static void
mpegts_base_loop (MpegTSBase * base)
{
GstFlowReturn ret = GST_FLOW_ERROR;
switch (base->mode) {
case BASE_MODE_SCANNING:
/* Find first sync point */
ret = mpegts_base_scan (base);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto error;
base->mode = BASE_MODE_STREAMING;
GST_DEBUG ("Changing to Streaming");
break;
case BASE_MODE_SEEKING:
/* FIXME : unclear if we still need mode_seeking... */
base->mode = BASE_MODE_STREAMING;
break;
case BASE_MODE_STREAMING:
{
GstBuffer *buf = NULL;
GST_DEBUG ("Pulling data from %" G_GUINT64_FORMAT, base->seek_offset);
ret = gst_pad_pull_range (base->sinkpad, base->seek_offset,
100 * base->packetsize, &buf);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto error;
base->seek_offset += gst_buffer_get_size (buf);
ret = mpegts_base_chain (base->sinkpad, GST_OBJECT_CAST (base), buf);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto error;
}
break;
case BASE_MODE_PUSHING:
GST_WARNING ("wrong BASE_MODE_PUSHING mode in pull loop");
break;
}
return;
error:
{
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (base, "Pausing task, reason %s", reason);
if (ret == GST_FLOW_EOS) {
if (!GST_MPEGTS_BASE_GET_CLASS (base)->push_event (base,
gst_event_new_eos ()))
GST_ELEMENT_ERROR (base, STREAM, FAILED,
(_("Internal data stream error.")),
("No program activated before EOS"));
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (base, STREAM, FAILED,
(_("Internal data stream error.")),
("stream stopped, reason %s", reason));
GST_MPEGTS_BASE_GET_CLASS (base)->push_event (base, gst_event_new_eos ());
}
gst_pad_pause_task (base->sinkpad);
}
}
/**
* hls_progress_buffer_loop()
*
* Primary function for push-mode. Pulls data from progressbuffer's cache queue.
*/
static void hls_progress_buffer_loop(void *data)
{
HLSProgressBuffer* element = HLS_PROGRESS_BUFFER(data);
GstFlowReturn result = GST_FLOW_OK;
g_mutex_lock(element->lock);
while (element->srcresult == GST_FLOW_OK && !cache_has_enough_data(element->cache[element->cache_read_index]))
{
if (element->is_eos)
{
gst_pad_push_event(element->srcpad, gst_event_new_eos());
element->srcresult = GST_FLOW_WRONG_STATE;
break;
}
if (!element->is_eos)
{
g_cond_wait(element->add_cond, element->lock);
}
}
result = element->srcresult;
if (result == GST_FLOW_OK)
{
GstBuffer *buffer = NULL;
guint64 read_position = cache_read_buffer(element->cache[element->cache_read_index], &buffer);
if (read_position == element->cache_size[element->cache_read_index])
{
element->cache_write_ready[element->cache_read_index] = TRUE;
element->cache_read_index = (element->cache_read_index + 1) % NUM_OF_CACHED_SEGMENTS;
send_hls_not_full_message(element);
g_cond_signal(element->del_cond);
}
g_mutex_unlock(element->lock);
gst_buffer_set_caps(buffer, GST_PAD_CAPS(element->sinkpad));
// Send the data to the hls progressbuffer source pad
result = gst_pad_push(element->srcpad, buffer);
g_mutex_lock(element->lock);
if (GST_FLOW_OK == element->srcresult || GST_FLOW_OK != result)
element->srcresult = result;
else
result = element->srcresult;
g_mutex_unlock(element->lock);
}
else
{
g_mutex_unlock(element->lock);
}
if (result != GST_FLOW_OK && !element->is_flushing)
gst_pad_pause_task(element->srcpad);
}
static void
user_read_data (png_structp png_ptr, png_bytep data, png_size_t length)
{
GstPngDec *pngdec;
GstBuffer *buffer;
GstFlowReturn ret = GST_FLOW_OK;
guint size;
pngdec = GST_PNGDEC (png_get_io_ptr (png_ptr));
GST_LOG ("reading %" G_GSIZE_FORMAT " bytes of data at offset %d", length,
pngdec->offset);
ret = gst_pad_pull_range (pngdec->sinkpad, pngdec->offset, length, &buffer);
if (ret != GST_FLOW_OK)
goto pause;
size = GST_BUFFER_SIZE (buffer);
if (size != length)
goto short_buffer;
memcpy (data, GST_BUFFER_DATA (buffer), size);
gst_buffer_unref (buffer);
pngdec->offset += length;
return;
/* ERRORS */
pause:
{
GST_INFO_OBJECT (pngdec, "pausing task, reason %s",
gst_flow_get_name (ret));
gst_pad_pause_task (pngdec->sinkpad);
if (ret == GST_FLOW_UNEXPECTED) {
gst_pad_push_event (pngdec->srcpad, gst_event_new_eos ());
} else if (ret < GST_FLOW_UNEXPECTED || ret == GST_FLOW_NOT_LINKED) {
GST_ELEMENT_ERROR (pngdec, STREAM, FAILED,
(_("Internal data stream error.")),
("stream stopped, reason %s", gst_flow_get_name (ret)));
gst_pad_push_event (pngdec->srcpad, gst_event_new_eos ());
}
png_error (png_ptr, "Internal data stream error.");
return;
}
short_buffer:
{
gst_buffer_unref (buffer);
GST_ELEMENT_ERROR (pngdec, STREAM, FAILED,
(_("Internal data stream error.")),
("Read %u, needed %" G_GSIZE_FORMAT "bytes", size, length));
ret = GST_FLOW_ERROR;
goto pause;
}
}
static gboolean
gst_tcp_mix_src_pad_pause (GstTCPMixSrcPad * pad, GstTaskFunction func)
{
gboolean res;
if (GST_PAD_TASK (pad) != NULL) { // FIXME: pad->status == started
res = gst_pad_pause_task (GST_PAD (pad));
}
return res;
}
static gboolean
gst_rnd_buffer_size_src_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRndBufferSize *self;
GstSeekType start_type;
GstSeekFlags flags;
GstFormat format;
gint64 start;
if (GST_EVENT_TYPE (event) != GST_EVENT_SEEK) {
GST_WARNING_OBJECT (pad, "dropping %s event", GST_EVENT_TYPE_NAME (event));
return FALSE;
}
self = GST_RND_BUFFER_SIZE (parent);
gst_event_parse_seek (event, NULL, &format, &flags, &start_type, &start,
NULL, NULL);
if (format != GST_FORMAT_BYTES) {
GST_WARNING_OBJECT (pad, "only BYTE format supported");
return FALSE;
}
if (start_type != GST_SEEK_TYPE_SET) {
GST_WARNING_OBJECT (pad, "only SEEK_TYPE_SET supported");
return FALSE;
}
if ((flags & GST_SEEK_FLAG_FLUSH)) {
gst_pad_push_event (self->srcpad, gst_event_new_flush_start ());
gst_pad_push_event (self->sinkpad, gst_event_new_flush_start ());
} else {
gst_pad_pause_task (self->sinkpad);
}
GST_PAD_STREAM_LOCK (self->sinkpad);
if ((flags & GST_SEEK_FLAG_FLUSH)) {
gst_pad_push_event (self->srcpad, gst_event_new_flush_stop (TRUE));
gst_pad_push_event (self->sinkpad, gst_event_new_flush_stop (TRUE));
}
GST_INFO_OBJECT (pad, "seeking to offset %" G_GINT64_FORMAT, start);
self->offset = start;
self->need_newsegment = TRUE;
gst_pad_start_task (self->sinkpad, (GstTaskFunction) gst_rnd_buffer_size_loop,
self);
GST_PAD_STREAM_UNLOCK (self->sinkpad);
return TRUE;
}
static void
gst_vaapiencode_buffer_loop (GstVaapiEncode * encode)
{
GstFlowReturn ret;
const gint64 timeout = 50000; /* microseconds */
ret = gst_vaapiencode_push_frame (encode, timeout);
if (ret == GST_FLOW_OK || ret == GST_VAAPI_ENCODE_FLOW_TIMEOUT)
return;
gst_pad_pause_task (GST_VAAPI_PLUGIN_BASE_SRC_PAD (encode));
}
/* called repeadedly with @pad as the source pad. This function should push out
