static void
gst_ivf_parse_update_src_caps (GstIvfParse * ivf)
{
GstCaps *caps;
const gchar *media_type;
if (!ivf->update_caps &&
G_LIKELY (gst_pad_has_current_caps (GST_BASE_PARSE_SRC_PAD (ivf))))
return;
ivf->update_caps = FALSE;
media_type = fourcc_to_media_type (ivf->fourcc);
/* Create src pad caps */
caps = gst_caps_new_simple (media_type, "width", G_TYPE_INT, ivf->width,
"height", G_TYPE_INT, ivf->height, NULL);
if (ivf->fps_n > 0 && ivf->fps_d > 0) {
gst_base_parse_set_frame_rate (GST_BASE_PARSE_CAST (ivf),
ivf->fps_n, ivf->fps_d, 0, 0);
gst_caps_set_simple (caps, "framerate", GST_TYPE_FRACTION, ivf->fps_n,
ivf->fps_d, NULL);
}
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (ivf), caps);
gst_caps_unref (caps);
}
static gboolean
gst_srtp_dec_sink_event_rtcp (GstPad * pad, GstObject * parent,
GstEvent * event)
{
gboolean ret;
GstCaps *caps;
GstSrtpDec *filter = GST_SRTP_DEC (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
gst_event_parse_caps (event, &caps);
ret = gst_srtp_dec_sink_setcaps (pad, parent, caps, TRUE);
gst_event_unref (event);
return ret;
case GST_EVENT_SEGMENT:
/* Make sure to send a caps event downstream before the segment event,
* even if upstream didn't */
if (!gst_pad_has_current_caps (filter->rtcp_srcpad)) {
GstCaps *caps = gst_caps_new_empty_simple ("application/x-rtcp");
gst_pad_set_caps (filter->rtcp_srcpad, caps);
gst_caps_unref (caps);
}
filter->rtcp_has_segment = TRUE;
break;
case GST_EVENT_FLUSH_STOP:
filter->rtcp_has_segment = FALSE;
break;
default:
break;
}
return gst_pad_event_default (pad, parent, event);
}
/* Emitted by decodebin and links decodebin to vlcvideosink.
* Since only one elementary codec stream is fed to decodebin,
* this signal cannot be emitted more than once. */
static void pad_added_cb( GstElement *p_ele, GstPad *p_pad, gpointer p_data )
{
VLC_UNUSED( p_ele );
decoder_t *p_dec = p_data;
decoder_sys_t *p_sys = p_dec->p_sys;
if( likely( gst_pad_has_current_caps( p_pad ) ) )
{
GstPadLinkReturn ret;
GstPad *p_sinkpad;
msg_Dbg( p_dec, "linking the decoder with the vsink");
p_sinkpad = gst_element_get_static_pad(
p_sys->p_decode_out, "sink" );
ret = gst_pad_link( p_pad, p_sinkpad );
if( ret != GST_PAD_LINK_OK )
msg_Err( p_dec, "failed to link decoder with vsink");
gst_object_unref( p_sinkpad );
}
else
{
msg_Err( p_dec, "decodebin src pad has no caps" );
GST_ELEMENT_ERROR( p_sys->p_decoder, STREAM, FAILED,
( "vlc stream error" ), NULL );
}
}
static void
gst_ffmpegdeinterlace_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstFFMpegDeinterlace *self;
g_return_if_fail (GST_IS_FFMPEGDEINTERLACE (object));
self = GST_FFMPEGDEINTERLACE (object);
switch (prop_id) {
case PROP_MODE:{
gint new_mode;
GST_OBJECT_LOCK (self);
new_mode = g_value_get_enum (value);
if (self->mode != new_mode && gst_pad_has_current_caps (self->srcpad)) {
self->reconfigure = TRUE;
self->new_mode = new_mode;
} else {
self->mode = new_mode;
gst_ffmpegdeinterlace_update_passthrough (self);
}
GST_OBJECT_UNLOCK (self);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
}
}
static void
gst_aspect_ratio_crop_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAspectRatioCrop *aspect_ratio_crop;
gboolean recheck = FALSE;
aspect_ratio_crop = GST_ASPECT_RATIO_CROP (object);
GST_OBJECT_LOCK (aspect_ratio_crop);
switch (prop_id) {
case ARG_ASPECT_RATIO_CROP:
if (GST_VALUE_HOLDS_FRACTION (value)) {
aspect_ratio_crop->ar_num = gst_value_get_fraction_numerator (value);
aspect_ratio_crop->ar_denom =
gst_value_get_fraction_denominator (value);
recheck = gst_pad_has_current_caps (aspect_ratio_crop->sink);
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (aspect_ratio_crop);
if (recheck) {
GstCaps *caps = gst_pad_get_current_caps (aspect_ratio_crop->sink);
gst_aspect_ratio_crop_set_caps (aspect_ratio_crop, caps);
gst_caps_unref (caps);
}
}
static void
gst_interleave_release_pad (GstElement * element, GstPad * pad)
{
GstInterleave *self = GST_INTERLEAVE (element);
GList *l;
GstAudioChannelPosition position;
g_return_if_fail (GST_IS_INTERLEAVE_PAD (pad));
/* Take lock to make sure we're not changing this when processing buffers */
GST_OBJECT_LOCK (self->collect);
g_atomic_int_add (&self->channels, -1);
if (gst_pad_has_current_caps (pad))
g_atomic_int_add (&self->configured_sinkpads_counter, -1);
position = GST_INTERLEAVE_PAD_CAST (pad)->channel;
g_value_array_remove (self->input_channel_positions, position);
/* Update channel numbers */
GST_OBJECT_LOCK (self);
for (l = GST_ELEMENT_CAST (self)->sinkpads; l != NULL; l = l->next) {
GstInterleavePad *ipad = GST_INTERLEAVE_PAD (l->data);
if (GST_INTERLEAVE_PAD_CAST (pad)->channel < ipad->channel)
ipad->channel--;
}
GST_OBJECT_UNLOCK (self);
/* Update the src caps if we already have them */
if (self->sinkcaps) {
if (self->channels > 0) {
GstCaps *srccaps;
GstStructure *s;
srccaps = gst_caps_copy (self->sinkcaps);
s = gst_caps_get_structure (srccaps, 0);
gst_structure_set (s, "channels", G_TYPE_INT, self->channels, NULL);
gst_interleave_set_channel_positions (self, s);
gst_pad_set_active (self->src, TRUE);
gst_pad_set_caps (self->src, srccaps);
gst_caps_unref (srccaps);
} else {
gst_caps_replace (&self->sinkcaps, NULL);
}
}
GST_OBJECT_UNLOCK (self->collect);
gst_collect_pads_remove_pad (self->collect, pad);
gst_element_remove_pad (element, pad);
}
static GstFlowReturn
gst_ffmpegdeinterlace_chain (GstPad * pad, GstObject * parent,
GstBuffer * inbuf)
{
GstFFMpegDeinterlace *deinterlace = GST_FFMPEGDEINTERLACE (parent);
GstBuffer *outbuf = NULL;
GstFlowReturn result;
GstMapInfo from_map, to_map;
GST_OBJECT_LOCK (deinterlace);
if (deinterlace->reconfigure) {
if (deinterlace->new_mode != -1)
deinterlace->mode = deinterlace->new_mode;
deinterlace->new_mode = -1;
deinterlace->reconfigure = FALSE;
GST_OBJECT_UNLOCK (deinterlace);
if (gst_pad_has_current_caps (deinterlace->srcpad)) {
GstCaps *caps;
caps = gst_pad_get_current_caps (deinterlace->sinkpad);
gst_ffmpegdeinterlace_sink_setcaps (deinterlace->sinkpad, caps);
gst_caps_unref (caps);
}
} else {
GST_OBJECT_UNLOCK (deinterlace);
}
if (deinterlace->passthrough)
return gst_pad_push (deinterlace->srcpad, inbuf);
outbuf = gst_buffer_new_and_alloc (deinterlace->to_size);
gst_buffer_map (inbuf, &from_map, GST_MAP_READ);
gst_ffmpeg_avpicture_fill (&deinterlace->from_frame, from_map.data,
deinterlace->pixfmt, deinterlace->width, deinterlace->height);
gst_buffer_map (outbuf, &to_map, GST_MAP_WRITE);
gst_ffmpeg_avpicture_fill (&deinterlace->to_frame, to_map.data,
deinterlace->pixfmt, deinterlace->width, deinterlace->height);
avpicture_deinterlace (&deinterlace->to_frame, &deinterlace->from_frame,
deinterlace->pixfmt, deinterlace->width, deinterlace->height);
gst_buffer_unmap (outbuf, &to_map);
gst_buffer_unmap (inbuf, &from_map);
gst_buffer_copy_into (outbuf, inbuf, GST_BUFFER_COPY_TIMESTAMPS, 0, -1);
result = gst_pad_push (deinterlace->srcpad, outbuf);
gst_buffer_unref (inbuf);
return result;
}
static gboolean
gst_dvdec_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstDVDec *dvdec;
gboolean res = TRUE;
dvdec = GST_DVDEC (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&dvdec->segment, GST_FORMAT_UNDEFINED);
dvdec->need_segment = FALSE;
break;
case GST_EVENT_SEGMENT:{
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
GST_DEBUG_OBJECT (dvdec, "Got SEGMENT %" GST_SEGMENT_FORMAT, &segment);
gst_segment_copy_into (segment, &dvdec->segment);
if (!gst_pad_has_current_caps (dvdec->srcpad)) {
dvdec->need_segment = TRUE;
gst_event_unref (event);
event = NULL;
res = TRUE;
} else {
dvdec->need_segment = FALSE;
}
break;
}
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
res = gst_dvdec_sink_setcaps (dvdec, caps);
gst_event_unref (event);
event = NULL;
break;
}
default:
break;
}
if (event)
res = gst_pad_push_event (dvdec->srcpad, event);
return res;
}
/* make sure we are negotiated */
static GstFlowReturn
ensure_negotiated (GstMonoscope * monoscope)
{
gboolean reconfigure;
reconfigure = gst_pad_check_reconfigure (monoscope->srcpad);
