @@ gst_flac_enc_sink_getcaps (GstPad * pad)
GST_DEBUG_OBJECT (pad, "Return caps %" GST_PTR_FORMAT, ret);
return ret;
caps = gst_audio_encoder_proxy_getcaps (enc, ret);
gst_caps_unref (ret);
return caps;
}
static guint64
@ -716,45 +756,36 @@ done:
}
static gboolean
gst_flac_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
gst_flac_enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstFlacEnc *flacenc;
GstStructure *structure;
guint64 total_samples = GST_CLOCK_TIME_NONE;
FLAC__StreamEncoderInitStatus init_status;
gint depth, chans, rate, width;
GstCaps *caps;
flacenc = GST_FLAC_ENC (gst_pad_get_parent (pad));
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;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "channels", &chans) ||
!gst_structure_get_int (structure, "width", &width) ||
!gst_structure_get_int (structure, "depth", &depth) ||
!gst_structure_get_int (structure, "rate", &rate)) {
GST_DEBUG_OBJECT (flacenc, "incomplete caps: %" GST_PTR_FORMAT, caps);
return FALSE;
}
flacenc->channels = chans;
flacenc->width = width;
flacenc->depth = depth;
flacenc->sample_rate = rate;
flacenc->channels = GST_AUDIO_INFO_CHANNELS (info);
flacenc->width = GST_AUDIO_INFO_WIDTH (info);
flacenc->depth = GST_AUDIO_INFO_DEPTH (info);
flacenc->sample_rate = GST_AUDIO_INFO_RATE (info);
caps = gst_caps_new_simple ("audio/x-flac",
"channels", G_TYPE_INT, flacenc->channels,
"rate", G_TYPE_INT, flacenc->sample_rate, NULL);
if (!gst_pad_set_caps (flacenc->srcpad, caps))
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps))
goto setting_src_caps_failed;
gst_caps_unref (caps);
total_samples = gst_flac_enc_query_peer_total_samples (flacenc, pad);
total_samples = gst_flac_enc_query_peer_total_samples (flacenc,
GST_AUDIO_ENCODER_SINK_PAD (enc));
FLAC__stream_encoder_set_bits_per_sample (flacenc->encoder, flacenc->depth);
FLAC__stream_encoder_set_sample_rate (flacenc->encoder, flacenc->sample_rate);
@ -766,13 +797,17 @@ gst_flac_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
gst_flac_enc_set_metadata (flacenc, 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;
gst_object_unref (flacenc);
/* no special feedback to base class; should provide all available samples */
return TRUE;
@ -859,7 +894,7 @@ gst_flac_enc_seek_callback (const FLAC__StreamEncoder * encoder,
event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES,
absolute_byte_offset, GST_BUFFER_OFFSET_NONE, 0);
if ((peerpad = gst_pad_get_peer (flacenc->srcpad))) {
if ((peerpad = gst_pad_get_peer (GST_AUDIO_ENCODER_SRC_PAD (flacenc)))) {
gboolean ret = gst_pad_send_event (peerpad, event);
gst_object_unref (peerpad);
@ -900,7 +935,7 @@ notgst_value_array_append_buffer (GValue * array_val, GstBuffer * buf)
#define HDR_TYPE_STREAMINFO 0
#define HDR_TYPE_VORBISCOMMENT 4
static void
static GstFlowReturn
gst_flac_enc_process_stream_headers (GstFlacEnc * enc)
{
GstBuffer *vorbiscomment = NULL;
@ -909,6 +944,7 @@ gst_flac_enc_process_stream_headers (GstFlacEnc * enc)
GValue array = { 0, };
GstCaps *caps;
GList *l;
GstFlowReturn ret = GST_FLOW_OK;
caps = gst_caps_new_simple ("audio/x-flac",
"channels", G_TYPE_INT, enc->channels,
@ -984,8 +1020,6 @@ gst_flac_enc_process_stream_headers (GstFlacEnc * enc)
push_headers:
gst_pad_set_caps (enc->srcpad, caps);
/* push header buffers; update caps, so when we push the first buffer the
* negotiated caps will change to caps that include the streamheader field */
for (l = enc->headers; l != NULL; l = l->next) {
@ -997,13 +1031,15 @@ push_headers:
GST_BUFFER_SIZE (buf));
GST_MEMDUMP_OBJECT (enc, "header buffer", GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf));
(void) gst_pad_push (enc->srcpad, buf);
ret = gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), buf);
l->data = NULL;
}
g_list_free (enc->headers);
enc->headers = NULL;
gst_caps_unref (caps);
return ret;
}
static FLAC__StreamEncoderWriteStatus
@ -1023,31 +1059,6 @@ gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
outbuf = gst_buffer_new_and_alloc (bytes);
memcpy (GST_BUFFER_DATA (outbuf), buffer, bytes);
if (samples > 0 && flacenc->samples_written != (guint64) - 1) {
guint64 granulepos;
GST_BUFFER_TIMESTAMP (outbuf) = flacenc->start_ts +
GST_FRAMES_TO_CLOCK_TIME (flacenc->samples_written,
flacenc->sample_rate);