* data to the peer element. */
static void
gst_ts_shifter_loop (GstPad * pad)
{
GstTSShifter *ts;
GstFlowReturn ret;
ts = GST_TS_SHIFTER (GST_PAD_PARENT (pad));
/* have to lock for thread-safety */
FLOW_MUTEX_LOCK_CHECK (ts, ts->srcresult, out_flushing);
if (gst_ts_cache_is_empty (ts->cache) && !ts->is_eos) {
GST_CAT_LOG_OBJECT (ts_flow, ts, "empty, waiting for new data");
do {
/* Wait for data to be available, we could be unlocked because of a flush. */
FLOW_WAIT_ADD_CHECK (ts, ts->srcresult, out_flushing);
}
while (gst_ts_cache_is_empty (ts->cache) && !ts->is_eos);
}
ret = gst_ts_shifter_pop (ts);
ts->srcresult = ret;
if (ret != GST_FLOW_OK)
goto out_flushing;
FLOW_MUTEX_UNLOCK (ts);
return;
/* ERRORS */
out_flushing:
{
gboolean eos = ts->is_eos;
GstFlowReturn ret = ts->srcresult;
gst_pad_pause_task (ts->srcpad);
FLOW_MUTEX_UNLOCK (ts);
GST_CAT_LOG_OBJECT (ts_flow, ts,
"pause task, reason: %s", gst_flow_get_name (ts->srcresult));
/* let app know about us giving up if upstream is not expected to do so */
/* UNEXPECTED is already taken care of elsewhere */
if (eos && (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS)) {
GST_ELEMENT_ERROR (ts, STREAM, FAILED,
("Internal data flow error."),
("streaming task paused, reason %s (%d)",
gst_flow_get_name (ret), ret));
GST_CAT_LOG_OBJECT (ts_flow, ts, "pushing EOS");
gst_pad_push_event (ts->srcpad, gst_event_new_eos ());
}
return;
}
}
static gboolean
gst_type_find_element_activate_src_mode (GstPad * pad, GstObject * parent,
GstPadMode mode, gboolean active)
{
gboolean res;
GstTypeFindElement *typefind;
typefind = GST_TYPE_FIND_ELEMENT (parent);
switch (mode) {
case GST_PAD_MODE_PULL:
/* make sure our task stops pushing, we can't call _stop here because this
* activation might happen from the streaming thread. */
gst_pad_pause_task (typefind->sink);
res = gst_pad_activate_mode (typefind->sink, mode, active);
break;
default:
res = TRUE;
break;
}
return res;
}
static void
gst_tta_parse_loop (GstTtaParse * ttaparse)
{
GstFlowReturn ret;
if (!ttaparse->header_parsed)
if ((ret = gst_tta_parse_parse_header (ttaparse)) != GST_FLOW_OK)
goto pause;
if ((ret = gst_tta_parse_stream_data (ttaparse)) != GST_FLOW_OK)
goto pause;
return;
pause:
GST_LOG_OBJECT (ttaparse, "pausing task, %s", gst_flow_get_name (ret));
gst_pad_pause_task (ttaparse->sinkpad);
if (ret == GST_FLOW_UNEXPECTED) {
gst_pad_push_event (ttaparse->srcpad, gst_event_new_eos ());
} else if (ret < GST_FLOW_UNEXPECTED || ret == GST_FLOW_NOT_LINKED) {
GST_ELEMENT_FLOW_ERROR (ttaparse, ret);
gst_pad_push_event (ttaparse->srcpad, gst_event_new_eos ());
}
}
static GstFlowReturn
gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstRTPDTMFSrcEvent *event;
GstRTPDTMFSrc *dtmfsrc;
GstClock *clock;
GstClockID *clockid;
GstClockReturn clockret;
GstMessage *message;
GQueue messages = G_QUEUE_INIT;
dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
do {
if (dtmfsrc->payload == NULL) {
GST_DEBUG_OBJECT (dtmfsrc, "popping");
event = g_async_queue_pop (dtmfsrc->event_queue);
GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
switch (event->event_type) {
case RTP_DTMF_EVENT_TYPE_STOP:
GST_WARNING_OBJECT (dtmfsrc,
"Received a DTMF stop event when already stopped");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case RTP_DTMF_EVENT_TYPE_START:
dtmfsrc->first_packet = TRUE;
dtmfsrc->last_packet = FALSE;
/* Set the redundancy on the first packet */
dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
if (!gst_rtp_dtmf_prepare_timestamps (dtmfsrc))
goto no_clock;
g_queue_push_tail (&messages,
gst_dtmf_src_prepare_message (dtmfsrc, "dtmf-event-processed",
event));
dtmfsrc->payload = event->payload;
dtmfsrc->payload->duration =
dtmfsrc->ptime * dtmfsrc->clock_rate / 1000;
event->payload = NULL;
break;
case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed
*/
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
gst_rtp_dtmf_src_event_free (event);
} else if (!dtmfsrc->first_packet && !dtmfsrc->last_packet &&
(dtmfsrc->timestamp - dtmfsrc->start_timestamp) / GST_MSECOND >=
MIN_PULSE_DURATION) {
GST_DEBUG_OBJECT (dtmfsrc, "try popping");
event = g_async_queue_try_pop (dtmfsrc->event_queue);
if (event != NULL) {
GST_DEBUG_OBJECT (dtmfsrc, "try popped %d", event->event_type);
switch (event->event_type) {
case RTP_DTMF_EVENT_TYPE_START:
GST_WARNING_OBJECT (dtmfsrc,
"Received two consecutive DTMF start events");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case RTP_DTMF_EVENT_TYPE_STOP:
dtmfsrc->first_packet = FALSE;
dtmfsrc->last_packet = TRUE;
/* Set the redundancy on the last packet */
dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
g_queue_push_tail (&messages,
gst_dtmf_src_prepare_message (dtmfsrc, "dtmf-event-processed",
event));
break;
case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed)
*/
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
gst_rtp_dtmf_src_event_free (event);
}
}
} while (dtmfsrc->payload == NULL);
GST_DEBUG_OBJECT (dtmfsrc, "Processed events, now lets wait on the clock");
clock = gst_element_get_clock (GST_ELEMENT (basesrc));
if (!clock)
goto no_clock;
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
gst_object_unref (clock);
GST_OBJECT_LOCK (dtmfsrc);
if (!dtmfsrc->paused) {
dtmfsrc->clockid = clockid;
GST_OBJECT_UNLOCK (dtmfsrc);
clockret = gst_clock_id_wait (clockid, NULL);
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused)
clockret = GST_CLOCK_UNSCHEDULED;
} else {
clockret = GST_CLOCK_UNSCHEDULED;
}
gst_clock_id_unref (clockid);
dtmfsrc->clockid = NULL;
GST_OBJECT_UNLOCK (dtmfsrc);
while ((message = g_queue_pop_head (&messages)) != NULL)
gst_element_post_message (GST_ELEMENT (dtmfsrc), message);
if (clockret == GST_CLOCK_UNSCHEDULED) {
goto paused;
}
send_last:
if (dtmfsrc->dirty)
if (!gst_rtp_dtmf_src_negotiate (basesrc))
return GST_FLOW_NOT_NEGOTIATED;
/* create buffer to hold the payload */
*buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
if (dtmfsrc->redundancy_count)
dtmfsrc->redundancy_count--;
/* Only the very first one has a marker */
dtmfsrc->first_packet = FALSE;
/* This is the end of the event */
if (dtmfsrc->last_packet == TRUE && dtmfsrc->redundancy_count == 0) {
g_slice_free (GstRTPDTMFPayload, dtmfsrc->payload);
dtmfsrc->payload = NULL;
dtmfsrc->last_packet = FALSE;
}
return GST_FLOW_OK;
paused_locked:
GST_OBJECT_UNLOCK (dtmfsrc);
paused:
if (dtmfsrc->payload) {
dtmfsrc->first_packet = FALSE;
dtmfsrc->last_packet = TRUE;
/* Set the redundanc on the last packet */
dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
goto send_last;
} else {
return GST_FLOW_FLUSHING;
}
no_clock:
GST_ELEMENT_ERROR (dtmfsrc, STREAM, MUX, ("No available clock"),
("No available clock"));
gst_pad_pause_task (GST_BASE_SRC_PAD (dtmfsrc));
return GST_FLOW_ERROR;
}
static void
gst_rnd_buffer_size_loop (GstRndBufferSize * self)
{
GstBuffer *buf = NULL;
GstFlowReturn ret;
guint num_bytes, size;
if (G_UNLIKELY (self->min > self->max))
goto bogus_minmax;
if (G_UNLIKELY (self->min != self->max)) {
num_bytes = g_rand_int_range (self->rand, self->min, self->max);
} else {
num_bytes = self->min;
}
GST_LOG_OBJECT (self, "pulling %u bytes at offset %" G_GUINT64_FORMAT,
num_bytes, self->offset);
ret = gst_pad_pull_range (self->sinkpad, self->offset, num_bytes, &buf);
if (ret != GST_FLOW_OK)
goto pull_failed;
size = gst_buffer_get_size (buf);
if (size < num_bytes) {
GST_WARNING_OBJECT (self, "short buffer: %u bytes", size);
}
if (self->need_newsegment) {
GstSegment segment;
gst_segment_init (&segment, GST_FORMAT_BYTES);
segment.start = self->offset;
gst_pad_push_event (self->srcpad, gst_event_new_segment (&segment));
self->need_newsegment = FALSE;
}
self->offset += size;
ret = gst_pad_push (self->srcpad, buf);
if (ret != GST_FLOW_OK)
goto push_failed;
return;
pause_task:
{
GST_DEBUG_OBJECT (self, "pausing task");
gst_pad_pause_task (self->sinkpad);
return;
}
pull_failed:
{
if (ret == GST_FLOW_EOS) {
GST_DEBUG_OBJECT (self, "eos");
gst_pad_push_event (self->srcpad, gst_event_new_eos ());
} else {
GST_WARNING_OBJECT (self, "pull_range flow: %s", gst_flow_get_name (ret));
}
goto pause_task;
}
push_failed:
{
GST_DEBUG_OBJECT (self, "push flow: %s", gst_flow_get_name (ret));
if (ret == GST_FLOW_EOS) {
GST_DEBUG_OBJECT (self, "eos");
gst_pad_push_event (self->srcpad, gst_event_new_eos ());
} else if (ret < GST_FLOW_EOS || ret == GST_FLOW_NOT_LINKED) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
("Internal data stream error."),
("streaming stopped, reason: %s", gst_flow_get_name (ret)));
}
goto pause_task;
}
bogus_minmax:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS,
("The minimum buffer size is smaller than the maximum buffer size."),
("buffer sizes: max=%d, min=%d", self->min, self->max));
goto pause_task;
}
}
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, &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, &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, &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;
}
}
static void
gst_asf_parse_loop (GstPad * pad)
{
GstFlowReturn ret = GST_FLOW_OK;
GstAsfParse *asfparse = GST_ASF_PARSE_CAST (GST_OBJECT_PARENT (pad));
GST_LOG_OBJECT (asfparse, "Processing data in loop function");
switch (asfparse->parse_state) {
case ASF_PARSING_HEADERS:
GST_INFO_OBJECT (asfparse, "Starting to parse headers");
ret = gst_asf_parse_pull_headers (asfparse);
if (ret != GST_FLOW_OK)
goto pause;
asfparse->parse_state = ASF_PARSING_DATA;
case ASF_PARSING_DATA:
GST_INFO_OBJECT (asfparse, "Parsing data object headers");
ret = gst_asf_parse_pull_data_header (asfparse);
if (ret != GST_FLOW_OK)
goto pause;
asfparse->parse_state = ASF_PARSING_PACKETS;
case ASF_PARSING_PACKETS:
GST_INFO_OBJECT (asfparse, "Starting packet parsing");
GST_INFO_OBJECT (asfparse, "Broadcast mode %s",
asfparse->asfinfo->broadcast ? "on" : "off");
ret = gst_asf_parse_pull_packets (asfparse);
if (ret != GST_FLOW_OK)
goto pause;
/* test if all packets have been processed */
if (!asfparse->asfinfo->broadcast &&
asfparse->parsed_packets == asfparse->asfinfo->packets_count) {
GST_INFO_OBJECT (asfparse,
"All %" G_GUINT64_FORMAT " packets processed",
asfparse->parsed_packets);
asfparse->parse_state = ASF_PARSING_INDEXES;
}
case ASF_PARSING_INDEXES:
/* we currently don't care about indexes, so just push them forward */
GST_INFO_OBJECT (asfparse, "Starting indexes parsing");
ret = gst_asf_parse_pull_indexes (asfparse);
if (ret != GST_FLOW_OK)
goto pause;
default:
break;
}
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_INFO_OBJECT (asfparse, "Pausing sinkpad task");
gst_pad_pause_task (pad);
if (ret == GST_FLOW_EOS) {
gst_pad_push_event (asfparse->srcpad, gst_event_new_eos ());
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (asfparse, STREAM, FAILED,
(NULL), ("streaming task paused, reason %s (%d)", reason, ret));
gst_pad_push_event (asfparse->srcpad, gst_event_new_eos ());
}
}
}
static void
gst_type_find_element_loop (GstPad * pad)
{
GstTypeFindElement *typefind;
GstFlowReturn ret = GST_FLOW_OK;
typefind = GST_TYPE_FIND_ELEMENT (GST_PAD_PARENT (pad));
if (typefind->need_stream_start) {
gchar *stream_id;
GstEvent *event;
stream_id = gst_pad_create_stream_id (typefind->src,
GST_ELEMENT_CAST (typefind), NULL);
GST_DEBUG_OBJECT (typefind, "Pushing STREAM_START");
event = gst_event_new_stream_start (stream_id);
gst_event_set_group_id (event, gst_util_group_id_next ());
gst_pad_push_event (typefind->src, event);
typefind->need_stream_start = FALSE;
g_free (stream_id);
}
if (typefind->mode == MODE_TYPEFIND) {
GstPad *peer = NULL;
GstCaps *found_caps = NULL;
GstTypeFindProbability probability = GST_TYPE_FIND_NONE;
GST_DEBUG_OBJECT (typefind, "find type in pull mode");
GST_OBJECT_LOCK (typefind);
if (typefind->force_caps) {
found_caps = gst_caps_ref (typefind->force_caps);
probability = GST_TYPE_FIND_MAXIMUM;
}
GST_OBJECT_UNLOCK (typefind);
if (!found_caps) {
peer = gst_pad_get_peer (pad);
if (peer) {
gint64 size;
gchar *ext;
if (!gst_pad_query_duration (peer, GST_FORMAT_BYTES, &size)) {
GST_WARNING_OBJECT (typefind, "Could not query upstream length!");
gst_object_unref (peer);
ret = GST_FLOW_ERROR;
goto pause;
}
/* the size if 0, we cannot continue */
if (size == 0) {
/* keep message in sync with message in sink event handler */
GST_ELEMENT_ERROR (typefind, STREAM, TYPE_NOT_FOUND,
(_("Stream contains no data.")), ("Can't typefind empty stream"));
gst_object_unref (peer);
ret = GST_FLOW_ERROR;
goto pause;
}
ext = gst_type_find_get_extension (typefind, pad);
found_caps =
gst_type_find_helper_get_range (GST_OBJECT_CAST (peer),
GST_OBJECT_PARENT (peer),
(GstTypeFindHelperGetRangeFunction) (GST_PAD_GETRANGEFUNC (peer)),
(guint64) size, ext, &probability);
g_free (ext);
GST_DEBUG ("Found caps %" GST_PTR_FORMAT, found_caps);
gst_object_unref (peer);
}
}
if (!found_caps || probability < typefind->min_probability) {
GST_DEBUG ("Trying to guess using extension");
gst_caps_replace (&found_caps, NULL);
found_caps =
gst_type_find_guess_by_extension (typefind, pad, &probability);
}
if (!found_caps || probability < typefind->min_probability) {
GST_ELEMENT_ERROR (typefind, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
gst_caps_replace (&found_caps, NULL);
ret = GST_FLOW_ERROR;
goto pause;