/* we don't know an output format yet, pick one */
if (reconfigure || !gst_pad_has_current_caps (monoscope->srcpad)) {
if (!gst_monoscope_src_negotiate (monoscope))
return GST_FLOW_NOT_NEGOTIATED;
}
return GST_FLOW_OK;
}
static int
gst_dv1394src_iec61883_receive (unsigned char *data, int len,
int complete, void *cbdata)
{
GstDV1394Src *dv1394src = GST_DV1394SRC (cbdata);
if (G_UNLIKELY (!gst_pad_has_current_caps (GST_BASE_SRC_PAD (dv1394src)))) {
GstCaps *caps;
unsigned char *p = data;
// figure format (NTSC/PAL)
if (p[3] & 0x80) {
// PAL
dv1394src->frame_size = PAL_FRAMESIZE;
dv1394src->frame_rate = PAL_FRAMERATE;
GST_DEBUG ("PAL data");
caps = gst_caps_new_simple ("video/x-dv",
"format", G_TYPE_STRING, "PAL",
"systemstream", G_TYPE_BOOLEAN, TRUE, NULL);
} else {
// NTSC (untested)
dv1394src->frame_size = NTSC_FRAMESIZE;
dv1394src->frame_rate = NTSC_FRAMERATE;
GST_DEBUG
("NTSC data [untested] - please report success/failure to <*****@*****.**>");
caps = gst_caps_new_simple ("video/x-dv",
"format", G_TYPE_STRING, "NTSC",
"systemstream", G_TYPE_BOOLEAN, TRUE, NULL);
}
gst_pad_set_caps (GST_BASE_SRC_PAD (dv1394src), caps);
gst_caps_unref (caps);
}
dv1394src->frame = NULL;
if (G_LIKELY ((dv1394src->frame_sequence + 1) % (dv1394src->skip +
dv1394src->consecutive) < dv1394src->consecutive)) {
if (complete && len == dv1394src->frame_size) {
GstBuffer *buf;
buf = gst_buffer_new_and_alloc (dv1394src->frame_size);
GST_BUFFER_OFFSET (buf) = dv1394src->frame_sequence;
gst_buffer_fill (buf, 0, data, len);
dv1394src->buf = buf;
}
}
dv1394src->frame_sequence++;
return 0;
}
static GstCaps *
gst_data_uri_src_get_caps (GstBaseSrc * basesrc, GstCaps * filter)
{
GstDataURISrc *src = GST_DATA_URI_SRC (basesrc);
GstCaps *caps;
GST_OBJECT_LOCK (src);
if (gst_pad_has_current_caps (GST_BASE_SRC_PAD (basesrc)))
caps = gst_pad_get_current_caps (GST_BASE_SRC_PAD (basesrc));
else
caps = gst_caps_new_any ();
GST_OBJECT_UNLOCK (src);
return caps;
}
static GstFlowReturn
gst_kate_parse_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstKateParseClass *klass;
GstKateParse *parse;
parse = GST_KATE_PARSE (parent);
klass = GST_KATE_PARSE_CLASS (G_OBJECT_GET_CLASS (parse));
g_assert (klass->parse_packet != NULL);
if (G_UNLIKELY (!gst_pad_has_current_caps (pad)))
return GST_FLOW_NOT_NEGOTIATED;
return klass->parse_packet (parse, buffer);
}
/* Queue the segment event if there was no caps event */
static gboolean
gst_capsfilter_sink_event (GstBaseTransform * trans, GstEvent * event)
{
GstCapsFilter *filter = GST_CAPSFILTER (trans);
if (GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
GList *l;
for (l = filter->pending_events; l;) {
if (GST_EVENT_TYPE (l->data) == GST_EVENT_SEGMENT) {
gst_event_unref (l->data);
l = g_list_delete_link (l, l);
} else {
l = l->next;
}
}
}
if (!GST_EVENT_IS_STICKY (event) || GST_EVENT_TYPE (event) <= GST_EVENT_CAPS)
goto done;
/* If we get EOS before any buffers, just push all pending events */
if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
GList *l;
for (l = g_list_last (filter->pending_events); l; l = l->prev) {
GST_LOG_OBJECT (trans, "Forwarding %s event",
GST_EVENT_TYPE_NAME (l->data));
GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (trans, l->data);
}
g_list_free (filter->pending_events);
filter->pending_events = NULL;
} else if (!gst_pad_has_current_caps (trans->sinkpad)) {
GST_LOG_OBJECT (trans, "Got %s event before caps, queueing",
GST_EVENT_TYPE_NAME (event));
filter->pending_events = g_list_prepend (filter->pending_events, event);
return TRUE;
}
done:
GST_LOG_OBJECT (trans, "Forwarding %s event", GST_EVENT_TYPE_NAME (event));
return GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (trans, event);
}
static gboolean
gst_flxdec_sink_event_handler (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstFlxDec *flxdec;
gboolean ret;
flxdec = GST_FLXDEC (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:
{
gst_event_copy_segment (event, &flxdec->segment);
if (flxdec->segment.format != GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (flxdec, "generating TIME segment");
gst_segment_init (&flxdec->segment, GST_FORMAT_TIME);
gst_event_unref (event);
event = gst_event_new_segment (&flxdec->segment);
}
if (gst_pad_has_current_caps (flxdec->srcpad)) {
ret = gst_pad_event_default (pad, parent, event);
} else {
flxdec->need_segment = TRUE;
gst_event_unref (event);
ret = TRUE;
}
break;
}
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&flxdec->segment, GST_FORMAT_UNDEFINED);
ret = gst_pad_event_default (pad, parent, event);
break;
default:
ret = gst_pad_event_default (pad, parent, event);
break;
}
return ret;
}
static GstCaps *
gst_image_freeze_sink_getcaps (GstImageFreeze * self, GstCaps * filter)
{
GstCaps *ret, *tmp, *templ;
GstPad *pad;
pad = self->sinkpad;
if (gst_pad_has_current_caps (pad)) {
ret = gst_pad_get_current_caps (pad);
goto done;
}
if (filter) {
filter = gst_caps_copy (filter);
gst_image_freeze_remove_fps (self, filter);
}
templ = gst_pad_get_pad_template_caps (pad);
tmp = gst_pad_peer_query_caps (self->srcpad, filter);
if (tmp) {
GST_LOG_OBJECT (self, "peer caps %" GST_PTR_FORMAT, tmp);
ret = gst_caps_intersect (tmp, templ);
gst_caps_unref (tmp);
} else {
GST_LOG_OBJECT (self, "going to copy");
ret = gst_caps_copy (templ);
}
gst_caps_unref (templ);
if (filter)
gst_caps_unref (filter);
ret = gst_caps_make_writable (ret);
gst_image_freeze_remove_fps (self, ret);
done:
GST_LOG_OBJECT (pad, "Returning caps: %" GST_PTR_FORMAT, ret);
return ret;
}
static GstFlowReturn
gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstWavpackDec *dec;
GstBuffer *outbuf = NULL;
GstFlowReturn ret = GST_FLOW_OK;
WavpackHeader wph;
int32_t decoded, unpacked_size;
gboolean format_changed;
gint width, depth, i, j, max;
gint32 *dec_data = NULL;
guint8 *out_data;
GstMapInfo map, omap;
dec = GST_WAVPACK_DEC (bdec);
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
gst_buffer_map (buf, &map, GST_MAP_READ);
/* check input, we only accept framed input with complete chunks */
if (map.size < sizeof (WavpackHeader))
goto input_not_framed;
if (!gst_wavpack_read_header (&wph, map.data))
goto invalid_header;
if (map.size < wph.ckSize + 4 * 1 + 4)
goto input_not_framed;
if (!(wph.flags & INITIAL_BLOCK))
goto input_not_framed;
dec->wv_id.buffer = map.data;
dec->wv_id.length = map.size;
dec->wv_id.position = 0;
/* create a new wavpack context if there is none yet but if there
* was already one (i.e. caps were set on the srcpad) check whether
* the new one has the same caps */
if (!dec->context) {
gchar error_msg[80];
dec->context = WavpackOpenFileInputEx (dec->stream_reader,
&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
/* expect this to work */
if (!dec->context) {
GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg);
goto context_failed;
}
}
g_assert (dec->context != NULL);
format_changed =
(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
(dec->channels != WavpackGetNumChannels (dec->context)) ||
(dec->depth != WavpackGetBytesPerSample (dec->context) * 8) ||
(dec->channel_mask != WavpackGetChannelMask (dec->context));
if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (dec)) ||
format_changed) {
gint channel_mask;
dec->sample_rate = WavpackGetSampleRate (dec->context);
dec->channels = WavpackGetNumChannels (dec->context);
dec->depth = WavpackGetBytesPerSample (dec->context) * 8;
channel_mask = WavpackGetChannelMask (dec->context);
if (channel_mask == 0)
channel_mask = gst_wavpack_get_default_channel_mask (dec->channels);
dec->channel_mask = channel_mask;
gst_wavpack_dec_negotiate (dec);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
gst_wavpack_dec_post_tags (dec);
}
/* alloc output buffer */
dec_data = g_malloc (4 * wph.block_samples * dec->channels);
/* decode */
decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
outbuf = gst_buffer_new_and_alloc (unpacked_size);
/* legacy; pass along offset, whatever that might entail */
