static gboolean gst_imx_audio_uniaudio_dec_set_format(GstAudioDecoder *dec, GstCaps *caps)
{
UniACodecParameter parameter;
UniACodecMemoryOps memory_ops;
GstImxAudioUniaudioDec *imx_audio_uniaudio_dec = GST_IMX_AUDIO_UNIAUDIO_DEC(dec);
#define UNIA_SET_PARAMETER(PARAM_ID, DESC) \
do \
{ \
if (imx_audio_uniaudio_dec->codec->set_parameter(imx_audio_uniaudio_dec->handle, (PARAM_ID), ¶meter) != ACODEC_SUCCESS) \
{ \
GST_ERROR_OBJECT(dec, "setting %s parameter failed: %s", (DESC), imx_audio_uniaudio_dec->codec->get_last_error(imx_audio_uniaudio_dec->handle)); \
gst_imx_audio_uniaudio_dec_close_handle(imx_audio_uniaudio_dec); \
return FALSE; \
} \
} \
while (0)
#define UNIA_SET_PARAMETER_EX(PARAM_ID, DESC, VALUE) \
do \
{ \
if (imx_audio_uniaudio_dec->codec->set_parameter(imx_audio_uniaudio_dec->handle, (PARAM_ID), ((UniACodecParameter *)(VALUE))) != ACODEC_SUCCESS) \
{ \
GST_ERROR_OBJECT(dec, "setting %s parameter failed: %s", (DESC), imx_audio_uniaudio_dec->codec->get_last_error(imx_audio_uniaudio_dec->handle)); \
gst_imx_audio_uniaudio_dec_close_handle(imx_audio_uniaudio_dec); \
return FALSE; \
} \
} \
while (0)
if (imx_audio_uniaudio_dec->handle != NULL)
{
/* drain old decoder handle */
gst_imx_audio_uniaudio_dec_handle_frame(dec, NULL);
gst_imx_audio_uniaudio_dec_close_handle(imx_audio_uniaudio_dec);
}
if ((imx_audio_uniaudio_dec->codec = gst_imx_audio_uniaudio_codec_table_get_codec(caps)) == NULL)
{
GST_ERROR_OBJECT(dec, "found no suitable codec for caps %" GST_PTR_FORMAT, (gpointer)caps);
return FALSE;
}
memory_ops.Calloc = gst_imx_audio_uniaudio_dec_calloc;
memory_ops.Malloc = gst_imx_audio_uniaudio_dec_malloc;
memory_ops.Free = gst_imx_audio_uniaudio_dec_free;
memory_ops.ReAlloc = gst_imx_audio_uniaudio_dec_realloc;
if ((imx_audio_uniaudio_dec->handle = imx_audio_uniaudio_dec->codec->create_codec(&memory_ops)) == NULL)
{
GST_ERROR_OBJECT(dec, "creating codec handle for caps %" GST_PTR_FORMAT " failed", (gpointer)caps);
return FALSE;
}
/* Get configuration parameters from caps */
{
int samplerate, channels, bitrate, block_align, wmaversion;
gchar const *stream_format, *sample_format;
GValue const *value;
gboolean framed, is_vorbis;
GstBuffer *codec_data = NULL;
GstStructure *structure = gst_caps_get_structure(caps, 0);
imx_audio_uniaudio_dec->skip_header_counter = 0;
is_vorbis = (g_strcmp0(gst_structure_get_name(structure), "audio/x-vorbis") == 0);
parameter.framed = is_vorbis || (gst_structure_get_boolean(structure, "framed", &framed) && framed) || (gst_structure_get_boolean(structure, "parsed", &framed) && framed);
GST_DEBUG_OBJECT(dec, "input is framed: %d", parameter.framed);
UNIA_SET_PARAMETER(UNIA_FRAMED, "framed");
if (gst_structure_get_int(structure, "rate", &samplerate))
{
GST_DEBUG_OBJECT(dec, "input caps sample rate: %d Hz", samplerate);
parameter.samplerate = samplerate;
UNIA_SET_PARAMETER(UNIA_SAMPLERATE, "sample rate");
}
if (gst_structure_get_int(structure, "channels", &channels))
{
CHAN_TABLE table;
GST_DEBUG_OBJECT(dec, "input caps channel count: %d", channels);
parameter.channels = channels;
UNIA_SET_PARAMETER(UNIA_CHANNEL, "channel");
memset(&table, 0, sizeof(table));
table.size = CHANNEL_MAPS_SIZE;
memcpy(&table.channel_table, uniaudio_channel_maps, sizeof(uniaudio_channel_maps));
UNIA_SET_PARAMETER_EX(UNIA_CHAN_MAP_TABLE, "channel map", &table);
}
if (gst_structure_get_int(structure, "bitrate", &bitrate))
{
GST_DEBUG_OBJECT(dec, "input caps channel count: %d", bitrate);
parameter.bitrate = bitrate;
UNIA_SET_PARAMETER(UNIA_BITRATE, "bitrate");
}
if (gst_structure_get_int(structure, "block_align", &block_align))
{
GST_DEBUG_OBJECT(dec, "block alignment: %d", block_align);
parameter.blockalign = block_align;
UNIA_SET_PARAMETER(UNIA_WMA_BlOCKALIGN, "blockalign");
}
if (gst_structure_get_int(structure, "wmaversion", &wmaversion))
{
GST_DEBUG_OBJECT(dec, "WMA version: %d", wmaversion);
parameter.version = wmaversion;
UNIA_SET_PARAMETER(UNIA_WMA_VERSION, "wmaversion");
}
if ((stream_format = gst_structure_get_string(structure, "stream-format")) != NULL)
{
GST_DEBUG_OBJECT(dec, "input caps stream format: %s", stream_format);
if (g_strcmp0(stream_format, "raw") == 0)
parameter.stream_type = STREAM_ADTS;
if (g_strcmp0(stream_format, "adif") == 0)
parameter.stream_type = STREAM_ADIF;
if (g_strcmp0(stream_format, "raw") == 0)
parameter.stream_type = STREAM_RAW;
else
parameter.stream_type = STREAM_UNKNOW;
UNIA_SET_PARAMETER(UNIA_STREAM_TYPE, "stream type");
}
if ((sample_format = gst_structure_get_string(structure, "format")) != NULL)
{
GstAudioFormat fmt;
GstAudioFormatInfo const * fmtinfo;
GST_DEBUG_OBJECT(dec, "input caps stream sample format: %s", sample_format);
if ((fmt = gst_audio_format_from_string(sample_format)) == GST_AUDIO_FORMAT_UNKNOWN)
{
GST_ERROR_OBJECT(dec, "format is unknown, cannot continue");
return FALSE;
}
fmtinfo = gst_audio_format_get_info(fmt);
g_assert(fmtinfo != NULL);
parameter.depth = GST_AUDIO_FORMAT_INFO_DEPTH(fmtinfo);
UNIA_SET_PARAMETER(UNIA_DEPTH, "depth");
}
/* Handle codec data, either directly from a codec_data caps,
* or assemble it from a list of buffers specified by the
* streamheader caps (typically used by Vorbis audio) */
/* Cleanup old codec data first */
if (imx_audio_uniaudio_dec->codec_data != NULL)
{
gst_buffer_unref(imx_audio_uniaudio_dec->codec_data);
imx_audio_uniaudio_dec->codec_data = NULL;
}
/* Check if either codec_data or streamheader caps exist */
if ((value = gst_structure_get_value(structure, "codec_data")) != NULL)
{
/* codec_data caps exist - simply make a copy of its buffer
* (this makes sure we own that buffer properly) */
GstBuffer *caps_buffer;
GST_DEBUG_OBJECT(dec, "reading codec_data value");
caps_buffer = gst_value_get_buffer(value);
g_assert(caps_buffer != NULL);
codec_data = gst_buffer_copy(caps_buffer);
}
else if ((value = gst_structure_get_value(structure, "streamheader")) != NULL)
{
/* streamheader caps exist, which are a list of buffers
* these buffers need to be concatenated and then given as
* one consecutive codec data buffer to the decoder */
guint i, num_buffers = gst_value_array_get_size(value);
GstAdapter *streamheader_adapter = gst_adapter_new();
GST_DEBUG_OBJECT(dec, "reading streamheader value (%u headers)", num_buffers);
imx_audio_uniaudio_dec->num_vorbis_headers = num_buffers;
/* Use the GstAdapter to stitch these buffers together */
for (i = 0; i < num_buffers; ++i)
{
GValue const *array_value = gst_value_array_get_value(value, i);
GstBuffer *buf = gst_value_get_buffer(array_value);
GST_DEBUG_OBJECT(dec, "add streamheader buffer #%u with %" G_GSIZE_FORMAT " byte", i, gst_buffer_get_size(buf));
gst_adapter_push(streamheader_adapter, gst_buffer_copy(buf));
}
codec_data = gst_adapter_take_buffer(streamheader_adapter, gst_adapter_available(streamheader_adapter));
g_object_unref(G_OBJECT(streamheader_adapter));
}
/* At this point, if either codec_data or streamheader caps were found,
* the codec_data pointer will refer to a valid non-empty buffer with
* codec data inside. This buffer is owned by this audio decoder object,
* and must be kept around for as long as the decoder needs to be ran,
* since the set_parameter call below does *not* copy the codec data
* bytes into some internal buffer. Instead, the uniaudio decoder plugin
* expects the caller to keep the buffer valid. */
if ((codec_data != NULL) && (gst_buffer_get_size(codec_data) != 0))
{
GstMapInfo map;
gst_buffer_map(codec_data, &map, GST_MAP_READ);
parameter.codecData.size = map.size;
parameter.codecData.buf = (char *)(map.data);
UNIA_SET_PARAMETER(UNIA_CODEC_DATA, "codec data");
gst_buffer_unmap(codec_data, &map);
imx_audio_uniaudio_dec->codec_data = codec_data;
GST_DEBUG_OBJECT(dec, "codec data: %" G_GUINT32_FORMAT " byte", (guint32)(parameter.codecData.size));
}
}
GST_DEBUG_OBJECT(dec, "decoder configured");
imx_audio_uniaudio_dec->has_audioinfo_set = FALSE;
#undef UNIA_SET_PARAMETER
return TRUE;
}
static GstFlowReturn
gst_ac3_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
{
GstAc3Parse *ac3parse = GST_AC3_PARSE (parse);
GstBuffer *buf = frame->buffer;
GstByteReader reader;
gint off;
gboolean lost_sync, draining, eac, more = FALSE;
guint frmsiz, blocks, sid;
guint rate, chans;
gboolean update_rate = FALSE;
gint framesize = 0;
gint have_blocks = 0;
GstMapInfo map;
gboolean ret = FALSE;
GstFlowReturn res = GST_FLOW_OK;
gst_buffer_map (buf, &map, GST_MAP_READ);
if (G_UNLIKELY (map.size < 6)) {
*skipsize = 1;
goto cleanup;
}
gst_byte_reader_init (&reader, map.data, map.size);
off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffff0000, 0x0b770000,
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 cleanup;
}
/* possible frame header, but not at offset 0? skip bytes before sync */
if (off > 0) {
*skipsize = off;
goto cleanup;
}
/* make sure the values in the frame header look sane */
if (!gst_ac3_parse_frame_header (ac3parse, buf, 0, &frmsiz, &rate, &chans,
&blocks, &sid, &eac)) {
*skipsize = off + 2;
goto cleanup;
}
GST_LOG_OBJECT (parse, "size: %u, blocks: %u, rate: %u, chans: %u", frmsiz,
blocks, rate, chans);
framesize = frmsiz;
if (G_UNLIKELY (g_atomic_int_get (&ac3parse->align) ==
GST_AC3_PARSE_ALIGN_NONE))
gst_ac3_parse_set_alignment (ac3parse, eac);
GST_LOG_OBJECT (parse, "got frame");
lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
draining = GST_BASE_PARSE_DRAINING (parse);
if (g_atomic_int_get (&ac3parse->align) == GST_AC3_PARSE_ALIGN_IEC61937) {
/* We need 6 audio blocks from each substream, so we keep going forwards
* till we have it */
g_assert (blocks > 0);
GST_LOG_OBJECT (ac3parse, "Need %d frames before pushing", 6 / blocks);
if (sid != 0) {
/* We need the first substream to be the one with id 0 */
GST_LOG_OBJECT (ac3parse, "Skipping till we find sid 0");
*skipsize = off + 2;
goto cleanup;
}
framesize = 0;
/* Loop till we have 6 blocks per substream */
for (have_blocks = 0; !more && have_blocks < 6; have_blocks += blocks) {
/* Loop till we get one frame from each substream */
do {
framesize += frmsiz;
if (!gst_byte_reader_skip (&reader, frmsiz)
|| map.size < (framesize + 6)) {
more = TRUE;
break;
}
if (!gst_ac3_parse_frame_header (ac3parse, buf, framesize, &frmsiz,
NULL, NULL, NULL, &sid, &eac)) {
*skipsize = off + 2;
goto cleanup;
}
} while (sid);
}
/* We're now at the next frame, so no need to skip if resyncing */
frmsiz = 0;
}
if (lost_sync && !draining) {
guint16 word = 0;
GST_DEBUG_OBJECT (ac3parse, "resyncing; checking next frame syncword");
if (more || !gst_byte_reader_skip (&reader, frmsiz) ||
!gst_byte_reader_get_uint16_be (&reader, &word)) {
GST_DEBUG_OBJECT (ac3parse, "... but not sufficient data");
gst_base_parse_set_min_frame_size (parse, framesize + 6);
*skipsize = 0;
goto cleanup;
} else {
if (word != 0x0b77) {
GST_DEBUG_OBJECT (ac3parse, "0x%x not OK", word);
*skipsize = off + 2;
goto cleanup;
} else {
/* ok, got sync now, let's assume constant frame size */
gst_base_parse_set_min_frame_size (parse, framesize);
}
}
}
/* expect to have found a frame here */
g_assert (framesize);
ret = TRUE;
/* arrange for metadata setup */
if (G_UNLIKELY (sid)) {
/* dependent frame, no need to (ac)count for or consider further */
GST_LOG_OBJECT (parse, "sid: %d", sid);
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_NO_FRAME;
/* TODO maybe also mark as DELTA_UNIT,
* if that does not surprise baseparse elsewhere */
/* occupies same time space as previous base frame */
if (G_LIKELY (GST_BUFFER_TIMESTAMP (buf) >= GST_BUFFER_DURATION (buf)))
GST_BUFFER_TIMESTAMP (buf) -= GST_BUFFER_DURATION (buf);
/* only shortcut if we already arranged for caps */
if (G_LIKELY (ac3parse->sample_rate > 0))
goto cleanup;
}
if (G_UNLIKELY (ac3parse->sample_rate != rate || ac3parse->channels != chans
|| ac3parse->eac != eac)) {
GstCaps *caps = gst_caps_new_simple (eac ? "audio/x-eac3" : "audio/x-ac3",
"framed", G_TYPE_BOOLEAN, TRUE, "rate", G_TYPE_INT, rate,
"channels", G_TYPE_INT, chans, NULL);
gst_caps_set_simple (caps, "alignment", G_TYPE_STRING,
g_atomic_int_get (&ac3parse->align) == GST_AC3_PARSE_ALIGN_IEC61937 ?