}
GST_DEBUG ("Emiting found caps %" GST_PTR_FORMAT, found_caps);
gst_type_find_element_emit_have_type (typefind, probability, found_caps);
typefind->mode = MODE_NORMAL;
gst_caps_unref (found_caps);
} else if (typefind->mode == MODE_NORMAL) {
GstBuffer *outbuf = NULL;
if (typefind->need_segment) {
typefind->need_segment = FALSE;
gst_pad_push_event (typefind->src,
gst_event_new_segment (&typefind->segment));
}
/* Pull 4k blocks and send downstream */
ret = gst_pad_pull_range (typefind->sink, typefind->offset, 4096, &outbuf);
if (ret != GST_FLOW_OK)
goto pause;
typefind->offset += gst_buffer_get_size (outbuf);
ret = gst_pad_push (typefind->src, outbuf);
if (ret != GST_FLOW_OK)
goto pause;
} else {
/* Error out */
ret = GST_FLOW_ERROR;
goto pause;
}
return;
pause:
{
const gchar *reason = gst_flow_get_name (ret);
gboolean push_eos = FALSE;
GST_LOG_OBJECT (typefind, "pausing task, reason %s", reason);
gst_pad_pause_task (typefind->sink);
if (ret == GST_FLOW_EOS) {
/* perform EOS logic */
if (typefind->segment.flags & GST_SEGMENT_FLAG_SEGMENT) {
gint64 stop;
/* for segment playback we need to post when (in stream time)
* we stopped, this is either stop (when set) or the duration. */
if ((stop = typefind->segment.stop) == -1)
stop = typefind->offset;
GST_LOG_OBJECT (typefind, "Sending segment done, at end of segment");
gst_element_post_message (GST_ELEMENT (typefind),
gst_message_new_segment_done (GST_OBJECT (typefind),
GST_FORMAT_BYTES, stop));
gst_pad_push_event (typefind->src,
gst_event_new_segment_done (GST_FORMAT_BYTES, stop));
} else {
push_eos = TRUE;
}
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
/* for fatal errors we post an error message */
GST_ELEMENT_ERROR (typefind, STREAM, FAILED, (NULL),
("stream stopped, reason %s", reason));
push_eos = TRUE;
}
if (push_eos) {
/* send EOS, and prevent hanging if no streams yet */
GST_LOG_OBJECT (typefind, "Sending EOS, at end of stream");
gst_pad_push_event (typefind->src, gst_event_new_eos ());
}
return;
}
}
static gboolean
gst_type_find_element_seek (GstTypeFindElement * typefind, GstEvent * event)
{
GstSeekFlags flags;
GstSeekType start_type, stop_type;
GstFormat format;
gboolean flush;
gdouble rate;
gint64 start, stop;
GstSegment seeksegment = { 0, };
gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
&stop_type, &stop);
/* we can only seek on bytes */
if (format != GST_FORMAT_BYTES) {
GST_DEBUG_OBJECT (typefind, "Can only seek on BYTES");
return FALSE;
}
/* copy segment, we need this because we still need the old
* segment when we close the current segment. */
memcpy (&seeksegment, &typefind->segment, sizeof (GstSegment));
GST_DEBUG_OBJECT (typefind, "configuring seek");
gst_segment_do_seek (&seeksegment, rate, format, flags,
start_type, start, stop_type, stop, NULL);
flush = ! !(flags & GST_SEEK_FLAG_FLUSH);
GST_DEBUG_OBJECT (typefind, "New segment %" GST_SEGMENT_FORMAT, &seeksegment);
if (flush) {
GST_DEBUG_OBJECT (typefind, "Starting flush");
gst_pad_push_event (typefind->sink, gst_event_new_flush_start ());
gst_pad_push_event (typefind->src, gst_event_new_flush_start ());
} else {
GST_DEBUG_OBJECT (typefind, "Non-flushing seek, pausing task");
gst_pad_pause_task (typefind->sink);
}
/* now grab the stream lock so that streaming cannot continue, for
* non flushing seeks when the element is in PAUSED this could block
* forever. */
GST_DEBUG_OBJECT (typefind, "Waiting for streaming to stop");
GST_PAD_STREAM_LOCK (typefind->sink);
if (flush) {
GST_DEBUG_OBJECT (typefind, "Stopping flush");
gst_pad_push_event (typefind->sink, gst_event_new_flush_stop (TRUE));
gst_pad_push_event (typefind->src, gst_event_new_flush_stop (TRUE));
}
/* now update the real segment info */
GST_DEBUG_OBJECT (typefind, "Committing new seek segment");
memcpy (&typefind->segment, &seeksegment, sizeof (GstSegment));
typefind->offset = typefind->segment.start;
/* notify start of new segment */
if (typefind->segment.flags & GST_SEGMENT_FLAG_SEGMENT) {
GstMessage *msg;
msg = gst_message_new_segment_start (GST_OBJECT (typefind),
GST_FORMAT_BYTES, typefind->segment.start);
gst_element_post_message (GST_ELEMENT (typefind), msg);
}
typefind->need_segment = TRUE;
/* restart our task since it might have been stopped when we did the
* flush. */
gst_pad_start_task (typefind->sink,
(GstTaskFunction) gst_type_find_element_loop, typefind->sink, NULL);
/* streaming can continue now */
GST_PAD_STREAM_UNLOCK (typefind->sink);
return TRUE;
}
static void
gst_timidity_loop (GstPad * sinkpad)
{
GstTimidity *timidity = GST_TIMIDITY (GST_PAD_PARENT (sinkpad));
GstBuffer *out;
GstFlowReturn ret;
if (timidity->mididata_size == 0) {
if (!gst_timidity_get_upstream_size (timidity, &timidity->mididata_size)) {
GST_ELEMENT_ERROR (timidity, STREAM, DECODE, (NULL),
("Unable to get song length"));
goto paused;
}
if (timidity->mididata)
g_free (timidity->mididata);
timidity->mididata = g_malloc (timidity->mididata_size);
timidity->mididata_offset = 0;
return;
}
if (timidity->mididata_offset < timidity->mididata_size) {
GstBuffer *buffer;
gint64 size;
GST_DEBUG_OBJECT (timidity, "loading song");
ret =
gst_pad_pull_range (timidity->sinkpad, timidity->mididata_offset,
-1, &buffer);
if (ret != GST_FLOW_OK) {
GST_ELEMENT_ERROR (timidity, STREAM, DECODE, (NULL),
("Unable to load song"));
goto paused;
}
size = timidity->mididata_size - timidity->mididata_offset;
if (GST_BUFFER_SIZE (buffer) < size)
size = GST_BUFFER_SIZE (buffer);
memmove (timidity->mididata + timidity->mididata_offset,
GST_BUFFER_DATA (buffer), size);
gst_buffer_unref (buffer);
timidity->mididata_offset += size;
GST_DEBUG_OBJECT (timidity, "Song loaded");
return;
}
if (!timidity->song) {
MidIStream *stream;
GstTagList *tags = NULL;
gchar *text;
GST_DEBUG_OBJECT (timidity, "Parsing song");
stream =
mid_istream_open_mem (timidity->mididata, timidity->mididata_size, 0);
timidity->song = mid_song_load (stream, timidity->song_options);
mid_istream_close (stream);
if (!timidity->song) {
GST_ELEMENT_ERROR (timidity, STREAM, DECODE, (NULL),
("Unable to parse midi"));
goto paused;
}
mid_song_start (timidity->song);
timidity->o_len = (GST_MSECOND *
(GstClockTime) mid_song_get_total_time (timidity->song)) /
timidity->time_per_frame;
gst_segment_set_newsegment (timidity->o_segment, FALSE, 1.0,
GST_FORMAT_DEFAULT, 0, GST_CLOCK_TIME_NONE, 0);
gst_pad_push_event (timidity->srcpad,
gst_timidity_get_new_segment_event (timidity, GST_FORMAT_TIME, FALSE));
/* extract tags */
text = mid_song_get_meta (timidity->song, MID_SONG_TEXT);
if (text) {
tags = gst_tag_list_new ();
gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, text, NULL);