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET (buf);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = omap.data;
width = dec->width;
depth = dec->depth;
max = dec->channels * wph.block_samples;
if (width == 8) {
gint8 *outbuffer = (gint8 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint8) (dec_data[i + reorder_map[j]]);
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint16) (dec_data[i + reorder_map[j]]);
}
} else if (dec->width == 32) {
gint32 *outbuffer = (gint32 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
if (width != depth) {
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ =
(gint32) (dec_data[i + reorder_map[j]] << (width - depth));
}
} else {
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint32) (dec_data[i + reorder_map[j]]);
}
}
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unmap (buf, &map);
buf = NULL;
g_free (dec_data);
ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
out:
if (buf)
gst_buffer_unmap (buf, &map);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
}
return ret;
/* ERRORS */
input_not_framed:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
ret = GST_FLOW_ERROR;
goto out;
}
invalid_header:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
ret = GST_FLOW_ERROR;
goto out;
}
context_failed:
{
GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL),
("error creating Wavpack context"), ret);
goto out;
}
decode_error:
{
const gchar *reason = "unknown";
if (dec->context) {
reason = WavpackGetErrorMessage (dec->context);
} else {
reason = "couldn't create decoder context";
}
GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
("decoding error: %s", reason), ret);
g_free (dec_data);
if (ret == GST_FLOW_OK)
gst_audio_decoder_finish_frame (bdec, NULL, 1);
goto out;
}
}
static void
gst_omx_audio_dec_loop (GstOMXAudioDec * self)
{
GstOMXPort *port = self->dec_out_port;
GstOMXBuffer *buf = NULL;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstOMXAcquireBufferReturn acq_return;
OMX_ERRORTYPE err;
acq_return = gst_omx_port_acquire_buffer (port, &buf);
if (acq_return == GST_OMX_ACQUIRE_BUFFER_ERROR) {
goto component_error;
} else if (acq_return == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
goto flushing;
} else if (acq_return == GST_OMX_ACQUIRE_BUFFER_EOS) {
goto eos;
}
if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (self)) ||
acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
OMX_PARAM_PORTDEFINITIONTYPE port_def;
OMX_AUDIO_PARAM_PCMMODETYPE pcm_param;
GstAudioChannelPosition omx_position[OMX_AUDIO_MAXCHANNELS];
GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
gint i;
GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps");
/* Reallocate all buffers */
if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE
&& gst_omx_port_is_enabled (port)) {
err = gst_omx_port_set_enabled (port, FALSE);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_deallocate_buffers (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
}
/* Just update caps */
GST_AUDIO_DECODER_STREAM_LOCK (self);
gst_omx_port_get_port_definition (port, &port_def);
g_assert (port_def.format.audio.eEncoding == OMX_AUDIO_CodingPCM);
GST_OMX_INIT_STRUCT (&pcm_param);
pcm_param.nPortIndex = self->dec_out_port->index;
err =
gst_omx_component_get_parameter (self->dec, OMX_IndexParamAudioPcm,
&pcm_param);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Failed to get PCM parameters: %s (0x%08x)",
gst_omx_error_to_string (err), err);
goto caps_failed;
}
g_assert (pcm_param.ePCMMode == OMX_AUDIO_PCMModeLinear);
g_assert (pcm_param.bInterleaved == OMX_TRUE);
gst_audio_info_init (&self->info);
for (i = 0; i < pcm_param.nChannels; i++) {
switch (pcm_param.eChannelMapping[i]) {
case OMX_AUDIO_ChannelLF:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
break;
case OMX_AUDIO_ChannelRF:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
break;
case OMX_AUDIO_ChannelCF:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
break;
case OMX_AUDIO_ChannelLS:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
break;
case OMX_AUDIO_ChannelRS:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
break;
case OMX_AUDIO_ChannelLFE:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_LFE1;
break;
case OMX_AUDIO_ChannelCS:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
break;
case OMX_AUDIO_ChannelLR:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
break;
case OMX_AUDIO_ChannelRR:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
break;
case OMX_AUDIO_ChannelNone:
default:
/* This will break the outer loop too as the
* i == pcm_param.nChannels afterwards */
for (i = 0; i < pcm_param.nChannels; i++)
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
break;
}
}
if (pcm_param.nChannels == 1
&& omx_position[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER)
omx_position[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
if (omx_position[0] == GST_AUDIO_CHANNEL_POSITION_NONE
&& klass->get_channel_positions) {
GST_WARNING_OBJECT (self,
"Failed to get a valid channel layout, trying fallback");
klass->get_channel_positions (self, self->dec_out_port, omx_position);
}
memcpy (self->position, omx_position, sizeof (omx_position));
gst_audio_channel_positions_to_valid_order (self->position,
pcm_param.nChannels);
self->needs_reorder =
(memcmp (self->position, omx_position,
sizeof (GstAudioChannelPosition) * pcm_param.nChannels) != 0);
if (self->needs_reorder)
gst_audio_get_channel_reorder_map (pcm_param.nChannels, self->position,
omx_position, self->reorder_map);
gst_audio_info_set_format (&self->info,
gst_audio_format_build_integer (pcm_param.eNumData ==
OMX_NumericalDataSigned,
pcm_param.eEndian ==
OMX_EndianLittle ? G_LITTLE_ENDIAN : G_BIG_ENDIAN,
pcm_param.nBitPerSample, pcm_param.nBitPerSample),
pcm_param.nSamplingRate, pcm_param.nChannels, self->position);
GST_DEBUG_OBJECT (self,
"Setting output state: format %s, rate %u, channels %u",
gst_audio_format_to_string (self->info.finfo->format),
(guint) pcm_param.nSamplingRate, (guint) pcm_param.nChannels);
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (self),
&self->info)
|| !gst_audio_decoder_negotiate (GST_AUDIO_DECODER (self))) {
if (buf)
gst_omx_port_release_buffer (port, buf);
goto caps_failed;
}
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
err = gst_omx_port_set_enabled (port, TRUE);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_allocate_buffers (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_populate (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_mark_reconfigured (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
}
/* Now get a buffer */
if (acq_return != GST_OMX_ACQUIRE_BUFFER_OK) {
return;
}
}
g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK);
if (!buf) {
g_assert ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER));
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto eos;
}
/* This prevents a deadlock between the srcpad stream
* lock and the audiocodec stream lock, if ::reset()
* is called at the wrong time
*/
if (gst_omx_port_is_flushing (port)) {
GST_DEBUG_OBJECT (self, "Flushing");
gst_omx_port_release_buffer (port, buf);
goto flushing;
}
GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %" G_GUINT64_FORMAT,
(guint) buf->omx_buf->nFlags, (guint64) buf->omx_buf->nTimeStamp);
GST_AUDIO_DECODER_STREAM_LOCK (self);
if (buf->omx_buf->nFilledLen > 0) {
GstBuffer *outbuf;
gint nframes, spf;
GstMapInfo minfo;
GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
GST_DEBUG_OBJECT (self, "Handling output data");
if (buf->omx_buf->nFilledLen % self->info.bpf != 0) {
gst_omx_port_release_buffer (port, buf);
goto invalid_buffer;
}
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self),
buf->omx_buf->nFilledLen);
gst_buffer_map (outbuf, &minfo, GST_MAP_WRITE);
if (self->needs_reorder) {
gint i, n_samples, c, n_channels;
gint *reorder_map = self->reorder_map;
gint16 *dest, *source;
dest = (gint16 *) minfo.data;
source = (gint16 *) (buf->omx_buf->pBuffer + buf->omx_buf->nOffset);