"iec61937" : "frame", NULL);
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
gst_caps_unref (caps);
ac3parse->sample_rate = rate;
ac3parse->channels = chans;
ac3parse->eac = eac;
update_rate = TRUE;
}
if (G_UNLIKELY (ac3parse->blocks != blocks)) {
ac3parse->blocks = blocks;
update_rate = TRUE;
}
if (G_UNLIKELY (update_rate))
gst_base_parse_set_frame_rate (parse, rate, 256 * blocks, 2, 2);
cleanup:
gst_buffer_unmap (buf, &map);
if (ret && framesize <= map.size) {
res = gst_base_parse_finish_frame (parse, frame, framesize);
}
return res;
}
/**
* gst_type_find_helper_get_range:
* @obj: A #GstObject that will be passed as first argument to @func
* @parent: the parent of @obj or NULL
* @func: (scope call): A generic #GstTypeFindHelperGetRangeFunction that will
* be used to access data at random offsets when doing the typefinding
* @size: The length in bytes
* @extension: extension of the media
* @prob: (out) (allow-none): location to store the probability of the found
* caps, or #NULL
*
* Utility function to do pull-based typefinding. Unlike gst_type_find_helper()
* however, this function will use the specified function @func to obtain the
* data needed by the typefind functions, rather than operating on a given
* source pad. This is useful mostly for elements like tag demuxers which
* strip off data at the beginning and/or end of a file and want to typefind
* the stripped data stream before adding their own source pad (the specified
* callback can then call the upstream peer pad with offsets adjusted for the
* tag size, for example).
*
* When @extension is not NULL, this function will first try the typefind
* functions for the given extension, which might speed up the typefinding
* in many cases.
*
* Free-function: gst_caps_unref
*
* Returns: (transfer full): the #GstCaps corresponding to the data stream.
* Returns #NULL if no #GstCaps matches the data stream.
*/
GstCaps *
gst_type_find_helper_get_range (GstObject * obj, GstObject * parent,
GstTypeFindHelperGetRangeFunction func, guint64 size,
const gchar * extension, GstTypeFindProbability * prob)
{
GstTypeFindHelper helper;
GstTypeFind find;
GSList *walk;
GList *l, *type_list;
GstCaps *result = NULL;
gint pos = 0;
g_return_val_if_fail (GST_IS_OBJECT (obj), NULL);
g_return_val_if_fail (func != NULL, NULL);
helper.buffers = NULL;
helper.size = size;
helper.last_offset = 0;
helper.func = func;
helper.best_probability = GST_TYPE_FIND_NONE;
helper.caps = NULL;
helper.obj = obj;
helper.parent = parent;
find.data = &helper;
find.peek = helper_find_peek;
find.suggest = helper_find_suggest;
if (size == 0 || size == (guint64) - 1) {
find.get_length = NULL;
} else {
find.get_length = helper_find_get_length;
}
type_list = gst_type_find_factory_get_list ();
/* move the typefinders for the extension first in the list. The idea is that
* when one of them returns MAX we don't need to search further as there is a
* very high chance we got the right type. */
if (extension) {
GList *next;
GST_LOG_OBJECT (obj, "sorting typefind for extension %s to head",
extension);
for (l = type_list; l; l = next) {
const gchar *const *ext;
GstTypeFindFactory *factory;
next = l->next;
factory = GST_TYPE_FIND_FACTORY (l->data);
ext = gst_type_find_factory_get_extensions (factory);
if (ext == NULL)
continue;
GST_LOG_OBJECT (obj, "testing factory %s for extension %s",
GST_OBJECT_NAME (factory), extension);
while (*ext != NULL) {
if (strcmp (*ext, extension) == 0) {
/* found extension, move in front */
GST_LOG_OBJECT (obj, "moving typefind for extension %s to head",
extension);
/* remove entry from list */
type_list = g_list_delete_link (type_list, l);
/* insert at the position */
type_list = g_list_insert (type_list, factory, pos);
/* next element will be inserted after this one */
pos++;
break;
}
++ext;
}
}
}
for (l = type_list; l; l = l->next) {
helper.factory = GST_TYPE_FIND_FACTORY (l->data);
gst_type_find_factory_call_function (helper.factory, &find);
if (helper.best_probability >= GST_TYPE_FIND_MAXIMUM)
break;
}
gst_plugin_feature_list_free (type_list);
for (walk = helper.buffers; walk; walk = walk->next) {
GstMappedBuffer *bmap = (GstMappedBuffer *) walk->data;
gst_buffer_unmap (bmap->buffer, &bmap->map);
gst_buffer_unref (bmap->buffer);
g_slice_free (GstMappedBuffer, bmap);
}
g_slist_free (helper.buffers);
if (helper.best_probability > 0)
result = helper.caps;
if (prob)
*prob = helper.best_probability;
GST_LOG_OBJECT (obj, "Returning %" GST_PTR_FORMAT " (probability = %u)",
result, (guint) helper.best_probability);
return result;
}
static gboolean
daala_dec_set_format (GstVideoDecoder * bdec, GstVideoCodecState * state)
{
GstDaalaDec *dec;
dec = GST_DAALA_DEC (bdec);
/* Keep a copy of the input state */
if (dec->input_state)
gst_video_codec_state_unref (dec->input_state);
dec->input_state = gst_video_codec_state_ref (state);
/* FIXME : Interesting, we always accept any kind of caps ? */
if (state->codec_data) {
GstBuffer *buffer;
GstMapInfo minfo;
guint8 *data;
guint size;
guint offset;
buffer = state->codec_data;
gst_buffer_map (buffer, &minfo, GST_MAP_READ);
offset = 0;
size = minfo.size;
data = (guint8 *) minfo.data;
while (size > 2) {
guint psize;
GstBuffer *buf;
psize = (data[0] << 8) | data[1];
/* skip header */
data += 2;
size -= 2;
offset += 2;
/* make sure we don't read too much */
psize = MIN (psize, size);
buf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, offset, psize);
/* first buffer is a discont buffer */
if (offset == 2)
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
/* now feed it to the decoder we can ignore the error */
daala_dec_decode_buffer (dec, buf, NULL);
gst_buffer_unref (buf);
/* skip the data */
size -= psize;
data += psize;
offset += psize;
}
gst_buffer_unmap (buffer, &minfo);
}
GST_DEBUG_OBJECT (dec, "Done");
return TRUE;
}
static gboolean
gst_ac3_parse_frame_header_ac3 (GstAc3Parse * ac3parse, GstBuffer * buf,
gint skip, guint * frame_size, guint * rate, guint * chans, guint * blks,
guint * sid)
{
GstBitReader bits;
GstMapInfo map;
guint8 fscod, frmsizcod, bsid, acmod, lfe_on, rate_scale;
gboolean ret = FALSE;
GST_LOG_OBJECT (ac3parse, "parsing ac3");
gst_buffer_map (buf, &map, GST_MAP_READ);
gst_bit_reader_init (&bits, map.data, map.size);
gst_bit_reader_skip_unchecked (&bits, skip * 8);
gst_bit_reader_skip_unchecked (&bits, 16 + 16);
fscod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2);
frmsizcod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 6);
if (G_UNLIKELY (fscod == 3 || frmsizcod >= G_N_ELEMENTS (frmsizcod_table))) {
GST_DEBUG_OBJECT (ac3parse, "bad fscod=%d frmsizcod=%d", fscod, frmsizcod);
goto cleanup;
}
bsid = gst_bit_reader_get_bits_uint8_unchecked (&bits, 5);
gst_bit_reader_skip_unchecked (&bits, 3); /* bsmod */
acmod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3);
/* spec not quite clear here: decoder should decode if less than 8,
* but seemingly only defines 6 and 8 cases */
/* Files with 9 and 10 happen, and seem to comply with the <= 8
format, so let them through. The spec says nothing about 9 and 10 */
if (bsid > 10) {
GST_DEBUG_OBJECT (ac3parse, "unexpected bsid=%d", bsid);
goto cleanup;
} else if (bsid != 8 && bsid != 6) {
GST_DEBUG_OBJECT (ac3parse, "undefined bsid=%d", bsid);
}
if ((acmod & 0x1) && (acmod != 0x1)) /* 3 front channels */
gst_bit_reader_skip_unchecked (&bits, 2);
if ((acmod & 0x4)) /* if a surround channel exists */
gst_bit_reader_skip_unchecked (&bits, 2);
if (acmod == 0x2) /* if in 2/0 mode */
gst_bit_reader_skip_unchecked (&bits, 2);
lfe_on = gst_bit_reader_get_bits_uint8_unchecked (&bits, 1);
/* 6/8->0, 9->1, 10->2,
see http://matroska.org/technical/specs/codecid/index.html */
rate_scale = (CLAMP (bsid, 8, 10) - 8);
if (frame_size)
*frame_size = frmsizcod_table[frmsizcod].frame_size[fscod] * 2;
if (rate)
*rate = fscod_rates[fscod] >> rate_scale;
if (chans)
*chans = acmod_chans[acmod] + lfe_on;
if (blks)
*blks = 6;
if (sid)
*sid = 0;
ret = TRUE;
cleanup:
gst_buffer_unmap (buf, &map);
return ret;
}
static GstFlowReturn
gst_dvbsrc_read_device (GstDvbSrc * object, int size, GstBuffer ** buffer)
{
gint count = 0;
gint ret_val = 0;
GstBuffer *buf = gst_buffer_new_and_alloc (size);
GstClockTime timeout = object->timeout * GST_USECOND;
GstMapInfo map;
g_return_val_if_fail (GST_IS_BUFFER (buf), GST_FLOW_ERROR);
if (object->fd_dvr < 0)
return GST_FLOW_ERROR;
gst_buffer_map (buf, &map, GST_MAP_WRITE);
while (count < size) {
ret_val = gst_poll_wait (object->poll, timeout);
GST_LOG_OBJECT (object, "select returned %d", ret_val);
if (G_UNLIKELY (ret_val < 0)) {
if (errno == EBUSY)
goto stopped;
else if (errno == EINTR)
continue;
else
goto select_error;
} else if (G_UNLIKELY (ret_val == 0)) {
/* timeout, post element message */
gst_element_post_message (GST_ELEMENT_CAST (object),
gst_message_new_element (GST_OBJECT (object),
gst_structure_new_empty ("dvb-read-failure")));
} else {
int nread = read (object->fd_dvr, map.data + count, size - count);
if (G_UNLIKELY (nread < 0)) {
GST_WARNING_OBJECT
(object,
"Unable to read from device: /dev/dvb/adapter%d/dvr%d (%d)",
object->adapter_number, object->frontend_number, errno);
gst_element_post_message (GST_ELEMENT_CAST (object),
gst_message_new_element (GST_OBJECT (object),
gst_structure_new_empty ("dvb-read-failure")));
} else
count = count + nread;
}
}
gst_buffer_unmap (buf, &map);
gst_buffer_resize (buf, 0, count);
*buffer = buf;
return GST_FLOW_OK;
stopped:
{
GST_DEBUG_OBJECT (object, "stop called");
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
return GST_FLOW_FLUSHING;
}
select_error:
{
GST_ELEMENT_ERROR (object, RESOURCE, READ, (NULL),
("select error %d: %s (%d)", ret_val, g_strerror (errno), errno));
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_rsvg_decode_image (GstRsvgDec * rsvg, GstBuffer * buffer,
GstVideoCodecFrame * frame)
{
GstVideoDecoder *decoder = GST_VIDEO_DECODER (rsvg);
GstFlowReturn ret = GST_FLOW_OK;
cairo_t *cr;
cairo_surface_t *surface;
RsvgHandle *handle;
GError *error = NULL;
RsvgDimensionData dimension;
gdouble scalex, scaley;
GstMapInfo minfo;
GstVideoFrame vframe;
GstVideoCodecState *output_state;
GST_LOG_OBJECT (rsvg, "parsing svg");
if (!gst_buffer_map (buffer, &minfo, GST_MAP_READ)) {
GST_ERROR_OBJECT (rsvg, "Failed to get SVG image");
return GST_FLOW_ERROR;
}
handle = rsvg_handle_new_from_data (minfo.data, minfo.size, &error);
if (!handle) {
GST_ERROR_OBJECT (rsvg, "Failed to parse SVG image: %s", error->message);
g_error_free (error);
return GST_FLOW_ERROR;
}
rsvg_handle_get_dimensions (handle, &dimension);
output_state = gst_video_decoder_get_output_state (decoder);
if ((output_state == NULL)
|| GST_VIDEO_INFO_WIDTH (&output_state->info) != dimension.width
|| GST_VIDEO_INFO_HEIGHT (&output_state->info) != dimension.height) {
/* Create the output state */
if (output_state)
gst_video_codec_state_unref (output_state);
output_state =
gst_video_decoder_set_output_state (decoder, GST_RSVG_VIDEO_FORMAT,
dimension.width, dimension.height, rsvg->input_state);
}
ret = gst_video_decoder_allocate_output_frame (decoder, frame);
if (ret != GST_FLOW_OK) {
g_object_unref (handle);
gst_video_codec_state_unref (output_state);
GST_ERROR_OBJECT (rsvg, "Buffer allocation failed %s",
gst_flow_get_name (ret));
return ret;
}
GST_LOG_OBJECT (rsvg, "render image at %d x %d",
GST_VIDEO_INFO_HEIGHT (&output_state->info),