//g_free (text);
}
text = mid_song_get_meta (timidity->song, MID_SONG_COPYRIGHT);
if (text) {
if (tags == NULL)
tags = gst_tag_list_new ();
gst_tag_list_add (tags, GST_TAG_MERGE_APPEND,
GST_TAG_COPYRIGHT, text, NULL);
//g_free (text);
}
if (tags) {
gst_element_found_tags (GST_ELEMENT (timidity), tags);
}
GST_DEBUG_OBJECT (timidity, "Parsing song done");
return;
}
if (timidity->o_segment_changed) {
GstSegment *segment = gst_timidity_get_segment (timidity, GST_FORMAT_TIME,
!timidity->o_new_segment);
GST_LOG_OBJECT (timidity,
"sending newsegment from %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT
", pos=%" GST_TIME_FORMAT, GST_TIME_ARGS ((guint64) segment->start),
GST_TIME_ARGS ((guint64) segment->stop),
GST_TIME_ARGS ((guint64) segment->time));
if (timidity->o_segment->flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT (timidity),
gst_message_new_segment_start (GST_OBJECT (timidity),
segment->format, segment->start));
}
gst_segment_free (segment);
timidity->o_segment_changed = FALSE;
return;
}
if (timidity->o_seek) {
/* perform a seek internally */
timidity->o_segment->last_stop = timidity->o_segment->time;
mid_song_seek (timidity->song,
(timidity->o_segment->last_stop * timidity->time_per_frame) /
GST_MSECOND);
}
out = gst_timidity_get_buffer (timidity);
if (!out) {
GST_LOG_OBJECT (timidity, "Song ended, generating eos");
gst_pad_push_event (timidity->srcpad, gst_event_new_eos ());
timidity->o_seek = FALSE;
goto paused;
}
if (timidity->o_seek) {
GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
timidity->o_seek = FALSE;
}
gst_buffer_set_caps (out, timidity->out_caps);
ret = gst_pad_push (timidity->srcpad, out);
if (GST_FLOW_IS_FATAL (ret) || ret == GST_FLOW_NOT_LINKED)
goto error;
return;
paused:
{
GST_DEBUG_OBJECT (timidity, "pausing task");
gst_pad_pause_task (timidity->sinkpad);
return;
}
error:
{
GST_ELEMENT_ERROR (timidity, STREAM, FAILED,
("Internal data stream error"),
("Streaming stopped, reason %s", gst_flow_get_name (ret)));
gst_pad_push_event (timidity->srcpad, gst_event_new_eos ());
goto paused;
}
}
static void
gst_real_audio_demux_loop (GstRealAudioDemux * demux)
{
GstFlowReturn ret;
GstBuffer *buf;
guint bytes_needed;
/* check how much data we need */
switch (demux->state) {
case REAL_AUDIO_DEMUX_STATE_MARKER:
bytes_needed = 6 + 16; /* 16 are beginning of header */
break;
case REAL_AUDIO_DEMUX_STATE_HEADER:
if (!gst_real_audio_demux_get_data_offset_from_header (demux))
goto parse_header_error;
bytes_needed = demux->data_offset - (6 + 16);
break;
case REAL_AUDIO_DEMUX_STATE_DATA:
if (demux->packet_size > 0) {
/* TODO: should probably take into account width/height as well? */
bytes_needed = demux->packet_size;
} else {
bytes_needed = 1024;
}
break;
default:
g_return_if_reached ();
}
/* now get the data */
GST_LOG_OBJECT (demux, "getting data: %5u bytes @ %8" G_GINT64_MODIFIER "u",
bytes_needed, demux->offset);
if (demux->upstream_size > 0 && demux->offset >= demux->upstream_size)
goto eos;
ret = gst_pad_pull_range (demux->sinkpad, demux->offset, bytes_needed, &buf);
if (ret != GST_FLOW_OK)
goto pull_range_error;
if (GST_BUFFER_SIZE (buf) != bytes_needed)
goto pull_range_short_read;
ret = gst_real_audio_demux_handle_buffer (demux, buf);
if (ret != GST_FLOW_OK)
goto handle_flow_error;
/* TODO: increase this in chain function too (for timestamps)? */
demux->offset += bytes_needed;
/* check for the end of the segment */
if (demux->segment.stop != -1 && demux->segment.last_stop != -1 &&
demux->segment.last_stop > demux->segment.stop) {
GST_DEBUG_OBJECT (demux, "reached end of segment");
goto eos;
}
return;
/* ERRORS */
parse_header_error:
{
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL), (NULL));
goto pause_task;
}
handle_flow_error:
{
GST_WARNING_OBJECT (demux, "handle_buf flow: %s", gst_flow_get_name (ret));
goto pause_task;
}
pull_range_error:
{
GST_WARNING_OBJECT (demux, "pull range flow: %s", gst_flow_get_name (ret));
goto pause_task;
}
pull_range_short_read:
{
GST_WARNING_OBJECT (demux, "pull range short read: wanted %u bytes, but "
"got only %u bytes", bytes_needed, GST_BUFFER_SIZE (buf));
gst_buffer_unref (buf);
goto eos;
}
eos:
{
if (demux->state != REAL_AUDIO_DEMUX_STATE_DATA) {
GST_WARNING_OBJECT (demux, "reached EOS before finished parsing header");
goto parse_header_error;
}
GST_INFO_OBJECT (demux, "EOS");
if ((demux->segment.flags & GST_SEEK_FLAG_SEGMENT) != 0) {
gint64 stop;
/* for segment playback we need to post when (in stream time)
* we stopped, this is either stop (when set) or the duration. */
if ((stop = demux->segment.stop) == -1)
stop = demux->segment.duration;
GST_DEBUG_OBJECT (demux, "sending segment done, at end of segment");
gst_element_post_message (GST_ELEMENT (demux),
gst_message_new_segment_done (GST_OBJECT (demux), GST_FORMAT_TIME,
stop));
} else {
/* normal playback, send EOS event downstream */
GST_DEBUG_OBJECT (demux, "sending EOS event, at end of stream");
gst_pad_push_event (demux->srcpad, gst_event_new_eos ());
}
goto pause_task;
}
pause_task:
{
demux->segment_running = FALSE;
gst_pad_pause_task (demux->sinkpad);
GST_DEBUG_OBJECT (demux, "pausing task");
return;
}
}
static void
gst_aiff_parse_loop (GstPad * pad)
{
GstFlowReturn ret;
GstAiffParse *aiff = GST_AIFF_PARSE (GST_PAD_PARENT (pad));
GST_LOG_OBJECT (aiff, "process data");
switch (aiff->state) {
case AIFF_PARSE_START:
GST_INFO_OBJECT (aiff, "AIFF_PARSE_START");
if ((ret = gst_aiff_parse_stream_init (aiff)) != GST_FLOW_OK)
goto pause;
aiff->state = AIFF_PARSE_HEADER;
/* fall-through */
case AIFF_PARSE_HEADER:
GST_INFO_OBJECT (aiff, "AIFF_PARSE_HEADER");
if ((ret = gst_aiff_parse_stream_headers (aiff)) != GST_FLOW_OK)
goto pause;
aiff->state = AIFF_PARSE_DATA;
GST_INFO_OBJECT (aiff, "AIFF_PARSE_DATA");
/* fall-through */
case AIFF_PARSE_DATA:
if ((ret = gst_aiff_parse_stream_data (aiff)) != GST_FLOW_OK)
goto pause;
break;
default:
g_assert_not_reached ();
}
return;
/* ERRORS */
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (aiff, "pausing task, reason %s", reason);
aiff->segment_running = FALSE;
gst_pad_pause_task (pad);
if (ret == GST_FLOW_UNEXPECTED) {
/* perform EOS logic */
if (aiff->segment.flags & GST_SEEK_FLAG_SEGMENT) {
GstClockTime stop;
if ((stop = aiff->segment.stop) == -1)
stop = aiff->segment.duration;
gst_element_post_message (GST_ELEMENT_CAST (aiff),
gst_message_new_segment_done (GST_OBJECT_CAST (aiff),
aiff->segment.format, stop));
} else {
gst_pad_push_event (aiff->srcpad, gst_event_new_eos ());
}
} else if (ret < GST_FLOW_UNEXPECTED || ret == GST_FLOW_NOT_LINKED) {
/* for fatal errors we post an error message, post the error