n_samples = buf->omx_buf->nFilledLen / self->info.bpf;
n_channels = self->info.channels;
for (i = 0; i < n_samples; i++) {
for (c = 0; c < n_channels; c++) {
dest[i * n_channels + reorder_map[c]] = source[i * n_channels + c];
}
}
} else {
memcpy (minfo.data, buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
}
gst_buffer_unmap (outbuf, &minfo);
nframes = 1;
spf = klass->get_samples_per_frame (self, self->dec_out_port);
if (spf != -1) {
nframes = buf->omx_buf->nFilledLen / self->info.bpf;
if (nframes % spf != 0)
GST_WARNING_OBJECT (self, "Output buffer does not contain an integer "
"number of input frames (frames: %d, spf: %d)", nframes, spf);
nframes = (nframes + spf - 1) / spf;
}
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
OMX_TICKS_PER_SECOND);
if (buf->omx_buf->nTickCount != 0)
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
OMX_TICKS_PER_SECOND);
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf,
nframes);
}
GST_DEBUG_OBJECT (self, "Read frame from component");
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
if (buf) {
err = gst_omx_port_release_buffer (port, buf);
if (err != OMX_ErrorNone)
goto release_error;
}
self->downstream_flow_ret = flow_ret;
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
component_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("OpenMAX component in error state %s (0x%08x)",
gst_omx_component_get_last_error_string (self->dec),
gst_omx_component_get_last_error (self->dec)));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
return;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
return;
}
eos:
{
g_mutex_lock (&self->drain_lock);
if (self->draining) {
GST_DEBUG_OBJECT (self, "Drained");
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
flow_ret = GST_FLOW_OK;
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
} else {
GST_DEBUG_OBJECT (self, "Component signalled EOS");
flow_ret = GST_FLOW_EOS;
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_DECODER_STREAM_LOCK (self);
self->downstream_flow_ret = flow_ret;
/* Here we fallback and pause the task for the EOS case */
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
flow_error:
{
if (flow_ret == GST_FLOW_EOS) {
GST_DEBUG_OBJECT (self, "EOS");
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->started = FALSE;
} else if (flow_ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
("Internal data stream error."), ("stream stopped, reason %s",
gst_flow_get_name (flow_ret)));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->started = FALSE;
} else if (flow_ret == GST_FLOW_FLUSHING) {
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->started = FALSE;
}
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
reconfigure_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Unable to reconfigure output port"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
return;
}
invalid_buffer:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Invalid sized input buffer"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
caps_failed:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE;
return;
}
release_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to relase output buffer to component: %s (0x%08x)",
gst_omx_error_to_string (err), err));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
}
static GstFlowReturn
gst_app_sink_render (GstBaseSink * psink, GstBuffer * buffer)
{
GstFlowReturn ret;
GstAppSink *appsink = GST_APP_SINK_CAST (psink);
GstAppSinkPrivate *priv = appsink->priv;
gboolean emit;
restart:
g_mutex_lock (&priv->mutex);
if (priv->flushing)
goto flushing;
/* queue holding caps event might have been FLUSHed,
* but caps state still present in pad caps */
if (G_UNLIKELY (!priv->last_caps &&
gst_pad_has_current_caps (GST_BASE_SINK_PAD (psink)))) {
priv->last_caps = gst_pad_get_current_caps (GST_BASE_SINK_PAD (psink));
GST_DEBUG_OBJECT (appsink, "activating pad caps %" GST_PTR_FORMAT,
priv->last_caps);
}
GST_DEBUG_OBJECT (appsink, "pushing render buffer %p on queue (%d)",
buffer, priv->num_buffers);
while (priv->max_buffers > 0 && priv->num_buffers >= priv->max_buffers) {
if (priv->drop) {
GstBuffer *old;
/* we need to drop the oldest buffer and try again */
if ((old = dequeue_buffer (appsink))) {
GST_DEBUG_OBJECT (appsink, "dropping old buffer %p", old);
gst_buffer_unref (old);
}
} else {
GST_DEBUG_OBJECT (appsink, "waiting for free space, length %d >= %d",
priv->num_buffers, priv->max_buffers);
if (priv->unlock) {
/* we are asked to unlock, call the wait_preroll method */
g_mutex_unlock (&priv->mutex);
if ((ret = gst_base_sink_wait_preroll (psink)) != GST_FLOW_OK)
goto stopping;
/* we are allowed to continue now */
goto restart;
}
/* wait for a buffer to be removed or flush */
g_cond_wait (&priv->cond, &priv->mutex);
if (priv->flushing)
goto flushing;
}
}
/* we need to ref the buffer when pushing it in the queue */
g_queue_push_tail (priv->queue, gst_buffer_ref (buffer));
priv->num_buffers++;
g_cond_signal (&priv->cond);
emit = priv->emit_signals;
g_mutex_unlock (&priv->mutex);
if (priv->callbacks.new_sample) {
ret = priv->callbacks.new_sample (appsink, priv->user_data);
} else {
ret = GST_FLOW_OK;
if (emit)
g_signal_emit (appsink, gst_app_sink_signals[SIGNAL_NEW_SAMPLE], 0, &ret);
}
return ret;
flushing:
{
GST_DEBUG_OBJECT (appsink, "we are flushing");
g_mutex_unlock (&priv->mutex);
return GST_FLOW_FLUSHING;
}
stopping:
{
GST_DEBUG_OBJECT (appsink, "we are stopping");
return ret;
}
}
static void
gst_mpegv_parse_update_src_caps (GstMpegvParse * mpvparse)
{
GstCaps *caps = NULL;
/* only update if no src caps yet or explicitly triggered */
if (G_LIKELY (gst_pad_has_current_caps (GST_BASE_PARSE_SRC_PAD (mpvparse)) &&
!mpvparse->update_caps))
return;
/* carry over input caps as much as possible; override with our own stuff */
caps = gst_pad_get_current_caps (GST_BASE_PARSE_SINK_PAD (mpvparse));
if (caps) {
caps = gst_caps_make_writable (caps);
} else {
caps = gst_caps_new_empty_simple ("video/mpeg");
}
/* typically we don't output buffers until we have properly parsed some
* config data, so we should at least know about version.
* If not, it means it has been requested not to drop data, and
* upstream and/or app must know what they are doing ... */
gst_caps_set_simple (caps,
"mpegversion", G_TYPE_INT, (mpvparse->config_flags & FLAG_MPEG2) ? 2 : 1,
NULL);
gst_caps_set_simple (caps, "systemstream", G_TYPE_BOOLEAN, FALSE,
"parsed", G_TYPE_BOOLEAN, TRUE, NULL);
if (mpvparse->sequencehdr.width > 0 && mpvparse->sequencehdr.height > 0) {
gst_caps_set_simple (caps, "width", G_TYPE_INT, mpvparse->sequencehdr.width,
"height", G_TYPE_INT, mpvparse->sequencehdr.height, NULL);
}
/* perhaps we have a framerate */
if (mpvparse->fps_num > 0 && mpvparse->fps_den > 0) {
gint fps_num = mpvparse->fps_num;
gint fps_den = mpvparse->fps_den;
GstClockTime latency = gst_util_uint64_scale (GST_SECOND, fps_den, fps_num);
gst_caps_set_simple (caps, "framerate",
GST_TYPE_FRACTION, fps_num, fps_den, NULL);
gst_base_parse_set_frame_rate (GST_BASE_PARSE (mpvparse),
fps_num, fps_den, 0, 0);
gst_base_parse_set_latency (GST_BASE_PARSE (mpvparse), latency, latency);
}
/* or pixel-aspect-ratio */
if (mpvparse->sequencehdr.par_w && mpvparse->sequencehdr.par_h > 0) {
gst_caps_set_simple (caps, "pixel-aspect-ratio", GST_TYPE_FRACTION,
mpvparse->sequencehdr.par_w, mpvparse->sequencehdr.par_h, NULL);
}
if (mpvparse->config != NULL) {
gst_caps_set_simple (caps, "codec_data",
GST_TYPE_BUFFER, mpvparse->config, NULL);
}
if (mpvparse->config_flags & FLAG_SEQUENCE_EXT) {
const guint profile_c = mpvparse->sequenceext.profile;
const guint level_c = mpvparse->sequenceext.level;
const gchar *profile = NULL, *level = NULL;
/*
* Profile indication - 1 => High, 2 => Spatially Scalable,
* 3 => SNR Scalable, 4 => Main, 5 => Simple
* 4:2:2 and Multi-view have profile = 0, with the escape bit set to 1
*/
const gchar *const profiles[] =
{ "high", "spatial", "snr", "main", "simple" };
/*
* Level indication - 4 => High, 6 => High-1440, 8 => Main, 10 => Low,