GST_VIDEO_INFO_WIDTH (&output_state->info));
if (!gst_video_frame_map (&vframe,
&output_state->info, frame->output_buffer, GST_MAP_READWRITE)) {
GST_ERROR_OBJECT (rsvg, "Failed to get SVG image");
g_object_unref (handle);
gst_video_codec_state_unref (output_state);
return GST_FLOW_ERROR;
}
surface =
cairo_image_surface_create_for_data (GST_VIDEO_FRAME_PLANE_DATA (&vframe,
0), CAIRO_FORMAT_ARGB32, GST_VIDEO_FRAME_WIDTH (&vframe),
GST_VIDEO_FRAME_HEIGHT (&vframe), GST_VIDEO_FRAME_PLANE_STRIDE (&vframe,
0));
cr = cairo_create (surface);
cairo_set_operator (cr, CAIRO_OPERATOR_CLEAR);
cairo_set_source_rgba (cr, 1.0, 1.0, 1.0, 0.0);
cairo_paint (cr);
cairo_set_operator (cr, CAIRO_OPERATOR_OVER);
cairo_set_source_rgba (cr, 0.0, 0.0, 0.0, 1.0);
scalex = scaley = 1.0;
if (GST_VIDEO_INFO_WIDTH (&output_state->info) != dimension.width) {
scalex =
((gdouble) GST_VIDEO_INFO_WIDTH (&output_state->info)) /
((gdouble) dimension.width);
}
if (GST_VIDEO_INFO_HEIGHT (&output_state->info) != dimension.height) {
scaley =
((gdouble) GST_VIDEO_INFO_HEIGHT (&output_state->info)) /
((gdouble) dimension.height);
}
cairo_scale (cr, scalex, scaley);
rsvg_handle_render_cairo (handle, cr);
g_object_unref (handle);
cairo_destroy (cr);
cairo_surface_destroy (surface);
/* Now unpremultiply Cairo's ARGB to match GStreamer's */
gst_rsvg_decode_unpremultiply (GST_VIDEO_FRAME_PLANE_DATA (&vframe, 0),
GST_VIDEO_FRAME_WIDTH (&vframe), GST_VIDEO_FRAME_HEIGHT (&vframe));
gst_video_codec_state_unref (output_state);
gst_buffer_unmap (buffer, &minfo);
gst_video_frame_unmap (&vframe);
return ret;
}
static gboolean
verify_buffer (buffer_verify_data_s * vdata, GstBuffer * buffer)
{
if (vdata->discard) {
/* check separate header NALs */
gint i = vdata->buffer_counter;
guint ofs;
/* SEI with start code prefix with 2 0-bytes */
ofs = i == 1;
fail_unless (i <= 2);
fail_unless (gst_buffer_get_size (buffer) == ctx_headers[i].size - ofs);
fail_unless (gst_buffer_memcmp (buffer, 0, ctx_headers[i].data + ofs,
gst_buffer_get_size (buffer)) == 0);
} else {
GstMapInfo map;
gst_buffer_map (buffer, &map, GST_MAP_READ);
fail_unless (map.size > 4);
/* only need to check avc and bs-to-nal output case */
if (GST_READ_UINT24_BE (map.data) == 0x01) {
/* in bs-to-nal, a leading 0x00 is stripped from output */
fail_unless (gst_buffer_get_size (buffer) ==
vdata->data_to_verify_size - 1);
fail_unless (gst_buffer_memcmp (buffer, 0, vdata->data_to_verify + 1,
vdata->data_to_verify_size - 1) == 0);
gst_buffer_unmap (buffer, &map);
return TRUE;
} else if (GST_READ_UINT32_BE (map.data) == 0x01) {
/* this is not avc, use default tests from parser.c */
gst_buffer_unmap (buffer, &map);
return FALSE;
}
/* header is merged in initial frame */
if (vdata->buffer_counter == 0) {
guint8 *data = map.data;
fail_unless (map.size == vdata->data_to_verify_size +
ctx_headers[0].size + ctx_headers[1].size + ctx_headers[2].size);
fail_unless (GST_READ_UINT32_BE (data) == ctx_headers[0].size - 4);
fail_unless (memcmp (data + 4, ctx_headers[0].data + 4,
ctx_headers[0].size - 4) == 0);
data += ctx_headers[0].size;
fail_unless (GST_READ_UINT32_BE (data) == ctx_headers[1].size - 4);
fail_unless (memcmp (data + 4, ctx_headers[1].data + 4,
ctx_headers[1].size - 4) == 0);
data += ctx_headers[1].size;
fail_unless (GST_READ_UINT32_BE (data) == ctx_headers[2].size - 4);
fail_unless (memcmp (data + 4, ctx_headers[2].data + 4,
ctx_headers[2].size - 4) == 0);
data += ctx_headers[2].size;
fail_unless (GST_READ_UINT32_BE (data) == vdata->data_to_verify_size - 4);
fail_unless (memcmp (data + 4, vdata->data_to_verify + 4,
vdata->data_to_verify_size - 4) == 0);
} else {
fail_unless (GST_READ_UINT32_BE (map.data) == map.size - 4);
fail_unless (map.size == vdata->data_to_verify_size);
fail_unless (memcmp (map.data + 4, vdata->data_to_verify + 4,
map.size - 4) == 0);
}
gst_buffer_unmap (buffer, &map);
return TRUE;
}
return FALSE;
}
GstFlowReturn
gst_kate_util_decoder_base_chain_kate_packet (GstKateDecoderBase * decoder,
GstElement * element, GstPad * pad, GstBuffer * buf, GstPad * srcpad,
GstPad * tagpad, GstCaps ** src_caps, const kate_event ** ev)
{
kate_packet kp;
int ret;
GstFlowReturn rflow = GST_FLOW_OK;
gboolean is_header;
GstMapInfo info;
gsize header_size;
guint8 header[1];
header_size = gst_buffer_extract (buf, 0, header, 1);
GST_DEBUG_OBJECT (element,
"got kate packet, %" G_GSIZE_FORMAT " bytes, type %02x",
gst_buffer_get_size (buf), header_size == 0 ? -1 : header[0]);
is_header = header_size > 0 && (header[0] & 0x80);
if (!is_header && decoder->tags_changed) {
/* after we've processed headers, send any tags before processing the data packet */
GST_DEBUG_OBJECT (element, "Not a header, sending tags for pad %s:%s",
GST_DEBUG_PAD_NAME (tagpad));
gst_pad_push_event (tagpad,
gst_kate_util_decoder_base_get_tag_event (decoder));
}
if (gst_buffer_map (buf, &info, GST_MAP_READ)) {
kate_packet_wrap (&kp, info.size, info.data);
ret = kate_high_decode_packetin (&decoder->k, &kp, ev);
gst_buffer_unmap (buf, &info);
} else {
GST_ELEMENT_ERROR (element, STREAM, DECODE, (NULL),
("Failed to map buffer"));
return GST_FLOW_ERROR;
}
if (G_UNLIKELY (ret < 0)) {
GST_ELEMENT_ERROR (element, STREAM, DECODE, (NULL),
("Failed to decode Kate packet: %s",
gst_kate_util_get_error_message (ret)));
return GST_FLOW_ERROR;
}
if (G_UNLIKELY (ret > 0)) {
GST_DEBUG_OBJECT (element,
"kate_high_decode_packetin has received EOS packet");
}
/* headers may be interesting to retrieve information from */
if (G_UNLIKELY (is_header)) {
switch (header[0]) {
case 0x80: /* ID header */
GST_INFO_OBJECT (element, "Parsed ID header: language %s, category %s",
decoder->k.ki->language, decoder->k.ki->category);
if (src_caps) {
if (*src_caps) {
gst_caps_unref (*src_caps);
*src_caps = NULL;
}
if (strcmp (decoder->k.ki->category, "K-SPU") == 0 ||
strcmp (decoder->k.ki->category, "spu-subtitles") == 0) {
*src_caps = gst_caps_new_empty_simple ("subpicture/x-dvd");
} else if (decoder->k.ki->text_markup_type == kate_markup_none) {
*src_caps = gst_caps_new_simple ("text/x-raw", "format",
G_TYPE_STRING, "utf8", NULL);
} else {
*src_caps = gst_caps_new_simple ("text/x-raw", "format",
G_TYPE_STRING, "pango-markup", NULL);
}
GST_INFO_OBJECT (srcpad, "Setting caps: %" GST_PTR_FORMAT, *src_caps);
if (!gst_pad_set_caps (srcpad, *src_caps)) {
GST_ERROR_OBJECT (srcpad, "Failed to set caps %" GST_PTR_FORMAT,
*src_caps);
}
}
if (decoder->k.ki->language && *decoder->k.ki->language) {
GstTagList *tags = gst_tag_list_new_empty ();
gchar *lang_code;
/* en_GB -> en */
lang_code = g_ascii_strdown (decoder->k.ki->language, -1);
g_strdelimit (lang_code, NULL, '\0');
gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_LANGUAGE_CODE,
lang_code, NULL);
g_free (lang_code);
/* TODO: category - where should it go ? */
gst_kate_util_decoder_base_add_tags (decoder, tags, TRUE);
}
/* update properties */
if (decoder->language)
g_free (decoder->language);
decoder->language = g_strdup (decoder->k.ki->language);
if (decoder->category)
g_free (decoder->category);
decoder->category = g_strdup (decoder->k.ki->category);
decoder->original_canvas_width = decoder->k.ki->original_canvas_width;
decoder->original_canvas_height = decoder->k.ki->original_canvas_height;
/* we can now send away any event we've delayed, as the src pad now has caps */
gst_kate_util_decoder_base_drain_event_queue (decoder);
break;
case 0x81: /* Vorbis comments header */
GST_INFO_OBJECT (element, "Parsed comments header");
{
gchar *encoder = NULL;
GstTagList *list = gst_tag_list_from_vorbiscomment_buffer (buf,
(const guint8 *) "\201kate\0\0\0\0", 9, &encoder);
if (!list) {
GST_ERROR_OBJECT (element, "failed to decode comment header");
list = gst_tag_list_new_empty ();
}
if (encoder) {
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER, encoder, NULL);
g_free (encoder);
}
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_SUBTITLE_CODEC, "Kate", NULL);
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER_VERSION, decoder->k.ki->bitstream_version_major,
NULL);
gst_kate_util_decoder_base_add_tags (decoder, list, TRUE);
if (decoder->initialized) {
gst_pad_push_event (tagpad,
gst_event_new_tag (gst_tag_list_ref (decoder->tags)));
}
}
break;
default:
break;
}
}
#if ((KATE_VERSION_MAJOR<<16)|(KATE_VERSION_MINOR<<8)|KATE_VERSION_PATCH) >= 0x000400
else if (*ev && (*ev)->meta) {
int count = kate_meta_query_count ((*ev)->meta);
if (count > 0) {
GstTagList *evtags = gst_tag_list_new_empty ();
int idx;
GST_DEBUG_OBJECT (decoder, "Kate event has %d attached metadata", count);
for (idx = 0; idx < count; ++idx) {
const char *tag, *value;
size_t len;
if (kate_meta_query ((*ev)->meta, idx, &tag, &value, &len) < 0) {
GST_WARNING_OBJECT (decoder, "Failed to retrieve metadata %d", idx);
} else {
if (gst_kate_util_is_utf8_string (value, len)) {
gchar *compound = g_strdup_printf ("%s=%s", tag, value);
GST_DEBUG_OBJECT (decoder,
"Metadata %d: %s=%s (%" G_GSIZE_FORMAT " bytes)", idx, tag,
value, len);
gst_tag_list_add (evtags, GST_TAG_MERGE_APPEND,
GST_TAG_EXTENDED_COMMENT, compound, NULL);
g_free (compound);
} else {
GST_INFO_OBJECT (decoder,
"Metadata %d, (%s, %" G_GSIZE_FORMAT
" bytes) is binary, ignored", idx, tag, len);
}
}
}
gst_kate_util_decoder_base_add_tags (decoder, evtags, TRUE);
gst_pad_push_event (tagpad,
gst_kate_util_decoder_base_get_tag_event (decoder));
}
}
#endif
return rflow;
}
/*!
* \brief CvCapture_GStreamer::retrieveFrame
* \return IplImage pointer. [Transfer Full]
* Retreive the previously grabbed buffer, and wrap it in an IPLImage structure
*/
IplImage * CvCapture_GStreamer::retrieveFrame(int)
{
if(!buffer)
return 0;
//construct a frame header if we did not have any yet
if(!frame)
{
gint height, width;
//reuse the caps ptr
if (buffer_caps)
gst_caps_unref(buffer_caps);
#if GST_VERSION_MAJOR == 0
buffer_caps = gst_buffer_get_caps(buffer);
#else
buffer_caps = gst_sample_get_caps(sample);
#endif
// bail out in no caps
assert(gst_caps_get_size(buffer_caps) == 1);
GstStructure* structure = gst_caps_get_structure(buffer_caps, 0);
// bail out if width or height are 0
if(!gst_structure_get_int(structure, "width", &width) ||
!gst_structure_get_int(structure, "height", &height))
{
return 0;
}
int depth = 3;
#if GST_VERSION_MAJOR > 0
depth = 0;
const gchar* name = gst_structure_get_name(structure);
const gchar* format = gst_structure_get_string(structure, "format");
if (!name || !format)
return 0;
// we support 3 types of data:
// video/x-raw, format=BGR -> 8bit, 3 channels
// video/x-raw, format=GRAY8 -> 8bit, 1 channel
// video/x-bayer -> 8bit, 1 channel
// bayer data is never decoded, the user is responsible for that
// everything is 8 bit, so we just test the caps for bit depth
if (strcasecmp(name, "video/x-raw") == 0)
{
if (strcasecmp(format, "BGR") == 0) {
depth = 3;
}
else if(strcasecmp(format, "GRAY8") == 0){
depth = 1;
}
}
else if (strcasecmp(name, "video/x-bayer") == 0)
{
depth = 1;
}
#endif
if (depth > 0) {
frame = cvCreateImageHeader(cvSize(width, height), IPL_DEPTH_8U, depth);
}else{
return 0;
}
}
// gstreamer expects us to handle the memory at this point
// so we can just wrap the raw buffer and be done with it
#if GST_VERSION_MAJOR == 0
frame->imageData = (char *)GST_BUFFER_DATA(buffer);
#else
// the data ptr in GstMapInfo is only valid throughout the mapifo objects life.
