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 void gst_flac_enc_finalize (GObject * object);
+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 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 void gst_flac_enc_finalize (GObject * object);
static gboolean gst_flac_enc_update_quality (GstFlacEnc * flacenc,
gint quality);
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,
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&src_factory));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&sink_factory));
+ gst_element_class_add_static_pad_template (element_class, &src_factory);
+ gst_element_class_add_static_pad_template (element_class, &sink_factory);
gst_element_class_set_details_simple (element_class, "FLAC audio encoder",
"Codec/Encoder/Audio",
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",
-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
{
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;
+
+ gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
+
+ return TRUE;
+}
+
static void
add_one_tag (const GstTagList * list, const gchar * tag, gpointer 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))
+ if (strcmp (tag, GST_TAG_IMAGE) == 0
+ || strcmp (tag, GST_TAG_PREVIEW_IMAGE) == 0)
return;
comments = gst_tag_to_vorbis_comments (list, tag);
}
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);
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
}
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);
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;
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);
#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;
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,
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) {
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
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) {
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) {
+ }
+
+ 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;
}
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));
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) {
} 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;
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);
g_assert_not_reached ();
}
- gst_buffer_unref (buffer);
-
res = FLAC__stream_encoder_process_interleaved (flacenc->encoder,
(const FLAC__int32 *) data, samples / flacenc->channels);
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;
-}
projects / platform / upstream / gst-plugins-good.git / blobdiff