* except in the case of profile = 0
*/
const gchar *const levels[] = { "high", "high-1440", "main", "low" };
if (profile_c > 0 && profile_c < 6)
profile = profiles[profile_c - 1];
if ((level_c > 3) && (level_c < 11) && (level_c % 2 == 0))
level = levels[(level_c >> 1) - 2];
if (profile_c == 8) {
/* Non-hierarchical profile */
switch (level_c) {
case 2:
level = levels[0];
case 5:
level = levels[2];
profile = "4:2:2";
break;
case 10:
level = levels[0];
case 11:
level = levels[1];
case 13:
level = levels[2];
case 14:
level = levels[3];
profile = "multiview";
break;
default:
break;
}
}
/* FIXME does it make sense to expose profile/level in the caps ? */
GST_DEBUG_OBJECT (mpvparse, "profile:'%s' level:'%s'", profile, level);
if (profile)
gst_caps_set_simple (caps, "profile", G_TYPE_STRING, profile, NULL);
else
GST_DEBUG_OBJECT (mpvparse, "Invalid profile - %u", profile_c);
if (level)
gst_caps_set_simple (caps, "level", G_TYPE_STRING, level, NULL);
else
GST_DEBUG_OBJECT (mpvparse, "Invalid level - %u", level_c);
gst_caps_set_simple (caps, "interlace-mode",
G_TYPE_STRING,
(mpvparse->sequenceext.progressive ? "progressive" : "mixed"), NULL);
}
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (mpvparse), caps);
gst_caps_unref (caps);
mpvparse->update_caps = FALSE;
}
static GstFlowReturn gst_openh264dec_handle_frame(GstVideoDecoder *decoder, GstVideoCodecFrame *frame)
{
GstOpenh264Dec *openh264dec = GST_OPENH264DEC(decoder);
GstMapInfo map_info;
GstVideoCodecState *state;
SBufferInfo dst_buf_info;
DECODING_STATE ret;
guint8 *yuvdata[3];
GstFlowReturn flow_status;
GstVideoFrame video_frame;
guint actual_width, actual_height;
guint i;
guint8 *p;
guint row_stride, component_width, component_height, src_width, row;
if (frame) {
if (!gst_buffer_map(frame->input_buffer, &map_info, GST_MAP_READ)) {
GST_ERROR_OBJECT(openh264dec, "Cannot map input buffer!");
return GST_FLOW_ERROR;
}
GST_LOG_OBJECT(openh264dec, "handle frame, %d", map_info.size > 4 ? map_info.data[4] & 0x1f : -1);
memset (&dst_buf_info, 0, sizeof (SBufferInfo));
ret = openh264dec->priv->decoder->DecodeFrame2(map_info.data, map_info.size, yuvdata, &dst_buf_info);
if (ret == dsNoParamSets) {
GST_DEBUG_OBJECT(openh264dec, "Requesting a key unit");
gst_pad_push_event(GST_VIDEO_DECODER_SINK_PAD(decoder),
gst_video_event_new_upstream_force_key_unit(GST_CLOCK_TIME_NONE, FALSE, 0));
}
if (ret != dsErrorFree && ret != dsNoParamSets) {
GST_DEBUG_OBJECT(openh264dec, "Requesting a key unit");
gst_pad_push_event(GST_VIDEO_DECODER_SINK_PAD(decoder),
gst_video_event_new_upstream_force_key_unit(GST_CLOCK_TIME_NONE, FALSE, 0));
GST_LOG_OBJECT(openh264dec, "error decoding nal, return code: %d", ret);
}
gst_buffer_unmap(frame->input_buffer, &map_info);
gst_video_codec_frame_unref (frame);
frame = NULL;
} else {
memset (&dst_buf_info, 0, sizeof (SBufferInfo));
ret = openh264dec->priv->decoder->DecodeFrame2(NULL, 0, yuvdata, &dst_buf_info);
if (ret != dsErrorFree)
return GST_FLOW_EOS;
}
/* FIXME: openh264 has no way for us to get a connection
* between the input and output frames, we just have to
* guess based on the input. Fortunately openh264 can
* only do baseline profile. */
frame = gst_video_decoder_get_oldest_frame (decoder);
if (!frame) {
/* Can only happen in finish() */
return GST_FLOW_EOS;
}
/* No output available yet */
if (dst_buf_info.iBufferStatus != 1) {
return (frame ? GST_FLOW_OK : GST_FLOW_EOS);
}
actual_width = dst_buf_info.UsrData.sSystemBuffer.iWidth;
actual_height = dst_buf_info.UsrData.sSystemBuffer.iHeight;
if (!gst_pad_has_current_caps (GST_VIDEO_DECODER_SRC_PAD (openh264dec)) || actual_width != openh264dec->priv->width || actual_height != openh264dec->priv->height) {
state = gst_video_decoder_set_output_state(decoder,
GST_VIDEO_FORMAT_I420,
actual_width,
actual_height,
openh264dec->priv->input_state);
openh264dec->priv->width = actual_width;
openh264dec->priv->height = actual_height;
if (!gst_video_decoder_negotiate(decoder)) {
GST_ERROR_OBJECT(openh264dec, "Failed to negotiate with downstream elements");
return GST_FLOW_NOT_NEGOTIATED;
}
} else {
state = gst_video_decoder_get_output_state(decoder);
}
flow_status = gst_video_decoder_allocate_output_frame(decoder, frame);
if (flow_status != GST_FLOW_OK) {
gst_video_codec_state_unref (state);
return flow_status;
}
if (!gst_video_frame_map(&video_frame, &state->info, frame->output_buffer, GST_MAP_WRITE)) {
GST_ERROR_OBJECT(openh264dec, "Cannot map output buffer!");
gst_video_codec_state_unref (state);
return GST_FLOW_ERROR;
}
for (i = 0; i < 3; i++) {
p = GST_VIDEO_FRAME_COMP_DATA(&video_frame, i);
row_stride = GST_VIDEO_FRAME_COMP_STRIDE(&video_frame, i);
component_width = GST_VIDEO_FRAME_COMP_WIDTH(&video_frame, i);
component_height = GST_VIDEO_FRAME_COMP_HEIGHT(&video_frame, i);
src_width = i < 1 ? dst_buf_info.UsrData.sSystemBuffer.iStride[0] : dst_buf_info.UsrData.sSystemBuffer.iStride[1];
for (row = 0; row < component_height; row++) {
memcpy(p, yuvdata[i], component_width);
p += row_stride;
yuvdata[i] += src_width;
}
}
gst_video_codec_state_unref (state);
gst_video_frame_unmap(&video_frame);
return gst_video_decoder_finish_frame(decoder, frame);
}
static gboolean
gst_interleave_sink_event (GstCollectPads * pads, GstCollectData * data,
GstEvent * event, gpointer user_data)
{
GstInterleave *self = GST_INTERLEAVE (user_data);
gboolean ret = TRUE;
GST_DEBUG ("Got %s event on pad %s:%s", GST_EVENT_TYPE_NAME (event),
GST_DEBUG_PAD_NAME (data->pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
GST_OBJECT_LOCK (self);
gst_event_replace (&self->pending_segment, NULL);
GST_OBJECT_UNLOCK (self);
break;
case GST_EVENT_SEGMENT:
{
GST_OBJECT_LOCK (self);
gst_event_replace (&self->pending_segment, event);
GST_OBJECT_UNLOCK (self);
break;
}
case GST_EVENT_CAPS:
{
GstCaps *caps;
GstAudioInfo info;
GValue *val;
guint channel;
gst_event_parse_caps (event, &caps);
if (!gst_audio_info_from_caps (&info, caps)) {
GST_WARNING_OBJECT (self, "invalid sink caps");
gst_event_unref (event);
event = NULL;
ret = FALSE;
break;
}
if (self->channel_positions_from_input
&& GST_AUDIO_INFO_CHANNELS (&info) == 1) {
channel = GST_INTERLEAVE_PAD_CAST (data->pad)->channel;
val = g_value_array_get_nth (self->input_channel_positions, channel);
g_value_set_enum (val, GST_AUDIO_INFO_POSITION (&info, 0));
}
if (!gst_pad_has_current_caps (data->pad))
g_atomic_int_add (&self->configured_sinkpads_counter, 1);
/* Last caps that are set on a sink pad are used as output caps */
if (g_atomic_int_get (&self->configured_sinkpads_counter) ==
self->channels) {
ret = gst_interleave_sink_setcaps (self, data->pad, caps, &info);
gst_event_unref (event);
event = NULL;
}
break;
}
case GST_EVENT_TAG:
GST_FIXME_OBJECT (self, "FIXME: merge tags and send after stream-start");
break;
default:
break;
}
/* now GstCollectPads can take care of the rest, e.g. EOS */
if (event != NULL)
return gst_collect_pads_event_default (pads, data, event, FALSE);
return ret;
}
static void
gst_image_freeze_src_loop (GstPad * pad)
{
GstImageFreeze *self = GST_IMAGE_FREEZE (GST_PAD_PARENT (pad));
GstBuffer *buffer;
guint64 offset;
GstClockTime timestamp, timestamp_end;
guint64 cstart, cstop;
gboolean in_seg, eos;
GstFlowReturn flow_ret = GST_FLOW_OK;
g_mutex_lock (&self->lock);
if (!gst_pad_has_current_caps (self->srcpad)) {
GST_ERROR_OBJECT (pad, "Not negotiated yet");
flow_ret = GST_FLOW_NOT_NEGOTIATED;
g_mutex_unlock (&self->lock);
goto pause_task;
}
if (!self->buffer) {
GST_ERROR_OBJECT (pad, "Have no buffer yet");
flow_ret = GST_FLOW_ERROR;
g_mutex_unlock (&self->lock);
goto pause_task;
}
buffer = gst_buffer_ref (self->buffer);
buffer = gst_buffer_make_writable (buffer);
g_mutex_unlock (&self->lock);
if (self->need_segment) {
GstEvent *e;
GST_DEBUG_OBJECT (pad, "Pushing SEGMENT event: %" GST_SEGMENT_FORMAT,
&self->segment);
e = gst_event_new_segment (&self->segment);
g_mutex_lock (&self->lock);
if (self->segment.rate >= 0) {
self->offset =
gst_util_uint64_scale (self->segment.start, self->fps_n,