// TODO: check if reusing the mapinfo object is ok.
gboolean success = gst_buffer_map(buffer,info, (GstMapFlags)GST_MAP_READ);
if (!success){
//something weird went wrong here. abort. abort.
//fprintf(stderr,"GStreamer: unable to map buffer");
return 0;
}
frame->imageData = (char*)info->data;
gst_buffer_unmap(buffer,info);
#endif
return frame;
}
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;
}
/*!
* \brief CvVideoWriter_GStreamer::writeFrame
* \param image
* \return
* Pushes the given frame on the pipeline.
* The timestamp for the buffer is generated from the framerate set in open
* and ensures a smooth video
*/
bool CvVideoWriter_GStreamer::writeFrame( const IplImage * image )
{
CV_FUNCNAME("CvVideoWriter_GStreamer::writerFrame");
GstClockTime duration, timestamp;
GstFlowReturn ret;
int size;
__BEGIN__;
handleMessage(pipeline);
if (input_pix_fmt == GST_VIDEO_FORMAT_BGR) {
if (image->nChannels != 3 || image->depth != IPL_DEPTH_8U) {
CV_ERROR(CV_StsUnsupportedFormat, "cvWriteFrame() needs images with depth = IPL_DEPTH_8U and nChannels = 3.");
}
}
#if FULL_GST_VERSION >= VERSION_NUM(0,10,29)
else if (input_pix_fmt == GST_VIDEO_FORMAT_GRAY8) {
if (image->nChannels != 1 || image->depth != IPL_DEPTH_8U) {
CV_ERROR(CV_StsUnsupportedFormat, "cvWriteFrame() needs images with depth = IPL_DEPTH_8U and nChannels = 1.");
}
}
#endif
else {
assert(false);
}
size = image->imageSize;
duration = ((double)1/framerate) * GST_SECOND;
timestamp = num_frames * duration;
//gst_app_src_push_buffer takes ownership of the buffer, so we need to supply it a copy
#if GST_VERSION_MAJOR == 0
buffer = gst_buffer_new_and_alloc (size);
memcpy(GST_BUFFER_DATA (buffer), (guint8*)image->imageData, size);
GST_BUFFER_DURATION(buffer) = duration;
GST_BUFFER_TIMESTAMP(buffer) = timestamp;
#else
buffer = gst_buffer_new_allocate (NULL, size, NULL);
GstMapInfo info;
gst_buffer_map(buffer, &info, (GstMapFlags)GST_MAP_READ);
memcpy(info.data, (guint8*)image->imageData, size);
gst_buffer_unmap(buffer, &info);
GST_BUFFER_DURATION(buffer) = duration;
GST_BUFFER_PTS(buffer) = timestamp;
GST_BUFFER_DTS(buffer) = timestamp;
#endif
//set the current number in the frame
GST_BUFFER_OFFSET(buffer) = num_frames;
ret = gst_app_src_push_buffer(GST_APP_SRC(source), buffer);
if (ret != GST_FLOW_OK) {
/* something wrong, stop pushing */
assert(false);
}
//gst_debug_bin_to_dot_file (GST_BIN(pipeline), GST_DEBUG_GRAPH_SHOW_ALL, "pipeline");
++num_frames;
__END__;
return true;
}
static GstFlowReturn
gst_kate_parse_push_headers (GstKateParse * parse)
{
/* mark and put on caps */
GstCaps *caps;
GstBuffer *outbuf;
kate_packet packet;
GList *headers, *outbuf_list = NULL;
int ret;
gboolean res;
/* get the headers into the caps, passing them to kate as we go */
caps =
gst_kate_util_set_header_on_caps (&parse->element,
gst_pad_get_current_caps (parse->sinkpad), parse->streamheader);
if (G_UNLIKELY (!caps)) {
GST_ELEMENT_ERROR (parse, STREAM, DECODE, (NULL),
("Failed to set headers on caps"));
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (parse, "here are the caps: %" GST_PTR_FORMAT, caps);
res = gst_pad_set_caps (parse->srcpad, caps);
gst_caps_unref (caps);
if (G_UNLIKELY (!res)) {
GST_WARNING_OBJECT (parse->srcpad, "Failed to set caps on source pad");
return GST_FLOW_NOT_NEGOTIATED;
}
headers = parse->streamheader;
while (headers) {
GstMapInfo info;
outbuf = GST_BUFFER_CAST (headers->data);
if (!gst_buffer_map (outbuf, &info, GST_MAP_READ)) {
GST_WARNING_OBJECT (outbuf, "Failed to map buffer");
continue;
}
kate_packet_wrap (&packet, info.size, info.data);
ret = kate_decode_headerin (&parse->ki, &parse->kc, &packet);
if (G_UNLIKELY (ret < 0)) {
GST_WARNING_OBJECT (parse, "Failed to decode header: %s",
gst_kate_util_get_error_message (ret));
}
gst_buffer_unmap (outbuf, &info);
/* takes ownership of outbuf, which was previously in parse->streamheader */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_HEADER);
outbuf_list = g_list_append (outbuf_list, outbuf);
headers = headers->next;
}
/* first process queued events */
gst_kate_parse_drain_event_queue (parse);
/* push out buffers, ignoring return value... */
headers = outbuf_list;
while (headers) {
outbuf = GST_BUFFER_CAST (headers->data);
gst_pad_push (parse->srcpad, outbuf);
headers = headers->next;
}
g_list_free (outbuf_list);
g_list_free (parse->streamheader);
parse->streamheader = NULL;
parse->streamheader_sent = TRUE;
return GST_FLOW_OK;
}
static GstFlowReturn gst_imx_audio_uniaudio_dec_handle_frame(GstAudioDecoder *dec, GstBuffer *buffer)
{
GstMapInfo in_map;
GstBuffer *out_buffer;
gsize avail_out_size;
GstImxAudioUniaudioDec *imx_audio_uniaudio_dec = GST_IMX_AUDIO_UNIAUDIO_DEC(dec);
int32 dec_ret;
uint32 offset = 0;
uint8 *in_buf = NULL;
uint32 in_size = 0;
gboolean dec_loop = TRUE, flow_error = FALSE;
/* With some formats such as Vorbis, the first few buffers are actually redundant,
* since they contain codec data that was already specified in codec_data or
* streamheader caps earlier. If this is the case, skip these buffers. */
if (imx_audio_uniaudio_dec->skip_header_counter < imx_audio_uniaudio_dec->num_vorbis_headers)
{
GST_TRACE_OBJECT(dec, "skipping header buffer #%u", imx_audio_uniaudio_dec->skip_header_counter);
++imx_audio_uniaudio_dec->skip_header_counter;
return gst_audio_decoder_finish_frame(dec, NULL, 1);
}
if (buffer != NULL)
{
gst_buffer_map(buffer, &in_map, GST_MAP_READ);
in_buf = in_map.data;
in_size = in_map.size;
}
while (dec_loop)
{
GstBuffer *tmp_buf;
uint8 *out_buf = NULL;
uint32 out_size = 0;
if (buffer != NULL)
GST_TRACE_OBJECT(dec, "feeding %" G_GUINT32_FORMAT " bytes to the decoder", (guint32)in_size);
else
GST_TRACE_OBJECT(dec, "draining decoder");
dec_ret = imx_audio_uniaudio_dec->codec->decode_frame(
imx_audio_uniaudio_dec->handle,
in_buf, in_size,
&offset,
&out_buf, &out_size
);
GST_TRACE_OBJECT(dec, "decode_frame: return 0x%x offset %" G_GUINT32_FORMAT " out_size %" G_GUINT32_FORMAT, (unsigned int)dec_ret, (guint32)offset, (guint32)out_size);
if ((out_buf != NULL) && (out_size > 0))
{
tmp_buf = gst_audio_decoder_allocate_output_buffer(dec, out_size);
tmp_buf = gst_buffer_make_writable(tmp_buf);
gst_buffer_fill(tmp_buf, 0, out_buf, out_size);
gst_adapter_push(imx_audio_uniaudio_dec->out_adapter, tmp_buf);
}
if (out_buf != NULL)
{
gst_imx_audio_uniaudio_dec_free(out_buf);
}
if ((buffer != NULL) && (offset == in_map.size))
{
dec_loop = FALSE;
}
switch (dec_ret)
{
case ACODEC_SUCCESS:
break;
case ACODEC_END_OF_STREAM:
dec_loop = FALSE;
break;
case ACODEC_NOT_ENOUGH_DATA:
break;
case ACODEC_CAPIBILITY_CHANGE:
break;
default:
{
dec_loop = FALSE;
flow_error = TRUE;
GST_ELEMENT_ERROR(dec, STREAM, DECODE, ("could not decode"), ("error message: %s", imx_audio_uniaudio_dec->codec->get_last_error(imx_audio_uniaudio_dec->handle)));
}
}
}
if (buffer != NULL)
gst_buffer_unmap(buffer, &in_map);
if (flow_error)
return GST_FLOW_ERROR;
if (!(imx_audio_uniaudio_dec->has_audioinfo_set))
{
UniACodecParameter parameter;
GstAudioFormat pcm_fmt;
GstAudioInfo audio_info;
imx_audio_uniaudio_dec->codec->get_parameter(imx_audio_uniaudio_dec->handle, UNIA_OUTPUT_PCM_FORMAT, ¶meter);
if ((parameter.outputFormat.width == 0) || (parameter.outputFormat.depth == 0))
{
GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "no output format available yet");
return gst_audio_decoder_finish_frame(dec, NULL, 1);
}
GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "output sample width: %" G_GUINT32_FORMAT " depth: %" G_GUINT32_FORMAT, (guint32)(parameter.outputFormat.width), (guint32)(parameter.outputFormat.depth));
pcm_fmt = gst_audio_format_build_integer(TRUE, G_BYTE_ORDER, parameter.outputFormat.width, parameter.outputFormat.depth);
GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "setting output format to: %s %d Hz %d channels", gst_audio_format_to_string(pcm_fmt), (gint)(parameter.outputFormat.samplerate), (gint)(parameter.outputFormat.channels));
gst_imx_audio_uniaudio_dec_clear_channel_positions(imx_audio_uniaudio_dec);
gst_imx_audio_uniaudio_dec_fill_channel_positions(imx_audio_uniaudio_dec, parameter.outputFormat.layout, parameter.outputFormat.channels);
imx_audio_uniaudio_dec->pcm_format = pcm_fmt;
imx_audio_uniaudio_dec->num_channels = parameter.outputFormat.channels;
gst_audio_info_set_format(
&audio_info,
pcm_fmt,
parameter.outputFormat.samplerate,
parameter.outputFormat.channels,
imx_audio_uniaudio_dec->reordered_channel_positions
);
gst_audio_decoder_set_output_format(dec, &audio_info);
imx_audio_uniaudio_dec->has_audioinfo_set = TRUE;
}
avail_out_size = gst_adapter_available(imx_audio_uniaudio_dec->out_adapter);
if (avail_out_size > 0)
{
out_buffer = gst_adapter_take_buffer(imx_audio_uniaudio_dec->out_adapter, avail_out_size);
if (imx_audio_uniaudio_dec->original_channel_positions != imx_audio_uniaudio_dec->reordered_channel_positions)
{
gst_audio_buffer_reorder_channels(
out_buffer,
imx_audio_uniaudio_dec->pcm_format,
imx_audio_uniaudio_dec->num_channels,
imx_audio_uniaudio_dec->original_channel_positions,
imx_audio_uniaudio_dec->reordered_channel_positions
);
}
return gst_audio_decoder_finish_frame(dec, out_buffer, 1);
}
else
{
return gst_audio_decoder_finish_frame(dec, NULL, 1);
}
}
static gboolean
gst_amc_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
GstAmcAudioDec *self;
GstStructure *s;
GstAmcFormat *format;
const gchar *mime;
gboolean is_format_change = FALSE;
gboolean needs_disable = FALSE;
gchar *format_string;
gint rate, channels;
GError *err = NULL;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Setting new caps %" GST_PTR_FORMAT, caps);
/* Check if the caps change is a real format change or if only irrelevant
* parts of the caps have changed or nothing at all.
*/
is_format_change |= (!self->input_caps
|| !gst_caps_is_equal (self->input_caps, caps));
needs_disable = self->started;
/* If the component is not started and a real format change happens
* we have to restart the component. If no real format change
* happened we can just exit here.