* first so the app knows about the error first. */
GST_ELEMENT_ERROR (aiff, STREAM, FAILED,
(_("Internal data flow error.")),
("streaming task paused, reason %s (%d)", reason, ret));
gst_pad_push_event (aiff->srcpad, gst_event_new_eos ());
}
return;
}
}
static gboolean
gst_real_audio_demux_handle_seek (GstRealAudioDemux * demux, GstEvent * event)
{
GstFormat format;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
gboolean flush, update;
gdouble rate;
guint64 seek_pos;
gint64 cur, stop;
if (!demux->seekable)
goto not_seekable;
if (demux->byterate_num == 0 || demux->byterate_denom == 0)
goto no_bitrate;
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
if (format != GST_FORMAT_TIME)
goto only_time_format_supported;
if (rate <= 0.0)
goto cannot_do_backwards_playback;
flush = ((flags & GST_SEEK_FLAG_FLUSH) != 0);
GST_DEBUG_OBJECT (demux, "flush=%d, rate=%g", flush, rate);
/* unlock streaming thread and make streaming stop */
if (flush) {
gst_pad_push_event (demux->sinkpad, gst_event_new_flush_start ());
gst_pad_push_event (demux->srcpad, gst_event_new_flush_start ());
} else {
gst_pad_pause_task (demux->sinkpad);
}
GST_PAD_STREAM_LOCK (demux->sinkpad);
if (demux->segment_running && !flush) {
GstEvent *newseg;
newseg = gst_event_new_new_segment_full (TRUE, demux->segment.rate,
demux->segment.applied_rate, GST_FORMAT_TIME, demux->segment.start,
demux->segment.last_stop, demux->segment.time);
GST_DEBUG_OBJECT (demux, "sending NEWSEGMENT event to close the current "
"segment: %" GST_PTR_FORMAT, newseg);
gst_pad_push_event (demux->srcpad, newseg);
}
gst_segment_set_seek (&demux->segment, rate, format, flags,
cur_type, cur, stop_type, stop, &update);
GST_DEBUG_OBJECT (demux, "segment: %" GST_SEGMENT_FORMAT, &demux->segment);
seek_pos = gst_util_uint64_scale (demux->segment.start,
demux->byterate_num, demux->byterate_denom * GST_SECOND);
if (demux->packet_size > 0) {
seek_pos -= seek_pos % demux->packet_size;
}
seek_pos += demux->data_offset;
GST_DEBUG_OBJECT (demux, "seek_pos = %" G_GUINT64_FORMAT, seek_pos);
/* stop flushing */
gst_pad_push_event (demux->sinkpad, gst_event_new_flush_stop ());
gst_pad_push_event (demux->srcpad, gst_event_new_flush_stop ());
demux->offset = seek_pos;
demux->need_newsegment = TRUE;
/* notify start of new segment */
if (demux->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT (demux),
gst_message_new_segment_start (GST_OBJECT (demux),
GST_FORMAT_TIME, demux->segment.last_stop));
}
demux->segment_running = TRUE;
/* restart our task since it might have been stopped when we did the flush */
gst_pad_start_task (demux->sinkpad,
(GstTaskFunction) gst_real_audio_demux_loop, demux);
/* streaming can continue now */
GST_PAD_STREAM_UNLOCK (demux->sinkpad);
return TRUE;
/* ERRORS */
not_seekable:
{
GST_DEBUG_OBJECT (demux, "seek failed: cannot seek in streaming mode");
return FALSE;
}
no_bitrate:
{
GST_DEBUG_OBJECT (demux, "seek failed: bitrate unknown");
return FALSE;
}
only_time_format_supported:
{
GST_DEBUG_OBJECT (demux, "can only seek in TIME format");
return FALSE;
}
cannot_do_backwards_playback:
{
GST_DEBUG_OBJECT (demux, "can only seek with positive rate, not %lf", rate);
return FALSE;
}
}
static gboolean
gst_musepackdec_handle_seek_event (GstMusepackDec * dec, GstEvent * event)
{
GstSeekType start_type, stop_type;
GstSeekFlags flags;
GstSegment segment;
GstFormat format;
gboolean flush;
gdouble rate;
gint64 start, stop;
gint samplerate;
gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
&stop_type, &stop);
if (format != GST_FORMAT_TIME && format != GST_FORMAT_DEFAULT) {
GST_DEBUG_OBJECT (dec, "seek failed: only TIME or DEFAULT format allowed");
return FALSE;
}
samplerate = g_atomic_int_get (&dec->rate);
if (format == GST_FORMAT_TIME) {
if (start_type != GST_SEEK_TYPE_NONE)
start = gst_util_uint64_scale_int (start, samplerate, GST_SECOND);
if (stop_type != GST_SEEK_TYPE_NONE)
stop = gst_util_uint64_scale_int (stop, samplerate, GST_SECOND);
}
flush = ((flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH);
if (flush)
gst_pad_push_event (dec->srcpad, gst_event_new_flush_start ());
else
gst_pad_pause_task (dec->sinkpad); /* not _stop_task()? */
GST_PAD_STREAM_LOCK (dec->sinkpad);
/* operate on segment copy until we know the seek worked */
segment = dec->segment;
gst_segment_do_seek (&segment, rate, GST_FORMAT_DEFAULT,
flags, start_type, start, stop_type, stop, NULL);
gst_pad_push_event (dec->sinkpad, gst_event_new_flush_stop (TRUE));
GST_DEBUG_OBJECT (dec, "segment: [%" G_GINT64_FORMAT "-%" G_GINT64_FORMAT
"] = [%" GST_TIME_FORMAT "-%" GST_TIME_FORMAT "]",
segment.start, segment.stop,
GST_TIME_ARGS (segment.start * GST_SECOND / dec->rate),
GST_TIME_ARGS (segment.stop * GST_SECOND / dec->rate));
GST_DEBUG_OBJECT (dec, "performing seek to sample %" G_GINT64_FORMAT,
segment.start);
if (segment.start >= segment.duration) {
GST_WARNING_OBJECT (dec, "seek out of bounds");
goto failed;
}
if (mpc_demux_seek_sample (dec->d, segment.start) != MPC_STATUS_OK)
goto failed;
if ((flags & GST_SEEK_FLAG_SEGMENT) == GST_SEEK_FLAG_SEGMENT) {
GST_DEBUG_OBJECT (dec, "posting SEGMENT_START message");
gst_element_post_message (GST_ELEMENT (dec),
gst_message_new_segment_start (GST_OBJECT (dec), GST_FORMAT_TIME,
gst_util_uint64_scale_int (segment.start, GST_SECOND, dec->rate)));
}
if (flush) {
gst_pad_push_event (dec->srcpad, gst_event_new_flush_stop (TRUE));
}
segment.position = segment.start;
dec->segment = segment;
gst_musepackdec_send_newsegment (dec);
GST_DEBUG_OBJECT (dec, "seek successful");
gst_pad_start_task (dec->sinkpad,
(GstTaskFunction) gst_musepackdec_loop, dec->sinkpad, NULL);
GST_PAD_STREAM_UNLOCK (dec->sinkpad);
return TRUE;
failed:
{
GST_WARNING_OBJECT (dec, "seek failed");
GST_PAD_STREAM_UNLOCK (dec->sinkpad);
return FALSE;
}
}
static void
gst_pngdec_task (GstPad * pad)
{
GstPngDec *pngdec;
GstBuffer *buffer = NULL;
size_t buffer_size = 0;
gint i = 0;
png_bytep *rows, inp;
png_uint_32 rowbytes;
GstFlowReturn ret = GST_FLOW_OK;
pngdec = GST_PNGDEC (GST_OBJECT_PARENT (pad));
GST_LOG_OBJECT (pngdec, "read frame");
/* Let libpng come back here on error */
if (setjmp (png_jmpbuf (pngdec->png))) {
ret = GST_FLOW_ERROR;
goto pause;
}
/* Set reading callback */
png_set_read_fn (pngdec->png, pngdec, user_read_data);
/* Read info */
png_read_info (pngdec->png, pngdec->info);
/* Generate the caps and configure */
ret = gst_pngdec_caps_create_and_set (pngdec);
if (ret != GST_FLOW_OK) {
goto pause;
}
/* Allocate output buffer */
rowbytes = png_get_rowbytes (pngdec->png, pngdec->info);