self->fps_d * GST_SECOND);
} else {
self->offset =
gst_util_uint64_scale (self->segment.stop, self->fps_n,
self->fps_d * GST_SECOND);
}
g_mutex_unlock (&self->lock);
self->need_segment = FALSE;
gst_pad_push_event (self->srcpad, e);
}
g_mutex_lock (&self->lock);
offset = self->offset;
if (self->fps_n != 0) {
timestamp =
gst_util_uint64_scale (offset, self->fps_d * GST_SECOND, self->fps_n);
timestamp_end =
gst_util_uint64_scale (offset + 1, self->fps_d * GST_SECOND,
self->fps_n);
} else {
timestamp = self->segment.start;
timestamp_end = GST_CLOCK_TIME_NONE;
}
eos = (self->fps_n == 0 && offset > 0) ||
(self->segment.rate >= 0 && self->segment.stop != -1
&& timestamp > self->segment.stop) || (self->segment.rate < 0
&& offset == 0) || (self->segment.rate < 0
&& self->segment.start != -1 && timestamp_end < self->segment.start);
if (self->fps_n == 0 && offset > 0)
in_seg = FALSE;
else
in_seg =
gst_segment_clip (&self->segment, GST_FORMAT_TIME, timestamp,
timestamp_end, &cstart, &cstop);
if (in_seg) {
self->segment.position = cstart;
if (self->segment.rate >= 0)
self->segment.position = cstop;
}
if (self->segment.rate >= 0)
self->offset++;
else
self->offset--;
g_mutex_unlock (&self->lock);
GST_DEBUG_OBJECT (pad, "Handling buffer with timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (in_seg) {
GST_BUFFER_DTS (buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_PTS (buffer) = cstart;
GST_BUFFER_DURATION (buffer) = cstop - cstart;
GST_BUFFER_OFFSET (buffer) = offset;
GST_BUFFER_OFFSET_END (buffer) = offset + 1;
flow_ret = gst_pad_push (self->srcpad, buffer);
GST_DEBUG_OBJECT (pad, "Pushing buffer resulted in %s",
gst_flow_get_name (flow_ret));
if (flow_ret != GST_FLOW_OK)
goto pause_task;
} else {
gst_buffer_unref (buffer);
}
if (eos) {
flow_ret = GST_FLOW_EOS;
goto pause_task;
}
return;
pause_task:
{
const gchar *reason = gst_flow_get_name (flow_ret);
GST_LOG_OBJECT (self, "pausing task, reason %s", reason);
gst_pad_pause_task (pad);
if (flow_ret == GST_FLOW_EOS) {
if ((self->segment.flags & GST_SEEK_FLAG_SEGMENT)) {
GstMessage *m;
GstEvent *e;
GST_DEBUG_OBJECT (pad, "Sending segment done at end of segment");
if (self->segment.rate >= 0) {
m = gst_message_new_segment_done (GST_OBJECT_CAST (self),
GST_FORMAT_TIME, self->segment.stop);
e = gst_event_new_segment_done (GST_FORMAT_TIME, self->segment.stop);
} else {
m = gst_message_new_segment_done (GST_OBJECT_CAST (self),
GST_FORMAT_TIME, self->segment.start);
e = gst_event_new_segment_done (GST_FORMAT_TIME, self->segment.start);
}
gst_element_post_message (GST_ELEMENT_CAST (self), m);
gst_pad_push_event (self->srcpad, e);
} else {
GST_DEBUG_OBJECT (pad, "Sending EOS at end of segment");
gst_pad_push_event (self->srcpad, gst_event_new_eos ());
}
} else if (flow_ret == GST_FLOW_NOT_LINKED || flow_ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
("Internal data stream error."),
("stream stopped, reason %s", reason));
gst_pad_push_event (self->srcpad, gst_event_new_eos ());
}
return;
}
}
static FLAC__StreamDecoderWriteStatus
gst_flac_dec_write (GstFlacDec * flacdec, const FLAC__Frame * frame,
const FLAC__int32 * const buffer[])
{
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
guint depth = frame->header.bits_per_sample;
guint width, gdepth;
guint sample_rate = frame->header.sample_rate;
guint channels = frame->header.channels;
guint samples = frame->header.blocksize;
guint j, i;
GstMapInfo map;
gboolean caps_changed;
GST_LOG_OBJECT (flacdec, "samples in frame header: %d", samples);
if (depth == 0) {
if (flacdec->depth < 4 || flacdec->depth > 32) {
GST_ERROR_OBJECT (flacdec, "unsupported depth %d from STREAMINFO",
flacdec->depth);
ret = GST_FLOW_ERROR;
goto done;
}
depth = flacdec->depth;
}
switch (depth) {
case 8:
gdepth = width = 8;
break;
case 12:
case 16:
gdepth = width = 16;
break;
case 20:
case 24:
gdepth = 24;
width = 32;
break;
case 32:
gdepth = width = 32;
break;
default:
GST_ERROR_OBJECT (flacdec, "unsupported depth %d", depth);
ret = GST_FLOW_ERROR;
goto done;
}
if (sample_rate == 0) {
if (flacdec->info.rate != 0) {
sample_rate = flacdec->info.rate;
} else {
GST_ERROR_OBJECT (flacdec, "unknown sample rate");
ret = GST_FLOW_ERROR;
goto done;
}
}
caps_changed = (sample_rate != GST_AUDIO_INFO_RATE (&flacdec->info))
|| (width != GST_AUDIO_INFO_WIDTH (&flacdec->info))
|| (gdepth != GST_AUDIO_INFO_DEPTH (&flacdec->info))
|| (channels != GST_AUDIO_INFO_CHANNELS (&flacdec->info));
if (caps_changed
|| !gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (flacdec))) {
GST_DEBUG_OBJECT (flacdec, "Negotiating %d Hz @ %d channels", sample_rate,
channels);
gst_audio_info_set_format (&flacdec->info,
gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth),
sample_rate, channels, NULL);
memcpy (flacdec->info.position,
channel_positions[flacdec->info.channels - 1],
sizeof (GstAudioChannelPosition) * flacdec->info.channels);
gst_audio_channel_positions_to_valid_order (flacdec->info.position,
flacdec->info.channels);
/* Note: we create the inverse reordering map here */
gst_audio_get_channel_reorder_map (flacdec->info.channels,
flacdec->info.position, channel_positions[flacdec->info.channels - 1],
flacdec->channel_reorder_map);
flacdec->depth = depth;
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (flacdec),
&flacdec->info);
}
outbuf =
gst_buffer_new_allocate (NULL, samples * channels * (width / 8), NULL);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
if (width == 8) {
gint8 *outbuffer = (gint8 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint8) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint8) buffer[reorder_map[j]][i];
}
}
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint16) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint16) buffer[reorder_map[j]][i];
}
}
}
} else if (width == 32) {
gint32 *outbuffer = (gint32 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint32) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint32) buffer[reorder_map[j]][i];
}
}
}
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (outbuf, &map);
GST_DEBUG_OBJECT (flacdec, "pushing %d samples", samples);
if (flacdec->error_count)
flacdec->error_count--;
ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (flacdec), outbuf, 1);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (flacdec, "finish_frame flow %s", gst_flow_get_name (ret));
}
done:
/* we act on the flow return value later in the handle_frame function, as we
* don't want to mess up the internal decoder state by returning ABORT when
* the error is in fact non-fatal (like a pad in flushing mode) and we want
* to continue later. So just pretend everything's dandy and act later. */
flacdec->last_flow = ret;
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
static GstFlowReturn
gst_irtsp_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
{
GstIRTSPParse *IRTSPParse = GST_IRTSP_PARSE (parse);
GstBuffer *buf = frame->buffer;
GstByteReader reader;
gint off;
GstMapInfo map;
gboolean ret = FALSE;
guint framesize;
gst_buffer_map (buf, &map, GST_MAP_READ);
if (G_UNLIKELY (map.size < 4))
goto exit;
gst_byte_reader_init (&reader, map.data, map.size);
off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffff0000,
0x24000000 + (IRTSPParse->channel_id << 16), 0, map.size);
GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
/* didn't find anything that looks like a sync word, skip */
if (off < 0) {
*skipsize = map.size - 3;
goto exit;
}
/* possible frame header, but not at offset 0? skip bytes before sync */
if (off > 0) {
*skipsize = off;
goto exit;
}
framesize = GST_READ_UINT16_BE (map.data + 2) + 4;
GST_LOG_OBJECT (parse, "got frame size %d", framesize);
ret = TRUE;
if (!gst_pad_has_current_caps (GST_BASE_PARSE_SRC_PAD (parse))) {
GstCaps *caps;
caps = gst_caps_new_empty_simple ("application/x-rtp");
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
gst_caps_unref (caps);
}
exit:
gst_buffer_unmap (buf, &map);
if (ret && framesize <= map.size) {
/* HACK HACK skip header.