*/
if (needs_disable && !is_format_change) {
/* Framerate or something minor changed */
self->input_caps_changed = TRUE;
GST_DEBUG_OBJECT (self,
"Already running and caps did not change the format");
return TRUE;
}
if (needs_disable && is_format_change) {
gst_amc_audio_dec_drain (self);
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
gst_amc_audio_dec_stop (GST_AUDIO_DECODER (self));
GST_AUDIO_DECODER_STREAM_LOCK (self);
gst_amc_audio_dec_close (GST_AUDIO_DECODER (self));
if (!gst_amc_audio_dec_open (GST_AUDIO_DECODER (self))) {
GST_ERROR_OBJECT (self, "Failed to open codec again");
return FALSE;
}
if (!gst_amc_audio_dec_start (GST_AUDIO_DECODER (self))) {
GST_ERROR_OBJECT (self, "Failed to start codec again");
}
}
/* srcpad task is not running at this point */
mime = caps_to_mime (caps);
if (!mime) {
GST_ERROR_OBJECT (self, "Failed to convert caps to mime");
return FALSE;
}
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "rate", &rate) ||
!gst_structure_get_int (s, "channels", &channels)) {
GST_ERROR_OBJECT (self, "Failed to get rate/channels");
return FALSE;
}
format = gst_amc_format_new_audio (mime, rate, channels, &err);
if (!format) {
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
/* FIXME: These buffers needs to be valid until the codec is stopped again */
g_list_foreach (self->codec_datas, (GFunc) gst_buffer_unref, NULL);
g_list_free (self->codec_datas);
self->codec_datas = NULL;
if (gst_structure_has_field (s, "codec_data")) {
const GValue *h = gst_structure_get_value (s, "codec_data");
GstBuffer *codec_data = gst_value_get_buffer (h);
GstMapInfo minfo;
guint8 *data;
gst_buffer_map (codec_data, &minfo, GST_MAP_READ);
data = g_memdup (minfo.data, minfo.size);
self->codec_datas = g_list_prepend (self->codec_datas, data);
gst_amc_format_set_buffer (format, "csd-0", data, minfo.size, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_buffer_unmap (codec_data, &minfo);
} else if (gst_structure_has_field (s, "streamheader")) {
const GValue *sh = gst_structure_get_value (s, "streamheader");
gint nsheaders = gst_value_array_get_size (sh);
GstBuffer *buf;
const GValue *h;
gint i, j;
gchar *fname;
GstMapInfo minfo;
guint8 *data;
for (i = 0, j = 0; i < nsheaders; i++) {
h = gst_value_array_get_value (sh, i);
buf = gst_value_get_buffer (h);
if (strcmp (mime, "audio/vorbis") == 0) {
guint8 header_type;
gst_buffer_extract (buf, 0, &header_type, 1);
/* Only use the identification and setup packets */
if (header_type != 0x01 && header_type != 0x05)
continue;
}
fname = g_strdup_printf ("csd-%d", j);
gst_buffer_map (buf, &minfo, GST_MAP_READ);
data = g_memdup (minfo.data, minfo.size);
self->codec_datas = g_list_prepend (self->codec_datas, data);
gst_amc_format_set_buffer (format, fname, data, minfo.size, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_buffer_unmap (buf, &minfo);
g_free (fname);
j++;
}
}
format_string = gst_amc_format_to_string (format, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
GST_DEBUG_OBJECT (self, "Configuring codec with format: %s",
GST_STR_NULL (format_string));
g_free (format_string);
if (!gst_amc_codec_configure (self->codec, format, NULL, 0, &err)) {
GST_ERROR_OBJECT (self, "Failed to configure codec");
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
gst_amc_format_free (format);
if (!gst_amc_codec_start (self->codec, &err)) {
GST_ERROR_OBJECT (self, "Failed to start codec");
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
self->spf = -1;
/* TODO: Implement for other codecs too */
if (gst_structure_has_name (s, "audio/mpeg")) {
gint mpegversion = -1;
gst_structure_get_int (s, "mpegversion", &mpegversion);
if (mpegversion == 1) {
gint layer = -1, mpegaudioversion = -1;
gst_structure_get_int (s, "layer", &layer);
gst_structure_get_int (s, "mpegaudioversion", &mpegaudioversion);
if (layer == 1)
self->spf = 384;
else if (layer == 2)
self->spf = 1152;
else if (layer == 3 && mpegaudioversion != -1)
self->spf = (mpegaudioversion == 1 ? 1152 : 576);
}
}
self->started = TRUE;
self->input_caps_changed = TRUE;
/* Start the srcpad loop again */
self->flushing = FALSE;
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_amc_audio_dec_loop, decoder, NULL);
return TRUE;
}
static GstFlowReturn
gst_openjpeg_enc_handle_frame (GstVideoEncoder * encoder,
GstVideoCodecFrame * frame)
{
GstOpenJPEGEnc *self = GST_OPENJPEG_ENC (encoder);
GstFlowReturn ret = GST_FLOW_OK;
#ifdef HAVE_OPENJPEG_1
opj_cinfo_t *enc;
GstMapInfo map;
guint length;
opj_cio_t *io;
#else
opj_codec_t *enc;
opj_stream_t *stream;
MemStream mstream;
#endif
opj_image_t *image;
GstVideoFrame vframe;
GST_DEBUG_OBJECT (self, "Handling frame");
enc = opj_create_compress (self->codec_format);
if (!enc)
goto initialization_error;
#ifdef HAVE_OPENJPEG_1
if (G_UNLIKELY (gst_debug_category_get_threshold (GST_CAT_DEFAULT) >=
GST_LEVEL_TRACE)) {
opj_event_mgr_t callbacks;
callbacks.error_handler = gst_openjpeg_enc_opj_error;
callbacks.warning_handler = gst_openjpeg_enc_opj_warning;
callbacks.info_handler = gst_openjpeg_enc_opj_info;
opj_set_event_mgr ((opj_common_ptr) enc, &callbacks, self);
} else {
opj_set_event_mgr ((opj_common_ptr) enc, NULL, NULL);
}
#else
if (G_UNLIKELY (gst_debug_category_get_threshold (GST_CAT_DEFAULT) >=
GST_LEVEL_TRACE)) {
opj_set_info_handler (enc, gst_openjpeg_enc_opj_info, self);
opj_set_warning_handler (enc, gst_openjpeg_enc_opj_warning, self);
opj_set_error_handler (enc, gst_openjpeg_enc_opj_error, self);
} else {
opj_set_info_handler (enc, NULL, NULL);
opj_set_warning_handler (enc, NULL, NULL);
opj_set_error_handler (enc, NULL, NULL);
}
#endif
if (!gst_video_frame_map (&vframe, &self->input_state->info,
frame->input_buffer, GST_MAP_READ))
goto map_read_error;
image = gst_openjpeg_enc_fill_image (self, &vframe);
if (!image)
goto fill_image_error;
gst_video_frame_unmap (&vframe);
if (vframe.info.finfo->flags & GST_VIDEO_FORMAT_FLAG_RGB) {
self->params.tcp_mct = 1;
}
opj_setup_encoder (enc, &self->params, image);
#ifdef HAVE_OPENJPEG_1
io = opj_cio_open ((opj_common_ptr) enc, NULL, 0);
if (!io)
goto open_error;
if (!opj_encode (enc, io, image, NULL))
goto encode_error;
opj_image_destroy (image);
length = cio_tell (io);
ret =
gst_video_encoder_allocate_output_frame (encoder, frame,
length + (self->is_jp2c ? 8 : 0));
if (ret != GST_FLOW_OK)
goto allocate_error;
gst_buffer_fill (frame->output_buffer, self->is_jp2c ? 8 : 0, io->buffer,
length);
if (self->is_jp2c) {
gst_buffer_map (frame->output_buffer, &map, GST_MAP_WRITE);
GST_WRITE_UINT32_BE (map.data, length + 8);
GST_WRITE_UINT32_BE (map.data + 4, GST_MAKE_FOURCC ('j', 'p', '2', 'c'));
gst_buffer_unmap (frame->output_buffer, &map);
}
opj_cio_close (io);
opj_destroy_compress (enc);
#else
stream = opj_stream_create (4096, OPJ_FALSE);
if (!stream)
goto open_error;
mstream.allocsize = 4096;
mstream.data = g_malloc (mstream.allocsize);
mstream.offset = 0;
mstream.size = 0;
opj_stream_set_read_function (stream, read_fn);
opj_stream_set_write_function (stream, write_fn);
opj_stream_set_skip_function (stream, skip_fn);
opj_stream_set_seek_function (stream, seek_fn);
opj_stream_set_user_data (stream, &mstream, NULL);
opj_stream_set_user_data_length (stream, mstream.size);
if (!opj_start_compress (enc, image, stream))
goto encode_error;
if (!opj_encode (enc, stream))
goto encode_error;
if (!opj_end_compress (enc, stream))
goto encode_error;
opj_image_destroy (image);
opj_stream_destroy (stream);
opj_destroy_codec (enc);
frame->output_buffer = gst_buffer_new ();
if (self->is_jp2c) {
GstMapInfo map;
GstMemory *mem;
mem = gst_allocator_alloc (NULL, 8, NULL);
gst_memory_map (mem, &map, GST_MAP_WRITE);
GST_WRITE_UINT32_BE (map.data, mstream.size + 8);
GST_WRITE_UINT32_BE (map.data + 4, GST_MAKE_FOURCC ('j', 'p', '2', 'c'));
gst_memory_unmap (mem, &map);
gst_buffer_append_memory (frame->output_buffer, mem);
}
gst_buffer_append_memory (frame->output_buffer,
gst_memory_new_wrapped (0, mstream.data, mstream.allocsize, 0,
mstream.size, NULL, (GDestroyNotify) g_free));
#endif
GST_VIDEO_CODEC_FRAME_SET_SYNC_POINT (frame);
ret = gst_video_encoder_finish_frame (encoder, frame);
return ret;
initialization_error:
{
gst_video_codec_frame_unref (frame);
GST_ELEMENT_ERROR (self, LIBRARY, INIT,
("Failed to initialize OpenJPEG encoder"), (NULL));
return GST_FLOW_ERROR;
}
map_read_error:
{
#ifdef HAVE_OPENJPEG_1
opj_destroy_compress (enc);
#else
opj_destroy_codec (enc);
#endif
gst_video_codec_frame_unref (frame);
GST_ELEMENT_ERROR (self, CORE, FAILED,
("Failed to map input buffer"), (NULL));
return GST_FLOW_ERROR;
}
fill_image_error:
{
#ifdef HAVE_OPENJPEG_1
opj_destroy_compress (enc);
#else
opj_destroy_codec (enc);
#endif
gst_video_frame_unmap (&vframe);
gst_video_codec_frame_unref (frame);
GST_ELEMENT_ERROR (self, LIBRARY, INIT,
("Failed to fill OpenJPEG image"), (NULL));
return GST_FLOW_ERROR;
}
open_error:
{
opj_image_destroy (image);
#ifdef HAVE_OPENJPEG_1
opj_destroy_compress (enc);
#else
opj_destroy_codec (enc);
#endif
gst_video_codec_frame_unref (frame);
GST_ELEMENT_ERROR (self, LIBRARY, INIT,
("Failed to open OpenJPEG data"), (NULL));
return GST_FLOW_ERROR;
}
encode_error:
{
#ifdef HAVE_OPENJPEG_1
opj_cio_close (io);
opj_image_destroy (image);
opj_destroy_compress (enc);
#else
opj_stream_destroy (stream);
g_free (mstream.data);
opj_image_destroy (image);
opj_destroy_codec (enc);
#endif
gst_video_codec_frame_unref (frame);
GST_ELEMENT_ERROR (self, STREAM, ENCODE,
("Failed to encode OpenJPEG stream"), (NULL));
return GST_FLOW_ERROR;
}
#ifdef HAVE_OPENJPEG_1
allocate_error:
{
opj_cio_close (io);
opj_destroy_compress (enc);
gst_video_codec_frame_unref (frame);
GST_ELEMENT_ERROR (self, CORE, FAILED,
("Failed to allocate output buffer"), (NULL));
return ret;
}
#endif
}
static GstFlowReturn
gst_sbc_dec_handle_frame (GstAudioDecoder * audio_dec, GstBuffer * buf)
{
GstSbcDec *dec = GST_SBC_DEC (audio_dec);
GstBuffer *outbuf = NULL;
GstMapInfo out_map;
GstMapInfo in_map;
gsize output_size;
guint num_frames, i;
/* no fancy draining */
if (G_UNLIKELY (buf == NULL))
return GST_FLOW_OK;
gst_buffer_map (buf, &in_map, GST_MAP_READ);
if (G_UNLIKELY (in_map.size == 0))
goto done;
/* we assume all frames are of the same size, this is implied by the
* input caps applying to the whole input buffer, and the parser should
* also have made sure of that */
if (G_UNLIKELY (in_map.size % dec->frame_len != 0))
goto mixed_frames;
num_frames = in_map.size / dec->frame_len;
output_size = num_frames * dec->samples_per_frame * sizeof (gint16);
outbuf = gst_audio_decoder_allocate_output_buffer (audio_dec, output_size);
if (outbuf == NULL)
goto no_buffer;
gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE);
for (i = 0; i < num_frames; ++i) {
gssize ret;
gsize written;
ret = sbc_decode (&dec->sbc, in_map.data + (i * dec->frame_len),
dec->frame_len, out_map.data + (i * dec->samples_per_frame * 2),
dec->samples_per_frame * 2, &written);
if (ret <= 0 || written != (dec->samples_per_frame * 2)) {
GST_WARNING_OBJECT (dec, "decoding error, ret = %" G_GSSIZE_FORMAT ", "
"written = %" G_GSSIZE_FORMAT, ret, written);
break;
}
}
gst_buffer_unmap (outbuf, &out_map);
if (i > 0)
gst_buffer_set_size (outbuf, i * dec->samples_per_frame * 2);
else
gst_buffer_replace (&outbuf, NULL);
done:
gst_buffer_unmap (buf, &in_map);
return gst_audio_decoder_finish_frame (audio_dec, outbuf, 1);
/* ERRORS */
mixed_frames:
{
GST_WARNING_OBJECT (dec, "inconsistent input data/frames, skipping");
goto done;
}
no_buffer:
{
GST_ERROR_OBJECT (dec, "could not allocate output buffer");
goto done;
}
}
static GstFlowReturn
gst_dvdec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstDVDec *dvdec;
guint8 *inframe;
guint8 *outframe_ptrs[3];
gint outframe_pitches[3];
GstMapInfo map;
GstVideoFrame frame;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
guint length;
guint64 cstart, cstop;
gboolean PAL, wide;
dvdec = GST_DVDEC (parent);
gst_buffer_map (buf, &map, GST_MAP_READ);
inframe = map.data;
/* buffer should be at least the size of one NTSC frame, this should
* be enough to decode the header. */
if (G_UNLIKELY (map.size < NTSC_BUFFER))
goto wrong_size;
/* preliminary dropping. unref and return if outside of configured segment */
if ((dvdec->segment.format == GST_FORMAT_TIME) &&
(!(gst_segment_clip (&dvdec->segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (buf),
GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf),
&cstart, &cstop))))
goto dropping;
if (G_UNLIKELY (dv_parse_header (dvdec->decoder, inframe) < 0))
goto parse_header_error;
/* get size */
PAL = dv_system_50_fields (dvdec->decoder);
wide = dv_format_wide (dvdec->decoder);
/* check the buffer is of right size after we know if we are
* dealing with PAL or NTSC */
length = (PAL ? PAL_BUFFER : NTSC_BUFFER);
if (G_UNLIKELY (map.size < length))
goto wrong_size;
dv_parse_packs (dvdec->decoder, inframe);
if (dvdec->video_offset % dvdec->drop_factor != 0)
goto skip;
/* renegotiate on change */
if (PAL != dvdec->PAL || wide != dvdec->wide) {
dvdec->src_negotiated = FALSE;
dvdec->PAL = PAL;
dvdec->wide = wide;
}
dvdec->height = (dvdec->PAL ? PAL_HEIGHT : NTSC_HEIGHT);
dvdec->interlaced = !dv_is_progressive (dvdec->decoder);
/* negotiate if not done yet */
if (!dvdec->src_negotiated) {
if (!gst_dvdec_src_negotiate (dvdec))
goto not_negotiated;
}
if (gst_pad_check_reconfigure (dvdec->srcpad)) {
GstCaps *caps;
caps = gst_pad_get_current_caps (dvdec->srcpad);
gst_dvdec_negotiate_pool (dvdec, caps, &dvdec->vinfo);
gst_caps_unref (caps);
}
ret = gst_buffer_pool_acquire_buffer (dvdec->pool, &outbuf, NULL);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto no_buffer;
gst_video_frame_map (&frame, &dvdec->vinfo, outbuf, GST_MAP_WRITE);
outframe_ptrs[0] = GST_VIDEO_FRAME_COMP_DATA (&frame, 0);
outframe_pitches[0] = GST_VIDEO_FRAME_COMP_STRIDE (&frame, 0);
/* the rest only matters for YUY2 */
if (dvdec->bpp < 3) {
outframe_ptrs[1] = GST_VIDEO_FRAME_COMP_DATA (&frame, 1);
outframe_ptrs[2] = GST_VIDEO_FRAME_COMP_DATA (&frame, 2);
outframe_pitches[1] = GST_VIDEO_FRAME_COMP_STRIDE (&frame, 1);
outframe_pitches[2] = GST_VIDEO_FRAME_COMP_STRIDE (&frame, 2);
}
GST_DEBUG_OBJECT (dvdec, "decoding and pushing buffer");
dv_decode_full_frame (dvdec->decoder, inframe,
e_dv_color_yuv, outframe_ptrs, outframe_pitches);
gst_video_frame_unmap (&frame);
GST_BUFFER_FLAG_UNSET (outbuf, GST_VIDEO_BUFFER_FLAG_TFF);
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET (buf);
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_END (buf);
GST_BUFFER_TIMESTAMP (outbuf) = cstart;
GST_BUFFER_DURATION (outbuf) = cstop - cstart;
ret = gst_pad_push (dvdec->srcpad, outbuf);
skip:
dvdec->video_offset++;
done:
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
return ret;
/* ERRORS */
wrong_size:
{
GST_ELEMENT_ERROR (dvdec, STREAM, DECODE,
(NULL), ("Input buffer too small"));
ret = GST_FLOW_ERROR;
goto done;
}
parse_header_error:
{
GST_ELEMENT_ERROR (dvdec, STREAM, DECODE,
(NULL), ("Error parsing DV header"));
ret = GST_FLOW_ERROR;
goto done;
}
not_negotiated:
{
GST_DEBUG_OBJECT (dvdec, "could not negotiate output");
ret = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
no_buffer:
{
GST_DEBUG_OBJECT (dvdec, "could not allocate buffer");
goto done;
}
dropping:
{
GST_DEBUG_OBJECT (dvdec,
"dropping buffer since it's out of the configured segment");
goto done;
}
}
static GstBuffer *
gst_rtp_amr_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpAMRDepay *rtpamrdepay;
const gint *frame_size;
GstBuffer *outbuf = NULL;
gint payload_len;
GstRTPBuffer rtp = { NULL };
GstMapInfo map;
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
/* setup frame size pointer */
if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB)
frame_size = nb_frame_size;
else
frame_size = wb_frame_size;
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
/* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC,
* no robust sorting, no interleaving data is to be depayloaded */
{
guint8 *payload, *p, *dp;
gint i, num_packets, num_nonempty_packets;
gint amr_len;
gint ILL, ILP;
payload_len = gst_rtp_buffer_get_payload_len (&rtp);
/* need at least 2 bytes for the header */
if (payload_len < 2)
goto too_small;
payload = gst_rtp_buffer_get_payload (&rtp);
/* depay CMR. The CMR is used by the sender to request
* a new encoding mode.
*
* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* | CMR |R|R|R|R|
* +-+-+-+-+-+-+-+-+
*/
/* CMR = (payload[0] & 0xf0) >> 4; */
/* strip CMR header now, pack FT and the data for the decoder */
payload_len -= 1;
payload += 1;
GST_DEBUG_OBJECT (rtpamrdepay, "payload len %d", payload_len);
if (rtpamrdepay->interleaving) {
ILL = (payload[0] & 0xf0) >> 4;
ILP = (payload[0] & 0x0f);
payload_len -= 1;
payload += 1;
if (ILP > ILL)
goto wrong_interleaving;
}
/*
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
* +-+-+-+-+-+-+-+-+..
* |F| FT |Q|P|P| more FT..
* +-+-+-+-+-+-+-+-+..
*/
/* count number of packets by counting the FTs. Also
* count number of amr data bytes and number of non-empty
* packets (this is also the number of CRCs if present). */
amr_len = 0;
num_nonempty_packets = 0;
num_packets = 0;
for (i = 0; i < payload_len; i++) {
gint fr_size;
guint8 FT;
FT = (payload[i] & 0x78) >> 3;
fr_size = frame_size[FT];
GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size);
if (fr_size == -1)
goto wrong_framesize;
if (fr_size > 0) {
amr_len += fr_size;
num_nonempty_packets++;
}
num_packets++;
if ((payload[i] & 0x80) == 0)
break;
}
if (rtpamrdepay->crc) {
/* data len + CRC len + header bytes should be smaller than payload_len */
if (num_packets + num_nonempty_packets + amr_len > payload_len)
goto wrong_length_1;
} else {
/* data len + header bytes should be smaller than payload_len */
if (num_packets + amr_len > payload_len)
goto wrong_length_2;
}
outbuf = gst_buffer_new_and_alloc (payload_len);
/* point to destination */
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
/* point to first data packet */
p = map.data;
dp = payload + num_packets;
if (rtpamrdepay->crc) {
/* skip CRC if present */
dp += num_nonempty_packets;
}
for (i = 0; i < num_packets; i++) {
gint fr_size;
/* copy FT, clear F bit */
*p++ = payload[i] & 0x7f;
fr_size = frame_size[(payload[i] & 0x78) >> 3];
if (fr_size > 0) {
/* copy data packet, FIXME, calc CRC here. */
memcpy (p, dp, fr_size);
p += fr_size;
dp += fr_size;
}
}
gst_buffer_unmap (outbuf, &map);
/* we can set the duration because each packet is 20 milliseconds */
GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;
if (gst_rtp_buffer_get_marker (&rtp)) {
/* marker bit marks a buffer after a talkspurt. */
GST_DEBUG_OBJECT (depayload, "marker bit was set");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
GST_DEBUG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (outbuf));
}
static GstFlowReturn
gst_dirac_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
{
int off;
guint32 next_header;
GstMapInfo map;
guint8 *data;
gsize size;
gboolean have_picture = FALSE;
int offset;
guint framesize = 0;
gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
data = map.data;
size = map.size;
if (G_UNLIKELY (size < 13)) {
*skipsize = 1;
goto out;
}
GST_DEBUG ("%" G_GSIZE_FORMAT ": %02x %02x %02x %02x", size, data[0], data[1],
data[2], data[3]);
if (GST_READ_UINT32_BE (data) != 0x42424344) {
GstByteReader reader;
gst_byte_reader_init (&reader, data, size);
off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffffffff,
0x42424344, 0, size);
if (off < 0) {
*skipsize = size - 3;
goto out;
}
GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
GST_DEBUG ("skipping %d", off);
*skipsize = off;
goto out;
}
/* have sync, parse chunks */
offset = 0;
while (!have_picture) {
GST_DEBUG ("offset %d:", offset);
if (offset + 13 >= size) {
framesize = offset + 13;
goto out;
}
GST_DEBUG ("chunk type %02x", data[offset + 4]);
if (GST_READ_UINT32_BE (data + offset) != 0x42424344) {
GST_DEBUG ("bad header");
*skipsize = 3;
goto out;
}
next_header = GST_READ_UINT32_BE (data + offset + 5);
GST_DEBUG ("next_header %d", next_header);
if (next_header == 0)
next_header = 13;
if (SCHRO_PARSE_CODE_IS_PICTURE (data[offset + 4])) {
have_picture = TRUE;
}
offset += next_header;
if (offset >= size) {
framesize = offset;
goto out;
}
}
gst_buffer_unmap (frame->buffer, &map);
framesize = offset;
GST_DEBUG ("framesize %d", framesize);
g_assert (framesize <= size);
if (data[4] == SCHRO_PARSE_CODE_SEQUENCE_HEADER) {
GstCaps *caps;
GstDiracParse *diracparse = GST_DIRAC_PARSE (parse);
DiracSequenceHeader sequence_header;
int ret;
ret = dirac_sequence_header_parse (&sequence_header, data + 13, size - 13);
if (ret) {
memcpy (&diracparse->sequence_header, &sequence_header,
sizeof (sequence_header));
caps = gst_caps_new_simple ("video/x-dirac",
"width", G_TYPE_INT, sequence_header.width,
"height", G_TYPE_INT, sequence_header.height,
"framerate", GST_TYPE_FRACTION,
sequence_header.frame_rate_numerator,
sequence_header.frame_rate_denominator,
"pixel-aspect-ratio", GST_TYPE_FRACTION,
sequence_header.aspect_ratio_numerator,
sequence_header.aspect_ratio_denominator,
"interlaced", G_TYPE_BOOLEAN, sequence_header.interlaced,
"profile", G_TYPE_INT, sequence_header.profile,
"level", G_TYPE_INT, sequence_header.level, NULL);
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
gst_caps_unref (caps);
gst_base_parse_set_frame_rate (parse,
sequence_header.frame_rate_numerator,
sequence_header.frame_rate_denominator, 0, 0);
}
}
gst_base_parse_set_min_frame_size (parse, 13);
return gst_base_parse_finish_frame (parse, frame, framesize);
out:
gst_buffer_unmap (frame->buffer, &map);
if (framesize)
gst_base_parse_set_min_frame_size (parse, framesize);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_dtmf_detect_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
{
GstDtmfDetect *self = GST_DTMF_DETECT (trans);
gint dtmf_count;
gchar dtmfbuf[MAX_DTMF_DIGITS] = "";
gint i;
GstMapInfo map;
if (GST_BUFFER_IS_DISCONT (buf))
gst_dtmf_detect_state_reset (self);
if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP))
return GST_FLOW_OK;
gst_buffer_map (buf, &map, GST_MAP_READ);
dtmf_rx (self->dtmf_state, (gint16 *) map.data, map.size / 2);
dtmf_count = dtmf_rx_get (self->dtmf_state, dtmfbuf, MAX_DTMF_DIGITS);
if (dtmf_count)
GST_DEBUG_OBJECT (self, "Got %d DTMF events: %s", dtmf_count, dtmfbuf);
else
GST_LOG_OBJECT (self, "Got no DTMF events");
gst_buffer_unmap (buf, &map);
for (i = 0; i < dtmf_count; i++) {
GstMessage *dtmf_message = NULL;
GstStructure *structure;
gint dtmf_payload_event;
GST_DEBUG_OBJECT (self, "Got DTMF event %c", dtmfbuf[i]);
switch (dtmfbuf[i]) {
case '0':
dtmf_payload_event = 0;
break;
case '1':
dtmf_payload_event = 1;
break;
case '2':
dtmf_payload_event = 2;
break;
case '3':
dtmf_payload_event = 3;
break;
case '4':
dtmf_payload_event = 4;
break;
case '5':
dtmf_payload_event = 5;
break;
case '6':
dtmf_payload_event = 6;
break;
case '7':
dtmf_payload_event = 7;
break;
case '8':
dtmf_payload_event = 8;
break;
case '9':
dtmf_payload_event = 9;
break;
case '*':
dtmf_payload_event = 10;
break;
case '#':
dtmf_payload_event = 11;
break;
case 'A':
dtmf_payload_event = 12;
break;
case 'B':
dtmf_payload_event = 13;
break;
case 'C':
dtmf_payload_event = 14;
break;
case 'D':
dtmf_payload_event = 15;
break;
default:
continue;
}
structure = gst_structure_new ("dtmf-event",
"type", G_TYPE_INT, 1,
"number", G_TYPE_INT, dtmf_payload_event,
"method", G_TYPE_INT, 2, NULL);
dtmf_message = gst_message_new_element (GST_OBJECT (self), structure);
gst_element_post_message (GST_ELEMENT (self), dtmf_message);
}
return GST_FLOW_OK;
}
static GstFlowReturn
gst_curl_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
GstCurlBaseSink *sink;
GstMapInfo map;
guint8 *data;
size_t size;
GstFlowReturn ret;
gchar *error;
GST_LOG ("enter render");
sink = GST_CURL_BASE_SINK (bsink);
GST_OBJECT_LOCK (sink);
gst_buffer_map (buf, &map, GST_MAP_READ);
data = map.data;
size = map.size;
/* check if the transfer thread has encountered problems while the
* pipeline thread was working elsewhere */
if (sink->flow_ret != GST_FLOW_OK) {
goto done;
}
g_assert (sink->transfer_cond->data_available == FALSE);
/* if there is no transfer thread created, lets create one */
if (sink->transfer_thread == NULL) {
if (!gst_curl_base_sink_transfer_start_unlocked (sink)) {
sink->flow_ret = GST_FLOW_ERROR;
goto done;
}
}
/* make data available for the transfer thread and notify */
sink->transfer_buf->ptr = data;
sink->transfer_buf->len = size;
sink->transfer_buf->offset = 0;
gst_curl_base_sink_transfer_thread_notify_unlocked (sink);
/* wait for the transfer thread to send the data. This will be notified