if (rowbytes > (G_MAXUINT32 - 3) || pngdec->height > G_MAXUINT32 / rowbytes) {
ret = GST_FLOW_ERROR;
goto pause;
}
rowbytes = GST_ROUND_UP_4 (rowbytes);
buffer_size = pngdec->height * rowbytes;
ret =
gst_pad_alloc_buffer_and_set_caps (pngdec->srcpad, GST_BUFFER_OFFSET_NONE,
buffer_size, GST_PAD_CAPS (pngdec->srcpad), &buffer);
if (ret != GST_FLOW_OK)
goto pause;
rows = (png_bytep *) g_malloc (sizeof (png_bytep) * pngdec->height);
inp = GST_BUFFER_DATA (buffer);
for (i = 0; i < pngdec->height; i++) {
rows[i] = inp;
inp += rowbytes;
}
/* Read the actual picture */
png_read_image (pngdec->png, rows);
g_free (rows);
/* Push the raw RGB frame */
ret = gst_pad_push (pngdec->srcpad, buffer);
if (ret != GST_FLOW_OK)
goto pause;
/* And we are done */
gst_pad_pause_task (pngdec->sinkpad);
gst_pad_push_event (pngdec->srcpad, gst_event_new_eos ());
return;
pause:
{
GST_INFO_OBJECT (pngdec, "pausing task, reason %s",
gst_flow_get_name (ret));
gst_pad_pause_task (pngdec->sinkpad);
if (GST_FLOW_IS_FATAL (ret) || ret == GST_FLOW_NOT_LINKED) {
GST_ELEMENT_ERROR (pngdec, STREAM, FAILED,
(_("Internal data stream error.")),
("stream stopped, reason %s", gst_flow_get_name (ret)));
gst_pad_push_event (pngdec->srcpad, gst_event_new_eos ());
}
}
}
static gboolean
gst_raw_parse_handle_seek_pull (GstRawParse * rp, GstEvent * event)
{
gdouble rate;
GstFormat format;
GstSeekFlags flags;
GstSeekType start_type, stop_type;
gint64 start, stop;
gint64 last_stop;
gboolean ret = FALSE;
gboolean flush;
GstSegment seeksegment;
if (event) {
gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
&stop_type, &stop);
/* convert input offsets to time */
ret = gst_raw_parse_convert (rp, format, start, GST_FORMAT_TIME, &start);
ret &= gst_raw_parse_convert (rp, format, stop, GST_FORMAT_TIME, &stop);
if (!ret)
goto convert_failed;
GST_DEBUG_OBJECT (rp, "converted start - stop to time");
format = GST_FORMAT_TIME;
gst_event_unref (event);
} else {
format = GST_FORMAT_TIME;
flags = 0;
}
flush = ((flags & GST_SEEK_FLAG_FLUSH) != 0);
/* start flushing up and downstream so that the loop function pauses and we
* can acquire the STREAM_LOCK. */
if (flush) {
GST_LOG_OBJECT (rp, "flushing");
gst_pad_push_event (rp->sinkpad, gst_event_new_flush_start ());
gst_pad_push_event (rp->srcpad, gst_event_new_flush_start ());
} else {
GST_LOG_OBJECT (rp, "pause task");
gst_pad_pause_task (rp->sinkpad);
}
GST_PAD_STREAM_LOCK (rp->sinkpad);
memcpy (&seeksegment, &rp->segment, sizeof (GstSegment));
if (event) {
/* configure the seek values */
gst_segment_do_seek (&seeksegment, rate, format, flags,
start_type, start, stop_type, stop, NULL);
}
/* get the desired position */
last_stop = seeksegment.position;
GST_LOG_OBJECT (rp, "seeking to %" GST_TIME_FORMAT,
GST_TIME_ARGS (last_stop));
/* convert the desired position to bytes */
ret =
gst_raw_parse_convert (rp, format, last_stop, GST_FORMAT_BYTES,
&last_stop);
/* prepare for streaming */
if (flush) {
GST_LOG_OBJECT (rp, "stop flush");
gst_pad_push_event (rp->sinkpad, gst_event_new_flush_stop (TRUE));
gst_pad_push_event (rp->srcpad, gst_event_new_flush_stop (TRUE));
}
if (ret) {
/* seek done */
/* Seek on a frame boundary */
last_stop -= last_stop % rp->framesize;
rp->offset = last_stop;
rp->n_frames = last_stop / rp->framesize;
GST_LOG_OBJECT (rp, "seeking to bytes %" G_GINT64_FORMAT, last_stop);
memcpy (&rp->segment, &seeksegment, sizeof (GstSegment));
if (rp->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (rp),
gst_message_new_segment_start (GST_OBJECT_CAST (rp),
rp->segment.format, rp->segment.position));
}
/* for deriving a stop position for the playback segment from the seek
* segment, we must take the duration when the stop is not set */
if ((stop = rp->segment.stop) == -1)
stop = rp->segment.duration;
GST_DEBUG_OBJECT (rp, "preparing newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, rp->segment.start, stop);
/* now replace the old segment so that we send it in the stream thread the
* next time it is scheduled. */
if (rp->start_segment)
gst_event_unref (rp->start_segment);
rp->start_segment = gst_event_new_segment (&rp->segment);
}
rp->discont = TRUE;
GST_LOG_OBJECT (rp, "start streaming");
gst_pad_start_task (rp->sinkpad, (GstTaskFunction) gst_raw_parse_loop, rp,
NULL);
GST_PAD_STREAM_UNLOCK (rp->sinkpad);
return ret;
/* ERRORS */
convert_failed:
{
GST_DEBUG_OBJECT (rp, "Seek failed: couldn't convert to byte positions");
return FALSE;
}
}
static void
gst_raw_parse_loop (GstElement * element)
{
GstRawParse *rp = GST_RAW_PARSE (element);
GstRawParseClass *rp_class = GST_RAW_PARSE_GET_CLASS (rp);
GstFlowReturn ret;
GstBuffer *buffer;
gint size;
if (!gst_raw_parse_set_src_caps (rp))
goto no_caps;
if (rp->start_segment) {
GST_DEBUG_OBJECT (rp, "sending start segment");
gst_pad_push_event (rp->srcpad, rp->start_segment);
rp->start_segment = NULL;
}
if (rp_class->multiple_frames_per_buffer && rp->framesize < 4096)
size = 4096 - (4096 % rp->framesize);
else
size = rp->framesize;
if (rp->segment.rate >= 0) {
if (rp->offset + size > rp->upstream_length) {
GstFormat fmt = GST_FORMAT_BYTES;
if (!gst_pad_peer_query_duration (rp->sinkpad, fmt, &rp->upstream_length)) {
GST_WARNING_OBJECT (rp,
"Could not get upstream duration, trying to pull frame by frame");
size = rp->framesize;
} else if (rp->upstream_length < rp->offset + rp->framesize) {
ret = GST_FLOW_EOS;
goto pause;
} else if (rp->offset + size > rp->upstream_length) {
size = rp->upstream_length - rp->offset;
size -= size % rp->framesize;
}
}
} else {
if (rp->offset == 0) {
ret = GST_FLOW_EOS;
goto pause;
} else if (rp->offset < size) {
size -= rp->offset;
}
rp->offset -= size;
}
buffer = NULL;
ret = gst_pad_pull_range (rp->sinkpad, rp->offset, size, &buffer);
if (ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (rp, "pull_range (%" G_GINT64_FORMAT ", %u) "
"failed, flow: %s", rp->offset, size, gst_flow_get_name (ret));
buffer = NULL;
goto pause;
}
if (gst_buffer_get_size (buffer) < size) {
GST_DEBUG_OBJECT (rp, "Short read at offset %" G_GINT64_FORMAT
", got only %" G_GSIZE_FORMAT " of %u bytes", rp->offset,
gst_buffer_get_size (buffer), size);
if (size > rp->framesize) {
gst_buffer_set_size (buffer, gst_buffer_get_size (buffer) -
gst_buffer_get_size (buffer) % rp->framesize);
} else {
gst_buffer_unref (buffer);
buffer = NULL;
ret = GST_FLOW_EOS;
goto pause;
}
}
ret = gst_raw_parse_push_buffer (rp, buffer);
if (ret != GST_FLOW_OK)
goto pause;
return;
/* ERRORS */
no_caps:
{
GST_ERROR_OBJECT (rp, "could not negotiate caps");
ret = GST_FLOW_NOT_NEGOTIATED;
goto pause;
}
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_LOG_OBJECT (rp, "pausing task, reason %s", reason);