GST_BUFFER_DURATION (outbuf) =
GST_FRAMES_TO_CLOCK_TIME (samples, flacenc->sample_rate);
/* offset_end = granulepos for ogg muxer */
granulepos =
flacenc->granulepos_offset + flacenc->samples_written + samples;
GST_BUFFER_OFFSET_END (outbuf) = granulepos;
/* offset = timestamp corresponding to granulepos for ogg muxer
* (see vorbisenc for a much more elaborate version of this) */
GST_BUFFER_OFFSET (outbuf) =
GST_FRAMES_TO_CLOCK_TIME (granulepos, flacenc->sample_rate);
} else {
GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET (outbuf) =
flacenc->samples_written * flacenc->width * flacenc->channels;
GST_BUFFER_OFFSET_END (outbuf) = 0;
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_IN_CAPS);
}
/* we assume libflac passes us stuff neatly framed */
if (!flacenc->got_headers) {
if (samples == 0) {
@ -1058,32 +1069,34 @@ gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
goto out;
} else {
GST_INFO_OBJECT (flacenc, "Non-header packet, we have all headers now");
gst_flac_enc_process_stream_headers (flacenc);
ret = gst_flac_enc_process_stream_headers (flacenc);
flacenc->got_headers = TRUE;
}
} else 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);
GST_MEMDUMP_OBJECT (flacenc, "Presumed header fragment",
GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf));
gst_buffer_set_caps (outbuf,
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (flacenc)));
ret = gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (flacenc), outbuf);
} else {
/* regular frame data, pass to base class */
GST_LOG ("Pushing buffer: ts=%" GST_TIME_FORMAT ", samples=%u, size=%u, "
"pos=%" G_GUINT64_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
samples, (guint) bytes, flacenc->offset);
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (flacenc),
outbuf, samples);
}
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (flacenc->srcpad));
ret = gst_pad_push (flacenc->srcpad, outbuf);
if (ret != GST_FLOW_OK)
GST_DEBUG_OBJECT (flacenc, "flow: %s", gst_flow_get_name (ret));
flacenc->last_flow = ret;
out:
flacenc->offset += bytes;
flacenc->samples_written += samples;
if (ret != GST_FLOW_OK)
return FLAC__STREAM_ENCODER_WRITE_STATUS_FATAL_ERROR;
@ -1103,13 +1116,13 @@ gst_flac_enc_tell_callback (const FLAC__StreamEncoder * encoder,
}
static gboolean
gst_flac_enc_sink_event (GstPad * pad, GstEvent * event)
gst_flac_enc_sink_event (GstAudioEncoder * enc, GstEvent * event)
{
GstFlacEnc *flacenc;
GstTagList *taglist;
gboolean ret = TRUE;
gboolean ret = FALSE;
flacenc = GST_FLAC_ENC (gst_pad_get_parent (pad));
flacenc = GST_FLAC_ENC (enc);
GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event));
@ -1135,20 +1148,20 @@ gst_flac_enc_sink_event (GstPad * pad, GstEvent * event)
GstEvent *e = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_BYTES,
0, -1, 0);
ret = gst_pad_push_event (flacenc->srcpad, e);
ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), e);
}
if (stream_time > 0) {
GST_DEBUG ("Not handling non-zero stream time");
}
gst_event_unref (event);
/* don't push it downstream, we'll generate our own via seek to 0 */
gst_event_unref (event);
ret = TRUE;
break;
}
case GST_EVENT_EOS:
FLAC__stream_encoder_finish (flacenc->encoder);
ret = gst_pad_event_default (pad, event);
flacenc->eos = TRUE;
break;
case GST_EVENT_TAG:
if (flacenc->tags) {
@ -1158,42 +1171,16 @@ gst_flac_enc_sink_event (GstPad * pad, GstEvent * event)
} else {
g_assert_not_reached ();
}
ret = gst_pad_event_default (pad, event);
break;
default:
ret = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (flacenc);
return ret;
}
static gboolean
gst_flac_enc_check_discont (GstFlacEnc * flacenc, GstClockTime expected,
GstClockTime timestamp)
{
guint allowed_diff = GST_SECOND / flacenc->sample_rate / 2;
if ((timestamp + allowed_diff < expected)
|| (timestamp > expected + allowed_diff)) {
GST_ELEMENT_WARNING (flacenc, STREAM, FORMAT, (NULL),
("Stream discontinuity detected (wanted %" GST_TIME_FORMAT " got %"
GST_TIME_FORMAT "). The output will have wrong timestamps,"
" consider using audiorate to handle discontinuities",
GST_TIME_ARGS (expected), GST_TIME_ARGS (timestamp)));
return TRUE;
}
/* TODO: Do something to handle discontinuities in the stream. The FLAC encoder