* either when transfer is completed by the curl read callback or by
* the thread function if an error has occurred. */
gst_curl_base_sink_wait_for_transfer_thread_to_send_unlocked (sink);
done:
gst_buffer_unmap (buf, &map);
/* Hand over error from transfer thread to streaming thread */
error = sink->error;
sink->error = NULL;
ret = sink->flow_ret;
GST_OBJECT_UNLOCK (sink);
if (error != NULL) {
GST_ERROR_OBJECT (sink, "%s", error);
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, ("%s", error), (NULL));
g_free (error);
}
GST_LOG ("exit render");
return ret;
}
// checkBuffer
void gstCamera::checkBuffer()
{
if( !mAppSink )
return;
// block waiting for the buffer
GstSample* gstSample = gst_app_sink_pull_sample(mAppSink);
if( !gstSample )
{
printf(LOG_GSTREAMER "gstreamer camera -- gst_app_sink_pull_sample() returned NULL...\n");
return;
}
GstBuffer* gstBuffer = gst_sample_get_buffer(gstSample);
if( !gstBuffer )
{
printf(LOG_GSTREAMER "gstreamer camera -- gst_sample_get_buffer() returned NULL...\n");
return;
}
// retrieve
GstMapInfo map;
if( !gst_buffer_map(gstBuffer, &map, GST_MAP_READ) )
{
printf(LOG_GSTREAMER "gstreamer camera -- gst_buffer_map() failed...\n");
return;
}
//gst_util_dump_mem(map.data, map.size);
void* gstData = map.data; //GST_BUFFER_DATA(gstBuffer);
const uint32_t gstSize = map.size; //GST_BUFFER_SIZE(gstBuffer);
if( !gstData )
{
printf(LOG_GSTREAMER "gstreamer camera -- gst_buffer had NULL data pointer...\n");
release_return;
}
// retrieve caps
GstCaps* gstCaps = gst_sample_get_caps(gstSample);
if( !gstCaps )
{
printf(LOG_GSTREAMER "gstreamer camera -- gst_buffer had NULL caps...\n");
release_return;
}
GstStructure* gstCapsStruct = gst_caps_get_structure(gstCaps, 0);
if( !gstCapsStruct )
{
printf(LOG_GSTREAMER "gstreamer camera -- gst_caps had NULL structure...\n");
release_return;
}
// get width & height of the buffer
int width = 0;
int height = 0;
if( !gst_structure_get_int(gstCapsStruct, "width", &width) ||
!gst_structure_get_int(gstCapsStruct, "height", &height) )
{
printf(LOG_GSTREAMER "gstreamer camera -- gst_caps missing width/height...\n");
release_return;
}
if( width < 1 || height < 1 )
release_return;
mWidth = width;
mHeight = height;
mDepth = (gstSize * 8) / (width * height);
mSize = gstSize;
//printf(LOG_GSTREAMER "gstreamer camera recieved %ix%i frame (%u bytes, %u bpp)\n", width, height, gstSize, mDepth);
// make sure ringbuffer is allocated
if( !mRingbufferCPU[0] )
{
for( uint32_t n=0; n < NUM_RINGBUFFERS; n++ )
{
if( !cudaAllocMapped(&mRingbufferCPU[n], &mRingbufferGPU[n], gstSize) )
printf(LOG_CUDA "gstreamer camera -- failed to allocate ringbuffer %u (size=%u)\n", n, gstSize);
}
printf(LOG_CUDA "gstreamer camera -- allocated %u ringbuffers, %u bytes each\n", NUM_RINGBUFFERS, gstSize);
}
// copy to next ringbuffer
const uint32_t nextRingbuffer = (mLatestRingbuffer + 1) % NUM_RINGBUFFERS;
//printf(LOG_GSTREAMER "gstreamer camera -- using ringbuffer #%u for next frame\n", nextRingbuffer);
memcpy(mRingbufferCPU[nextRingbuffer], gstData, gstSize);
gst_buffer_unmap(gstBuffer, &map);
//gst_buffer_unref(gstBuffer);
gst_sample_unref(gstSample);
// update and signal sleeping threads
mRingMutex->lock();
mLatestRingbuffer = nextRingbuffer;
mLatestRetrieved = false;
mRingMutex->unlock();
mWaitEvent->wakeAll();
}
static GstFlowReturn
gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
GstBuffer * input_buffer)
{
GstMpg123AudioDec *mpg123_decoder;
int decode_error;
unsigned char *decoded_bytes;
size_t num_decoded_bytes;
GstFlowReturn retval;
mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
g_assert (mpg123_decoder->handle != NULL);
/* The actual decoding */
{
/* feed input data (if there is any) */
if (G_LIKELY (input_buffer != NULL)) {
GstMapInfo info;
if (gst_buffer_map (input_buffer, &info, GST_MAP_READ)) {
mpg123_feed (mpg123_decoder->handle, info.data, info.size);
gst_buffer_unmap (input_buffer, &info);
} else {
GST_ERROR_OBJECT (mpg123_decoder, "gst_memory_map() failed");
return GST_FLOW_ERROR;
}
}
/* Try to decode a frame */
decoded_bytes = NULL;
num_decoded_bytes = 0;
decode_error = mpg123_decode_frame (mpg123_decoder->handle,
&mpg123_decoder->frame_offset, &decoded_bytes, &num_decoded_bytes);
}
retval = GST_FLOW_OK;
switch (decode_error) {
case MPG123_NEW_FORMAT:
/* As mentioned in gst_mpg123_audio_dec_set_format(), the next audioinfo
* is not set immediately; instead, the code waits for mpg123 to take
* note of the new format, and then sets the audioinfo. This fixes glitches
* with mp3s containing several format headers (for example, first half
* using 44.1kHz, second half 32 kHz) */
GST_LOG_OBJECT (dec,
"mpg123 reported a new format -> setting next srccaps");
gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
num_decoded_bytes);
/* If there is a next audioinfo, use it, then set has_next_audioinfo to
* FALSE, to make sure gst_audio_decoder_set_output_format() isn't called
* again until set_format is called by the base class */
if (mpg123_decoder->has_next_audioinfo) {
if (!gst_audio_decoder_set_output_format (dec,
&(mpg123_decoder->next_audioinfo))) {
GST_WARNING_OBJECT (dec, "Unable to set output format");
retval = GST_FLOW_NOT_NEGOTIATED;
}
mpg123_decoder->has_next_audioinfo = FALSE;
}
break;
case MPG123_NEED_MORE:
case MPG123_OK:
retval = gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder,
decoded_bytes, num_decoded_bytes);
break;
case MPG123_DONE:
/* If this happens, then the upstream parser somehow missed the ending
* of the bitstream */
GST_LOG_OBJECT (dec, "mpg123 is done decoding");
gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
num_decoded_bytes);
retval = GST_FLOW_EOS;
break;
default:
{
/* Anything else is considered an error */
int errcode;
switch (decode_error) {
case MPG123_ERR:
errcode = mpg123_errcode (mpg123_decoder->handle);
break;
default:
errcode = decode_error;
}
switch (errcode) {
case MPG123_BAD_OUTFORMAT:{
GstCaps *input_caps =
gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (dec));
GST_ELEMENT_ERROR (dec, STREAM, FORMAT, (NULL),
("Output sample format could not be used when trying to decode frame. "
"This is typically caused when the input caps (often the sample "
"rate) do not match the actual format of the audio data. "
"Input caps: %" GST_PTR_FORMAT, input_caps)
);
gst_caps_unref (input_caps);
break;
}
default:{
char const *errmsg = mpg123_plain_strerror (errcode);
GST_ERROR_OBJECT (dec, "Reported error: %s", errmsg);
}
}
retval = GST_FLOW_ERROR;
}
}
return retval;
}
static gboolean
gst_ac3_parse_frame_header_eac3 (GstAc3Parse * ac3parse, GstBuffer * buf,
gint skip, guint * frame_size, guint * rate, guint * chans, guint * blks,
guint * sid)
{
GstBitReader bits;
GstMapInfo map;
guint16 frmsiz, sample_rate, blocks;
guint8 strmtyp, fscod, fscod2, acmod, lfe_on, strmid, numblkscod;
gboolean ret = FALSE;
GST_LOG_OBJECT (ac3parse, "parsing e-ac3");
gst_buffer_map (buf, &map, GST_MAP_READ);
gst_bit_reader_init (&bits, map.data, map.size);
gst_bit_reader_skip_unchecked (&bits, skip * 8);
gst_bit_reader_skip_unchecked (&bits, 16);
strmtyp = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* strmtyp */
if (G_UNLIKELY (strmtyp == 3)) {
GST_DEBUG_OBJECT (ac3parse, "bad strmtyp %d", strmtyp);
goto cleanup;
}
strmid = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3); /* substreamid */
frmsiz = gst_bit_reader_get_bits_uint16_unchecked (&bits, 11); /* frmsiz */
fscod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* fscod */
if (fscod == 3) {
fscod2 = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* fscod2 */
if (G_UNLIKELY (fscod2 == 3)) {
GST_DEBUG_OBJECT (ac3parse, "invalid fscod2");
goto cleanup;
}
sample_rate = fscod_rates[fscod2] / 2;
blocks = 6;
} else {
numblkscod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* numblkscod */
sample_rate = fscod_rates[fscod];
blocks = numblks[numblkscod];
}
acmod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3); /* acmod */
lfe_on = gst_bit_reader_get_bits_uint8_unchecked (&bits, 1); /* lfeon */
gst_bit_reader_skip_unchecked (&bits, 5); /* bsid */
if (frame_size)
*frame_size = (frmsiz + 1) * 2;
if (rate)
*rate = sample_rate;
if (chans)
*chans = acmod_chans[acmod] + lfe_on;
if (blks)
*blks = blocks;
if (sid)
*sid = (strmtyp & 0x1) << 3 | strmid;
ret = TRUE;
cleanup:
gst_buffer_unmap (buf, &map);
return ret;
}
static GstFlowReturn
gst_amc_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
{
GstAmcAudioDec *self;
gint idx;
GstAmcBuffer *buf;
GstAmcBufferInfo buffer_info;
guint offset = 0;
GstClockTime timestamp, duration, timestamp_offset = 0;
GstMapInfo minfo;
GError *err = NULL;
memset (&minfo, 0, sizeof (minfo));
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Handling frame");
/* Make sure to keep a reference to the input here,
* it can be unreffed from the other thread if
* finish_frame() is called */
if (inbuf)
inbuf = gst_buffer_ref (inbuf);
if (!self->started) {
GST_ERROR_OBJECT (self, "Codec not started yet");
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_NOT_NEGOTIATED;
}
if (self->flushing)
goto flushing;
if (self->downstream_flow_ret != GST_FLOW_OK)
goto downstream_error;
if (!inbuf)
return gst_amc_audio_dec_drain (self);
timestamp = GST_BUFFER_PTS (inbuf);
duration = GST_BUFFER_DURATION (inbuf);
gst_buffer_map (inbuf, &minfo, GST_MAP_READ);
while (offset < minfo.size) {
/* Make sure to release the base class stream lock, otherwise
* _loop() can't call _finish_frame() and we might block forever
* because no input buffers are released */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
/* Wait at most 100ms here, some codecs don't fail dequeueing if
* the codec is flushing, causing deadlocks during shutdown */
idx = gst_amc_codec_dequeue_input_buffer (self->codec, 100000, &err);
GST_AUDIO_DECODER_STREAM_LOCK (self);
if (idx < 0) {
if (self->flushing || self->downstream_flow_ret == GST_FLOW_FLUSHING) {
g_clear_error (&err);
goto flushing;
}
switch (idx) {
case INFO_TRY_AGAIN_LATER:
GST_DEBUG_OBJECT (self, "Dequeueing input buffer timed out");
continue; /* next try */
break;
case G_MININT:
GST_ERROR_OBJECT (self, "Failed to dequeue input buffer");
goto dequeue_error;
default:
g_assert_not_reached ();
break;
}
continue;
}
if (self->flushing) {
memset (&buffer_info, 0, sizeof (buffer_info));
gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info, NULL);
goto flushing;
}
if (self->downstream_flow_ret != GST_FLOW_OK) {
memset (&buffer_info, 0, sizeof (buffer_info));
gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info, &err);
if (err && !self->flushing)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_clear_error (&err);
goto downstream_error;
}
/* Now handle the frame */
/* Copy the buffer content in chunks of size as requested
* by the port */
buf = gst_amc_codec_get_input_buffer (self->codec, idx, &err);
if (!buf)
goto failed_to_get_input_buffer;
memset (&buffer_info, 0, sizeof (buffer_info));
buffer_info.offset = 0;
buffer_info.size = MIN (minfo.size - offset, buf->size);
gst_amc_buffer_set_position_and_limit (buf, NULL, buffer_info.offset,
buffer_info.size);
orc_memcpy (buf->data, minfo.data + offset, buffer_info.size);
gst_amc_buffer_free (buf);
buf = NULL;
/* Interpolate timestamps if we're passing the buffer
* in multiple chunks */
if (offset != 0 && duration != GST_CLOCK_TIME_NONE) {
timestamp_offset = gst_util_uint64_scale (offset, duration, minfo.size);
}
if (timestamp != GST_CLOCK_TIME_NONE) {
buffer_info.presentation_time_us =
gst_util_uint64_scale (timestamp + timestamp_offset, 1, GST_USECOND);
self->last_upstream_ts = timestamp + timestamp_offset;
}
if (duration != GST_CLOCK_TIME_NONE)
self->last_upstream_ts += duration;
if (offset == 0) {
if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DELTA_UNIT))
buffer_info.flags |= BUFFER_FLAG_SYNC_FRAME;
}
offset += buffer_info.size;
GST_DEBUG_OBJECT (self,
"Queueing buffer %d: size %d time %" G_GINT64_FORMAT " flags 0x%08x",
idx, buffer_info.size, buffer_info.presentation_time_us,
buffer_info.flags);
if (!gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info,
&err)) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
goto queue_error;
}
self->drained = FALSE;
}
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
return self->downstream_flow_ret;
downstream_error:
{
GST_ERROR_OBJECT (self, "Downstream returned %s",