* could also ask baseparse to skip this,
* but that would give us a discontinuity for free
* which is a bit too much to have on all our packets */
frame->out_buffer = gst_buffer_copy (frame->buffer);
gst_buffer_resize (frame->out_buffer, 4, -1);
GST_BUFFER_FLAG_UNSET (frame->out_buffer, GST_BUFFER_FLAG_DISCONT);
return gst_base_parse_finish_frame (parse, frame, framesize);
}
return GST_FLOW_OK;
}
static GstFlowReturn
gst_asteriskh263_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstAsteriskh263 *asteriskh263;
GstBuffer *outbuf;
GstFlowReturn ret;
asteriskh263 = GST_ASTERISK_H263 (parent);
if (!gst_rtp_buffer_validate (buf))
goto bad_packet;
{
gint payload_len;
guint8 *payload;
gboolean M;
guint32 timestamp;
guint32 samples;
guint16 asterisk_len;
GstRTPBuffer rtp = { NULL };
GstMapInfo map;
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
payload_len = gst_rtp_buffer_get_payload_len (&rtp);
payload = gst_rtp_buffer_get_payload (&rtp);
M = gst_rtp_buffer_get_marker (&rtp);
timestamp = gst_rtp_buffer_get_timestamp (&rtp);
gst_rtp_buffer_unmap (&rtp);
outbuf = gst_buffer_new_and_alloc (payload_len +
GST_ASTERISKH263_HEADER_LEN);
/* build the asterisk header */
asterisk_len = payload_len;
if (M)
asterisk_len |= 0x8000;
if (!asteriskh263->lastts)
asteriskh263->lastts = timestamp;
samples = timestamp - asteriskh263->lastts;
asteriskh263->lastts = timestamp;
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
GST_ASTERISKH263_HEADER_TIMESTAMP (map.data) = g_htonl (samples);
GST_ASTERISKH263_HEADER_LENGTH (map.data) = g_htons (asterisk_len);
/* copy the data into place */
memcpy (map.data + GST_ASTERISKH263_HEADER_LEN, payload, payload_len);
gst_buffer_unmap (outbuf, &map);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
if (!gst_pad_has_current_caps (asteriskh263->srcpad)) {
GstCaps *caps;
caps = gst_pad_get_pad_template_caps (asteriskh263->srcpad);
gst_pad_set_caps (asteriskh263->srcpad, caps);
gst_caps_unref (caps);
}
ret = gst_pad_push (asteriskh263->srcpad, outbuf);
gst_buffer_unref (buf);
}
return ret;
bad_packet:
{
GST_DEBUG ("Packet does not validate");
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static void
gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
{
GstOMXAudioEncClass *klass;
GstOMXPort *port = self->enc_out_port;
GstOMXBuffer *buf = NULL;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstOMXAcquireBufferReturn acq_return;
OMX_ERRORTYPE err;
klass = GST_OMX_AUDIO_ENC_GET_CLASS (self);
acq_return = gst_omx_port_acquire_buffer (port, &buf);
if (acq_return == GST_OMX_ACQUIRE_BUFFER_ERROR) {
goto component_error;
} else if (acq_return == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
goto flushing;
} else if (acq_return == GST_OMX_ACQUIRE_BUFFER_EOS) {
goto eos;
}
if (!gst_pad_has_current_caps (GST_AUDIO_ENCODER_SRC_PAD (self))
|| acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
GstAudioInfo *info =
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self));
GstCaps *caps;
GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps");
/* Reallocate all buffers */
if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
err = gst_omx_port_set_enabled (port, FALSE);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_deallocate_buffers (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
}
GST_AUDIO_ENCODER_STREAM_LOCK (self);
caps = klass->get_caps (self, self->enc_out_port, info);
if (!caps) {
if (buf)
gst_omx_port_release_buffer (self->enc_out_port, buf);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed;
}
GST_DEBUG_OBJECT (self, "Setting output caps: %" GST_PTR_FORMAT, caps);
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
gst_caps_unref (caps);
if (buf)
gst_omx_port_release_buffer (self->enc_out_port, buf);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed;
}
gst_caps_unref (caps);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
err = gst_omx_port_set_enabled (port, TRUE);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_allocate_buffers (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_populate (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_mark_reconfigured (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
}
/* Now get a buffer */
if (acq_return != GST_OMX_ACQUIRE_BUFFER_OK) {
return;
}
}
g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK);
if (!buf) {
g_assert ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER));
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto eos;
}
GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %" G_GUINT64_FORMAT,
(guint) buf->omx_buf->nFlags, (guint64) buf->omx_buf->nTimeStamp);
/* This prevents a deadlock between the srcpad stream
* lock and the videocodec stream lock, if ::reset()
* is called at the wrong time
*/
if (gst_omx_port_is_flushing (self->enc_out_port)) {
GST_DEBUG_OBJECT (self, "Flushing");
gst_omx_port_release_buffer (self->enc_out_port, buf);
goto flushing;
}
GST_AUDIO_ENCODER_STREAM_LOCK (self);
if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG)
&& buf->omx_buf->nFilledLen > 0) {
GstCaps *caps;
GstBuffer *codec_data;
GstMapInfo map = GST_MAP_INFO_INIT;
GST_DEBUG_OBJECT (self, "Handling codec data");
caps =
gst_caps_copy (gst_pad_get_current_caps (GST_AUDIO_ENCODER_SRC_PAD
(self)));
codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
memcpy (map.data,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
gst_buffer_unmap (codec_data, &map);
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
gst_caps_unref (caps);
if (buf)
gst_omx_port_release_buffer (self->enc_out_port, buf);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed;
}
gst_caps_unref (caps);
flow_ret = GST_FLOW_OK;
} else if (buf->omx_buf->nFilledLen > 0) {
GstBuffer *outbuf;
guint n_samples;
GST_DEBUG_OBJECT (self, "Handling output data");
n_samples =
klass->get_num_samples (self, self->enc_out_port,
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);
if (buf->omx_buf->nFilledLen > 0) {
GstMapInfo map = GST_MAP_INFO_INIT;
outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
memcpy (map.data,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
gst_buffer_unmap (outbuf, &map);
} else {
outbuf = gst_buffer_new ();
}
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
OMX_TICKS_PER_SECOND);
if (buf->omx_buf->nTickCount != 0)
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
OMX_TICKS_PER_SECOND);
flow_ret =
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self),
outbuf, n_samples);
}
GST_DEBUG_OBJECT (self, "Handled output data");
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
err = gst_omx_port_release_buffer (port, buf);
if (err != OMX_ErrorNone)
goto release_error;
self->downstream_flow_ret = flow_ret;
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return;
component_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("OpenMAX component in error state %s (0x%08x)",
gst_omx_component_get_last_error_string (self->enc),
gst_omx_component_get_last_error (self->enc)));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
return;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
return;
}
eos:
{
g_mutex_lock (&self->drain_lock);
if (self->draining) {
GST_DEBUG_OBJECT (self, "Drained");
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
flow_ret = GST_FLOW_OK;
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
} else {
GST_DEBUG_OBJECT (self, "Component signalled EOS");
flow_ret = GST_FLOW_EOS;
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_ENCODER_STREAM_LOCK (self);
self->downstream_flow_ret = flow_ret;
/* Here we fallback and pause the task for the EOS case */
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return;
}
flow_error:
{
if (flow_ret == GST_FLOW_EOS) {
GST_DEBUG_OBJECT (self, "EOS");
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
} else if (flow_ret == GST_FLOW_NOT_LINKED || flow_ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (self, STREAM, FAILED, ("Internal data stream error."),
("stream stopped, reason %s", gst_flow_get_name (flow_ret)));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
}
self->started = FALSE;
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return;
}
reconfigure_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Unable to reconfigure output port"));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE;
return;
}
caps_failed:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps"));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE;
return;
}
release_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to relase output buffer to component: %s (0x%08x)",
gst_omx_error_to_string (err), err));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return;
}
}
static GstCaps *
gst_shape_wipe_mask_sink_getcaps (GstPad * pad, GstCaps * filter)
{
GstShapeWipe *self = GST_SHAPE_WIPE (gst_pad_get_parent (pad));
GstCaps *ret, *tmp;
guint i, n;
if (gst_pad_has_current_caps (pad))
return gst_pad_get_current_caps (pad);
tmp = gst_pad_peer_query_caps (self->video_sinkpad, NULL);
if (tmp) {
ret =
gst_caps_intersect (tmp,
gst_pad_get_pad_template_caps (self->video_sinkpad));
gst_caps_unref (tmp);
} else {
ret = gst_pad_get_pad_template_caps (self->video_sinkpad);
}
GST_LOG_OBJECT (pad, "video sink accepted caps: %" GST_PTR_FORMAT, ret);
if (gst_caps_is_empty (ret))
goto done;
tmp = gst_pad_peer_query_caps (self->srcpad, NULL);
GST_LOG_OBJECT (pad, "srcpad accepted caps: %" GST_PTR_FORMAT, ret);
if (tmp) {
GstCaps *intersection;
intersection = gst_caps_intersect (ret, tmp);
gst_caps_unref (ret);
gst_caps_unref (tmp);
ret = intersection;
}
GST_LOG_OBJECT (pad, "intersection: %" GST_PTR_FORMAT, ret);
if (gst_caps_is_empty (ret))
goto done;
n = gst_caps_get_size (ret);
tmp = gst_caps_new_empty ();
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (ret, i);
GstStructure *t;
gst_structure_set_name (s, "video/x-raw");
gst_structure_remove_fields (s, "format", "framerate", NULL);
if (self->vinfo.width && self->vinfo.height)
gst_structure_set (s, "width", G_TYPE_INT, self->vinfo.width, "height",
G_TYPE_INT, self->vinfo.height, NULL);
gst_structure_set (s, "framerate", GST_TYPE_FRACTION, 0, 1, NULL);
t = gst_structure_copy (s);
gst_structure_set (s, "format", G_TYPE_STRING, GST_VIDEO_NE (GRAY16), NULL);
gst_structure_set (t, "format", G_TYPE_STRING, "GRAY8", NULL);
gst_caps_append_structure (tmp, t);
}
gst_caps_append (ret, tmp);
tmp = gst_pad_peer_query_caps (pad, NULL);
GST_LOG_OBJECT (pad, "peer accepted caps: %" GST_PTR_FORMAT, tmp);
if (tmp) {
GstCaps *intersection;
intersection = gst_caps_intersect (tmp, ret);
gst_caps_unref (tmp);
gst_caps_unref (ret);
ret = intersection;
}
done:
gst_object_unref (self);
GST_LOG_OBJECT (pad, "Returning caps: %" GST_PTR_FORMAT, ret);
return ret;
}
/* Ouput buffer preparation ... if the buffer has no caps, and our allowed
* output caps is fixed, then send the caps downstream, making sure caps are
* sent before segment event.