* unfortunately doesn't have any way to flush it's internal buffers */
return FALSE;
}
static GstFlowReturn
gst_flac_enc_chain (GstPad * pad, GstBuffer * buffer)
gst_flac_enc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
{
GstFlacEnc *flacenc;
FLAC__int32 *data;
@ -1202,42 +1189,26 @@ gst_flac_enc_chain (GstPad * pad, GstBuffer * buffer)
gulong i;
FLAC__bool res;
flacenc = GST_FLAC_ENC (GST_PAD_PARENT (pad));
flacenc = GST_FLAC_ENC (enc);
/* make sure setcaps has been called and the encoder is set up */
if (G_UNLIKELY (flacenc->depth == 0))
return GST_FLOW_NOT_NEGOTIATED;
/* base class ensures configuration */
g_return_val_if_fail (flacenc->depth != 0, GST_FLOW_NOT_NEGOTIATED);
width = flacenc->width;
/* Save the timestamp of the first buffer. This will be later
* used as offset for all following buffers */
if (flacenc->start_ts == GST_CLOCK_TIME_NONE) {
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
flacenc->start_ts = GST_BUFFER_TIMESTAMP (buffer);
flacenc->granulepos_offset = gst_util_uint64_scale
(GST_BUFFER_TIMESTAMP (buffer), flacenc->sample_rate, GST_SECOND);
if (G_UNLIKELY (!buffer)) {
if (flacenc->eos) {
FLAC__stream_encoder_finish (flacenc->encoder);
} else {
flacenc->start_ts = 0;
flacenc->granulepos_offset = 0;
/* 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;
}
/* Check if we have a continous stream, if not drop some samples or the buffer or
* insert some silence samples */
if (flacenc->next_ts != GST_CLOCK_TIME_NONE
&& GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
gst_flac_enc_check_discont (flacenc, flacenc->next_ts,
GST_BUFFER_TIMESTAMP (buffer));
}
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)
&& GST_BUFFER_DURATION_IS_VALID (buffer))
flacenc->next_ts =
GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
else
flacenc->next_ts = GST_CLOCK_TIME_NONE;
insize = GST_BUFFER_SIZE (buffer);
samples = insize / (width >> 3);
@ -1262,8 +1233,6 @@ gst_flac_enc_chain (GstPad * pad, GstBuffer * buffer)
g_assert_not_reached ();
}
gst_buffer_unref (buffer);
res = FLAC__stream_encoder_process_interleaved (flacenc->encoder,
(const FLAC__int32 *) data, samples / flacenc->channels);
@ -1428,64 +1397,3 @@ gst_flac_enc_get_property (GObject * object, guint prop_id,
GST_OBJECT_UNLOCK (this);
}
static GstStateChangeReturn
gst_flac_enc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstFlacEnc *flacenc = GST_FLAC_ENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
case GST_STATE_CHANGE_READY_TO_PAUSED:
flacenc->stopped = FALSE;
flacenc->start_ts = GST_CLOCK_TIME_NONE;
flacenc->next_ts = GST_CLOCK_TIME_NONE;
flacenc->granulepos_offset = 0;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (FLAC__stream_encoder_get_state (flacenc->encoder) !=
FLAC__STREAM_ENCODER_UNINITIALIZED) {
flacenc->stopped = TRUE;
FLAC__stream_encoder_finish (flacenc->encoder);
}
flacenc->offset = 0;
flacenc->samples_written = 0;
flacenc->channels = 0;
flacenc->depth = 0;
flacenc->sample_rate = 0;
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]);
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;
flacenc->got_headers = FALSE;
flacenc->last_flow = GST_FLOW_OK;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
default:
break;
}
return ret;
}

15
ext/flac/gstflacenc.h
View file

@ -22,6 +22,7 @@
#define __GST_FLAC_ENC_H__
#include <gst/gst.h>
#include <gst/audio/gstaudioencoder.h>
#include <FLAC/all.h>
@ -37,19 +38,15 @@ typedef struct _GstFlacEnc GstFlacEnc;
typedef struct _GstFlacEncClass GstFlacEncClass;
struct _GstFlacEnc {
GstElement element;
GstAudioEncoder element;
/* < private > */
GstPad *sinkpad;
GstPad *srcpad;
GstFlowReturn last_flow; /* save flow from last push so we can pass the
* correct flow return upstream in case the push
* fails for some reason */
guint64 offset;
guint64 samples_written;
gint channels;
gint width;
gint depth;
@ -68,18 +65,14 @@ struct _GstFlacEnc {
GstTagList * tags;
gboolean eos;
/* queue headers until we have them all so we can add streamheaders to caps */
gboolean got_headers;
GList *headers;
/* Timestamp and granulepos tracking */
GstClockTime start_ts;
GstClockTime next_ts;
guint64 granulepos_offset;
};
struct _GstFlacEncClass {
GstElementClass parent_class;
GstAudioEncoderClass parent_class;
};
GType gst_flac_enc_get_type(void);