gst_flow_get_name (self->downstream_flow_ret));
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return self->downstream_flow_ret;
}
failed_to_get_input_buffer:
{
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
dequeue_error:
{
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
queue_error:
{
GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, err);
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING");
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_FLUSHING;
}
}
static GstFlowReturn
gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstFlowReturn ret;
GstRtpCELTPay *rtpceltpay;
gsize payload_len;
GstMapInfo map;
GstClockTime duration, packet_dur;
guint i, ssize, packet_len;
rtpceltpay = GST_RTP_CELT_PAY (basepayload);
ret = GST_FLOW_OK;
gst_buffer_map (buffer, &map, GST_MAP_READ);
switch (rtpceltpay->packet) {
case 0:
/* ident packet. We need to parse the headers to construct the RTP
* properties. */
if (!gst_rtp_celt_pay_parse_ident (rtpceltpay, map.data, map.size))
goto parse_error;
goto cleanup;
case 1:
/* comment packet, we ignore it */
goto cleanup;
default:
/* other packets go in the payload */
break;
}
gst_buffer_unmap (buffer, &map);
duration = GST_BUFFER_DURATION (buffer);
GST_LOG_OBJECT (rtpceltpay,
"got buffer of duration %" GST_TIME_FORMAT ", size %" G_GSIZE_FORMAT,
GST_TIME_ARGS (duration), map.size);
/* calculate the size of the size field and the payload */
ssize = 1;
for (i = map.size; i > 0xff; i -= 0xff)
ssize++;
GST_DEBUG_OBJECT (rtpceltpay, "bytes for size %u", ssize);
/* calculate what the new size and duration would be of the packet */
payload_len = ssize + map.size + rtpceltpay->bytes + rtpceltpay->sbytes;
if (rtpceltpay->qduration != -1 && duration != -1)
packet_dur = rtpceltpay->qduration + duration;
else
packet_dur = 0;
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
if (gst_rtp_base_payload_is_filled (basepayload, packet_len, packet_dur)) {
/* size or duration would overflow the packet, flush the queued data */
ret = gst_rtp_celt_pay_flush_queued (rtpceltpay);
}
/* queue the packet */
gst_rtp_celt_pay_add_queued (rtpceltpay, buffer, ssize, map.size, duration);
done:
rtpceltpay->packet++;
return ret;
/* ERRORS */
cleanup:
{
gst_buffer_unmap (buffer, &map);
goto done;
}
parse_error:
{
GST_ELEMENT_ERROR (rtpceltpay, STREAM, DECODE, (NULL),
("Error parsing first identification packet."));
gst_buffer_unmap (buffer, &map);
return GST_FLOW_ERROR;
}
}
static void
gst_amc_audio_dec_loop (GstAmcAudioDec * self)
{
GstFlowReturn flow_ret = GST_FLOW_OK;
gboolean is_eos;
GstAmcBuffer *buf;
GstAmcBufferInfo buffer_info;
gint idx;
GError *err = NULL;
GST_AUDIO_DECODER_STREAM_LOCK (self);
retry:
/*if (self->input_caps_changed) {
idx = INFO_OUTPUT_FORMAT_CHANGED;
} else { */
GST_DEBUG_OBJECT (self, "Waiting for available output buffer");
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
/* Wait at most 100ms here, some codecs don't fail dequeueing if
* the codec is flushing, causing deadlocks during shutdown */
idx =
gst_amc_codec_dequeue_output_buffer (self->codec, &buffer_info, 100000,
&err);
GST_AUDIO_DECODER_STREAM_LOCK (self);
/*} */
if (idx < 0) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
switch (idx) {
case INFO_OUTPUT_BUFFERS_CHANGED:
/* Handled internally */
g_assert_not_reached ();
break;
case INFO_OUTPUT_FORMAT_CHANGED:{
GstAmcFormat *format;
gchar *format_string;
GST_DEBUG_OBJECT (self, "Output format has changed");
format = gst_amc_codec_get_output_format (self->codec, &err);
if (!format)
goto format_error;
format_string = gst_amc_format_to_string (format, &err);
if (err) {
gst_amc_format_free (format);
goto format_error;
}
GST_DEBUG_OBJECT (self, "Got new output format: %s", format_string);
g_free (format_string);
if (!gst_amc_audio_dec_set_src_caps (self, format)) {
gst_amc_format_free (format);
goto format_error;
}
gst_amc_format_free (format);
goto retry;
}
case INFO_TRY_AGAIN_LATER:
GST_DEBUG_OBJECT (self, "Dequeueing output buffer timed out");
goto retry;
case G_MININT:
GST_ERROR_OBJECT (self, "Failure dequeueing output buffer");
goto dequeue_error;
default:
g_assert_not_reached ();
break;
}
goto retry;
}
GST_DEBUG_OBJECT (self,
"Got output buffer at index %d: offset %d size %d time %" G_GINT64_FORMAT
" flags 0x%08x", idx, buffer_info.offset, buffer_info.size,
buffer_info.presentation_time_us, buffer_info.flags);
is_eos = ! !(buffer_info.flags & BUFFER_FLAG_END_OF_STREAM);
buf = gst_amc_codec_get_output_buffer (self->codec, idx, &err);
if (!buf)
goto failed_to_get_output_buffer;
if (buffer_info.size > 0) {
GstBuffer *outbuf;
GstMapInfo minfo;
/* This sometimes happens at EOS or if the input is not properly framed,
* let's handle it gracefully by allocating a new buffer for the current
* caps and filling it
*/
if (buffer_info.size % self->info.bpf != 0)
goto invalid_buffer_size;
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self),
buffer_info.size);
if (!outbuf)
goto failed_allocate;
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->data + buffer_info.offset);
n_samples = buffer_info.size / 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 {
orc_memcpy (minfo.data, buf->data + buffer_info.offset, buffer_info.size);
}
gst_buffer_unmap (outbuf, &minfo);
if (self->spf != -1) {
gst_adapter_push (self->output_adapter, outbuf);
} else {
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, 1);
}
}
gst_amc_buffer_free (buf);
buf = NULL;
if (self->spf != -1) {
GstBuffer *outbuf;
guint avail = gst_adapter_available (self->output_adapter);
guint nframes;
/* On EOS we take the complete adapter content, no matter
* if it is a multiple of the codec frame size or not.
* Otherwise we take a multiple of codec frames and push
* them downstream
*/
avail /= self->info.bpf;
if (!is_eos) {
nframes = avail / self->spf;
avail = nframes * self->spf;
} else {
nframes = (avail + self->spf - 1) / self->spf;
}
avail *= self->info.bpf;
if (avail > 0) {
outbuf = gst_adapter_take_buffer (self->output_adapter, avail);
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf,
nframes);
}
}
if (!gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err)) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
goto failed_release;
}
if (is_eos || flow_ret == GST_FLOW_EOS) {
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
if (self->draining) {
GST_DEBUG_OBJECT (self, "Drained");
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
} else if (flow_ret == GST_FLOW_OK) {
GST_DEBUG_OBJECT (self, "Component signalled EOS");
flow_ret = GST_FLOW_EOS;
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_DECODER_STREAM_LOCK (self);
} else {
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
}
self->downstream_flow_ret = flow_ret;
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
dequeue_error:
{
GST_ELEMENT_ERROR_FROM_ERROR (self, 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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
format_error:
{
if (err)
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
else
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Failed to handle format"));
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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
failed_release:
{
GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, 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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
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;
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));
} 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));
} else if (flow_ret == GST_FLOW_FLUSHING) {
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
}
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
failed_to_get_output_buffer:
{
GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, 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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
invalid_buffer_size:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Invalid buffer size %u (bfp %d)", buffer_info.size, self->info.bpf));
gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err);
if (err && !self->flushing)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_clear_error (&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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
failed_allocate:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to allocate output buffer"));
gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err);
if (err && !self->flushing)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_clear_error (&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;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
}
void ofGstVideoUtils::update(){
if (isLoaded()){
if(!isFrameByFrame()){
mutex->lock();
bHavePixelsChanged = bBackPixelsChanged;
if (bHavePixelsChanged){
bBackPixelsChanged=false;
math::Swap(pixels.imageData,backPixels.imageData);
//pixels.swap(backPixels);
#if GST_VERSION_MAJOR==0
if(prevBuffer) gst_buffer_unref (prevBuffer);
#else
if(prevBuffer) gst_sample_unref (prevBuffer);
#endif
prevBuffer = buffer;
}
mutex->unlock();
}else{
#if GST_VERSION_MAJOR==0
GstBuffer *buffer;
//get the buffer from appsink
if(isPaused()) buffer = gst_app_sink_pull_preroll (GST_APP_SINK (getSink()));
else buffer = gst_app_sink_pull_buffer (GST_APP_SINK (getSink()));
if(buffer){
if(pixels.imageDataSize!=0){
if(prevBuffer) gst_buffer_unref (prevBuffer);
pixels.setData(GST_BUFFER_DATA (buffer),pixels.Size,pixels.format);
prevBuffer = buffer;
bHavePixelsChanged=true;
}
}
}
#else
GstBuffer *buffer;
GstSample * sample;
//get the buffer from appsink
if(isPaused()){
sample = gst_app_sink_pull_preroll (GST_APP_SINK (getSink()));
}else{
sample = gst_app_sink_pull_sample (GST_APP_SINK (getSink()));
}
buffer = gst_sample_get_buffer(sample);
if(buffer){
if(pixels.isAllocated()){
if(prevBuffer) gst_sample_unref (prevBuffer);
gst_buffer_map (buffer, &mapinfo, GST_MAP_READ);
//TODO: stride = mapinfo.size / height;
pixels.setFromExternalPixels(mapinfo.data,pixels.getWidth(),pixels.getHeight(),pixels.getNumChannels());
prevBuffer = sample;
bHavePixelsChanged=true;
gst_buffer_unmap(buffer,&mapinfo);
}
}
}
#endif
}else{
static void
vorbis_parse_push_headers (GstVorbisParse * parse)
{
/* mark and put on caps */
GstCaps *caps;
GstBuffer *outbuf, *outbuf1, *outbuf2, *outbuf3;
ogg_packet packet;
GstMapInfo map;
outbuf = GST_BUFFER_CAST (parse->streamheader->data);
gst_buffer_map (outbuf, &map, GST_MAP_READ);
packet.packet = map.data;
packet.bytes = map.size;
packet.granulepos = GST_BUFFER_OFFSET_END (outbuf);
packet.packetno = 1;
packet.e_o_s = 0;
packet.b_o_s = 1;
vorbis_synthesis_headerin (&parse->vi, &parse->vc, &packet);
gst_buffer_unmap (outbuf, &map);
parse->sample_rate = parse->vi.rate;
parse->channels = parse->vi.channels;
outbuf1 = outbuf;
outbuf = GST_BUFFER_CAST (parse->streamheader->next->data);
gst_buffer_map (outbuf, &map, GST_MAP_READ);
packet.packet = map.data;
packet.bytes = map.size;
packet.granulepos = GST_BUFFER_OFFSET_END (outbuf);
packet.packetno = 2;
packet.e_o_s = 0;
packet.b_o_s = 0;
vorbis_synthesis_headerin (&parse->vi, &parse->vc, &packet);
gst_buffer_unmap (outbuf, &map);
outbuf2 = outbuf;
outbuf = GST_BUFFER_CAST (parse->streamheader->next->next->data);
gst_buffer_map (outbuf, &map, GST_MAP_READ);
packet.packet = map.data;
packet.bytes = map.size;
packet.granulepos = GST_BUFFER_OFFSET_END (outbuf);
packet.packetno = 3;
packet.e_o_s = 0;
packet.b_o_s = 0;
vorbis_synthesis_headerin (&parse->vi, &parse->vc, &packet);
gst_buffer_unmap (outbuf, &map);
outbuf3 = outbuf;
/* get the headers into the caps, passing them to vorbis as we go */
caps = gst_caps_new_simple ("audio/x-vorbis",
"rate", G_TYPE_INT, parse->sample_rate,
"channels", G_TYPE_INT, parse->channels, NULL);;
vorbis_parse_set_header_on_caps (parse, caps);
GST_DEBUG_OBJECT (parse, "here are the caps: %" GST_PTR_FORMAT, caps);
gst_pad_set_caps (parse->srcpad, caps);
gst_caps_unref (caps);
/* first process queued events */
vorbis_parse_drain_event_queue (parse);
/* push out buffers, ignoring return value... */
gst_pad_push (parse->srcpad, outbuf1);
gst_pad_push (parse->srcpad, outbuf2);
gst_pad_push (parse->srcpad, outbuf3);
g_list_free (parse->streamheader);
parse->streamheader = NULL;
}