*
* This ensures that caps event is sent if we can, so that pipelines like:
* gst-launch filesrc location=rawsamples.raw !
* audio/x-raw,format=S16LE,rate=48000,channels=2 ! alsasink
* will work.
*/
static GstFlowReturn
gst_capsfilter_prepare_buf (GstBaseTransform * trans, GstBuffer * input,
GstBuffer ** buf)
{
GstFlowReturn ret = GST_FLOW_OK;
/* always return the input as output buffer */
*buf = input;
if (!gst_pad_has_current_caps (trans->sinkpad)) {
/* No caps. See if the output pad only supports fixed caps */
GstCapsFilter *filter = GST_CAPSFILTER (trans);
GstCaps *out_caps;
GList *pending_events = filter->pending_events;
GST_LOG_OBJECT (trans, "Input pad does not have caps");
filter->pending_events = NULL;
out_caps = gst_pad_get_current_caps (trans->srcpad);
if (out_caps == NULL) {
out_caps = gst_pad_get_allowed_caps (trans->srcpad);
g_return_val_if_fail (out_caps != NULL, GST_FLOW_ERROR);
}
out_caps = gst_caps_simplify (out_caps);
if (gst_caps_is_fixed (out_caps) && !gst_caps_is_empty (out_caps)) {
GST_DEBUG_OBJECT (trans, "Have fixed output caps %"
GST_PTR_FORMAT " to apply to srcpad", out_caps);
if (!gst_pad_has_current_caps (trans->srcpad)) {
if (gst_pad_set_caps (trans->srcpad, out_caps)) {
if (pending_events) {
GList *l;
for (l = g_list_last (pending_events); l; l = l->prev) {
GST_LOG_OBJECT (trans, "Forwarding %s event",
GST_EVENT_TYPE_NAME (l->data));
GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (trans,
l->data);
}
g_list_free (pending_events);
pending_events = NULL;
}
} else {
ret = GST_FLOW_NOT_NEGOTIATED;
}
}
g_list_free_full (pending_events, (GDestroyNotify) gst_event_unref);
gst_caps_unref (out_caps);
} else {
gchar *caps_str = gst_caps_to_string (out_caps);
GST_DEBUG_OBJECT (trans, "Cannot choose caps. Have unfixed output caps %"
GST_PTR_FORMAT, out_caps);
gst_caps_unref (out_caps);
GST_ELEMENT_ERROR (trans, STREAM, FORMAT,
("Filter caps do not completely specify the output format"),
("Output caps are unfixed: %s", caps_str));
g_free (caps_str);
g_list_free_full (pending_events, (GDestroyNotify) gst_event_unref);
ret = GST_FLOW_ERROR;
}
}
return ret;
}
static GstCaps *
gst_shape_wipe_src_getcaps (GstPad * pad, GstCaps * filter)
{
GstShapeWipe *self = GST_SHAPE_WIPE (gst_pad_get_parent (pad));
GstCaps *templ, *ret, *tmp;
if (gst_pad_has_current_caps (pad))
return gst_pad_get_current_caps (pad);
else if (gst_pad_has_current_caps (self->video_sinkpad))
return gst_pad_get_current_caps (self->video_sinkpad);
templ = gst_pad_get_pad_template_caps (self->video_sinkpad);
tmp = gst_pad_peer_query_caps (self->video_sinkpad, NULL);
if (tmp) {
ret = gst_caps_intersect (tmp, templ);
gst_caps_unref (templ);
gst_caps_unref (tmp);
} else {
ret = templ;
}
GST_LOG_OBJECT (pad, "video sink accepted caps: %" GST_PTR_FORMAT, ret);
if (gst_caps_is_empty (ret))
goto done;
tmp = gst_pad_peer_query_caps (pad, NULL);
GST_LOG_OBJECT (pad, "peer accepted caps: %" GST_PTR_FORMAT, ret);
if (tmp) {
GstCaps *intersection;
intersection = gst_caps_intersect (tmp, ret);
gst_caps_unref (tmp);
gst_caps_unref (ret);
ret = intersection;
}
GST_LOG_OBJECT (pad, "intersection: %" GST_PTR_FORMAT, ret);
if (gst_caps_is_empty (ret))
goto done;
if (self->vinfo.height && self->vinfo.width) {
guint i, n;
ret = gst_caps_make_writable (ret);
n = gst_caps_get_size (ret);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (ret, i);
gst_structure_set (s, "width", G_TYPE_INT, self->vinfo.width, "height",
G_TYPE_INT, self->vinfo.height, NULL);
}
}
tmp = gst_pad_peer_query_caps (self->mask_sinkpad, NULL);
GST_LOG_OBJECT (pad, "mask sink accepted caps: %" GST_PTR_FORMAT, ret);
if (tmp) {
GstCaps *intersection, *tmp2;
guint i, n;
tmp2 = gst_pad_get_pad_template_caps (self->mask_sinkpad);
intersection = gst_caps_intersect (tmp, tmp2);
gst_caps_unref (tmp);
gst_caps_unref (tmp2);
tmp = gst_caps_make_writable (intersection);
n = gst_caps_get_size (tmp);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (tmp, i);
gst_structure_remove_fields (s, "format", "framerate", NULL);
gst_structure_set_name (s, "video/x-raw");
}
intersection = gst_caps_intersect (tmp, ret);
gst_caps_unref (tmp);
gst_caps_unref (ret);
ret = intersection;
}
done:
gst_object_unref (self);
GST_LOG_OBJECT (pad, "Returning caps: %" GST_PTR_FORMAT, ret);
return ret;
}
static gboolean
gst_gdk_pixbuf_dec_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstFlowReturn res = GST_FLOW_OK;
gboolean ret = TRUE, forward = TRUE;
GstGdkPixbufDec *pixbuf;
pixbuf = GST_GDK_PIXBUF_DEC (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
ret = gst_gdk_pixbuf_dec_sink_setcaps (pixbuf, caps);
forward = FALSE;
break;
}
case GST_EVENT_EOS:
if (pixbuf->pixbuf_loader != NULL) {
gdk_pixbuf_loader_close (pixbuf->pixbuf_loader, NULL);
res = gst_gdk_pixbuf_dec_flush (pixbuf);
g_object_unref (G_OBJECT (pixbuf->pixbuf_loader));
pixbuf->pixbuf_loader = NULL;
/* as long as we don't have flow returns for event functions we need
* to post an error here, or the application might never know that
* things failed */
if (res != GST_FLOW_OK && res != GST_FLOW_FLUSHING
&& res != GST_FLOW_EOS && res != GST_FLOW_NOT_LINKED) {
GST_ELEMENT_ERROR (pixbuf, STREAM, FAILED, (NULL), ("Flow: %s",
gst_flow_get_name (res)));
forward = FALSE;
ret = FALSE;
}
}
break;
case GST_EVENT_FLUSH_STOP:
g_list_free_full (pixbuf->pending_events,
(GDestroyNotify) gst_event_unref);
pixbuf->pending_events = NULL;
/* Fall through */
case GST_EVENT_SEGMENT:
{
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
if (segment->format == GST_FORMAT_BYTES)
pixbuf->packetized = FALSE;
else
pixbuf->packetized = TRUE;
if (pixbuf->pixbuf_loader != NULL) {
gdk_pixbuf_loader_close (pixbuf->pixbuf_loader, NULL);
g_object_unref (G_OBJECT (pixbuf->pixbuf_loader));
pixbuf->pixbuf_loader = NULL;
}
break;
}
default:
break;
}
if (forward) {
if (!gst_pad_has_current_caps (pixbuf->srcpad) &&
GST_EVENT_IS_SERIALIZED (event)
&& GST_EVENT_TYPE (event) > GST_EVENT_CAPS
&& GST_EVENT_TYPE (event) != GST_EVENT_FLUSH_STOP
&& GST_EVENT_TYPE (event) != GST_EVENT_EOS) {
ret = TRUE;
pixbuf->pending_events = g_list_prepend (pixbuf->pending_events, event);
} else {
ret = gst_pad_event_default (pad, parent, event);
}
} else {
gst_event_unref (event);
}
return ret;
}