gst_pad_pause_task (rp->sinkpad);
if (ret == GST_FLOW_EOS) {
if (rp->segment.flags & GST_SEEK_FLAG_SEGMENT) {
GstClockTime stop;
GST_LOG_OBJECT (rp, "Sending segment done");
if ((stop = rp->segment.stop) == -1)
stop = rp->segment.duration;
gst_element_post_message (GST_ELEMENT_CAST (rp),
gst_message_new_segment_done (GST_OBJECT_CAST (rp),
rp->segment.format, stop));
gst_pad_push_event (rp->srcpad,
gst_event_new_segment_done (rp->segment.format, stop));
} else {
GST_LOG_OBJECT (rp, "Sending EOS, at end of stream");
gst_pad_push_event (rp->srcpad, gst_event_new_eos ());
}
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (rp, STREAM, FAILED,
("Internal data stream error."),
("stream stopped, reason %s", reason));
gst_pad_push_event (rp->srcpad, gst_event_new_eos ());
}
return;
}
}
static void
gst_musepackdec_loop (GstPad * sinkpad)
{
GstMusepackDec *musepackdec;
GstFlowReturn flow;
GstBuffer *out;
#ifdef MPC_IS_OLD_API
guint32 update_acc, update_bits;
#else
mpc_frame_info frame;
mpc_status err;
#endif
gint num_samples, samplerate, bitspersample;
musepackdec = GST_MUSEPACK_DEC (GST_PAD_PARENT (sinkpad));
samplerate = g_atomic_int_get (&musepackdec->rate);
if (samplerate == 0) {
if (!gst_musepack_stream_init (musepackdec))
goto pause_task;
gst_musepackdec_send_newsegment (musepackdec);
samplerate = g_atomic_int_get (&musepackdec->rate);
}
bitspersample = g_atomic_int_get (&musepackdec->bps);
flow = gst_pad_alloc_buffer_and_set_caps (musepackdec->srcpad, -1,
MPC_DECODER_BUFFER_LENGTH * 4, GST_PAD_CAPS (musepackdec->srcpad), &out);
if (flow != GST_FLOW_OK) {
GST_DEBUG_OBJECT (musepackdec, "Flow: %s", gst_flow_get_name (flow));
goto pause_task;
}
#ifdef MPC_IS_OLD_API
num_samples = mpc_decoder_decode (musepackdec->d,
(MPC_SAMPLE_FORMAT *) GST_BUFFER_DATA (out), &update_acc, &update_bits);
if (num_samples < 0) {
GST_ERROR_OBJECT (musepackdec, "Failed to decode sample");
GST_ELEMENT_ERROR (musepackdec, STREAM, DECODE, (NULL), (NULL));
goto pause_task;
} else if (num_samples == 0) {
goto eos_and_pause;
}
#else
frame.buffer = (MPC_SAMPLE_FORMAT *) GST_BUFFER_DATA (out);
err = mpc_demux_decode (musepackdec->d, &frame);
if (err != MPC_STATUS_OK) {
GST_ERROR_OBJECT (musepackdec, "Failed to decode sample");
GST_ELEMENT_ERROR (musepackdec, STREAM, DECODE, (NULL), (NULL));
goto pause_task;
} else if (frame.bits == -1) {
goto eos_and_pause;
}
num_samples = frame.samples;
#endif
GST_BUFFER_SIZE (out) = num_samples * bitspersample;
GST_BUFFER_OFFSET (out) = musepackdec->segment.last_stop;
GST_BUFFER_TIMESTAMP (out) =
gst_util_uint64_scale_int (musepackdec->segment.last_stop,
GST_SECOND, samplerate);
GST_BUFFER_DURATION (out) =
gst_util_uint64_scale_int (num_samples, GST_SECOND, samplerate);
musepackdec->segment.last_stop += num_samples;
GST_LOG_OBJECT (musepackdec, "Pushing buffer, timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out)));
flow = gst_pad_push (musepackdec->srcpad, out);
if (flow != GST_FLOW_OK) {
GST_DEBUG_OBJECT (musepackdec, "Flow: %s", gst_flow_get_name (flow));
goto pause_task;
}
/* check if we're at the end of a configured segment */
if (musepackdec->segment.stop != -1 &&
musepackdec->segment.last_stop >= musepackdec->segment.stop) {
gint64 stop_time;
GST_DEBUG_OBJECT (musepackdec, "Reached end of configured segment");
if ((musepackdec->segment.flags & GST_SEEK_FLAG_SEGMENT) == 0)
goto eos_and_pause;
GST_DEBUG_OBJECT (musepackdec, "Posting SEGMENT_DONE message");
stop_time = gst_util_uint64_scale_int (musepackdec->segment.stop,
GST_SECOND, samplerate);
gst_element_post_message (GST_ELEMENT (musepackdec),
gst_message_new_segment_done (GST_OBJECT (musepackdec),
GST_FORMAT_TIME, stop_time));
goto pause_task;
}
return;
eos_and_pause:
{
GST_DEBUG_OBJECT (musepackdec, "sending EOS event");
gst_pad_push_event (musepackdec->srcpad, gst_event_new_eos ());
/* fall through to pause */
}
pause_task:
{
GST_DEBUG_OBJECT (musepackdec, "Pausing task");
gst_pad_pause_task (sinkpad);
return;
}
}
/* push packets from the queue to the downstream demuxer */
static void
gst_rdt_manager_loop (GstPad * pad)
{
GstRDTManager *rdtmanager;
GstRDTManagerSession *session;
GstBuffer *buffer;
GstFlowReturn result;
rdtmanager = GST_RDT_MANAGER (GST_PAD_PARENT (pad));
session = gst_pad_get_element_private (pad);
JBUF_LOCK_CHECK (session, flushing);
GST_DEBUG_OBJECT (rdtmanager, "Peeking item");
while (TRUE) {
/* always wait if we are blocked */
if (!session->blocked) {
/* if we have a packet, we can exit the loop and grab it */
if (rdt_jitter_buffer_num_packets (session->jbuf) > 0)
break;
/* no packets but we are EOS, do eos logic */
if (session->eos)
goto do_eos;
}
/* underrun, wait for packets or flushing now */
session->waiting = TRUE;
JBUF_WAIT_CHECK (session, flushing);
session->waiting = FALSE;
}
buffer = rdt_jitter_buffer_pop (session->jbuf);
GST_DEBUG_OBJECT (rdtmanager, "Got item %p", buffer);
if (session->discont) {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
session->discont = FALSE;
}
JBUF_UNLOCK (session);
result = gst_pad_push (session->recv_rtp_src, buffer);
if (result != GST_FLOW_OK)
goto pause;
return;
/* ERRORS */
flushing:
{
GST_DEBUG_OBJECT (rdtmanager, "we are flushing");
gst_pad_pause_task (session->recv_rtp_src);
JBUF_UNLOCK (session);
return;
}
do_eos:
{
/* store result, we are flushing now */
GST_DEBUG_OBJECT (rdtmanager, "We are EOS, pushing EOS downstream");
session->srcresult = GST_FLOW_EOS;
gst_pad_pause_task (session->recv_rtp_src);
gst_pad_push_event (session->recv_rtp_src, gst_event_new_eos ());
JBUF_UNLOCK (session);
return;
}
pause:
{
GST_DEBUG_OBJECT (rdtmanager, "pausing task, reason %s",
gst_flow_get_name (result));
JBUF_LOCK (session);
/* store result */
session->srcresult = result;
/* we don't post errors or anything because upstream will do that for us
* when we pass the return value upstream. */
gst_pad_pause_task (session->recv_rtp_src);
JBUF_UNLOCK (session);
return;
}
}
static void
gst_tcp_mix_src_loop (GstTCPMixSrcPad * pad)
{
GstTCPMixSrc *src = GST_TCP_MIX_SRC (GST_PAD_PARENT (pad));
GstBuffer *buffer = NULL;
GstFlowReturn ret;
/*
INFO ("%p, %s", g_thread_self (), GST_PAD_NAME (pad));
*/
if (!pad->running) {
gst_tcp_mix_src_pad_wait_for_client (pad);
pad->running = TRUE;
}
ret = gst_tcp_mix_src_pad_read (pad, &buffer);
if (ret != GST_FLOW_OK)
goto error_read_pad;
ret = gst_pad_push (GST_PAD (pad), buffer);
if (ret != GST_FLOW_OK)
goto error_push_buffer;
return;
/* Handling Errors */
error_read_pad:
{
GST_ERROR_OBJECT (src, "Can't read from %s:%s",
GST_ELEMENT_NAME (src), GST_PAD_NAME (pad));
goto pause;
}
error_push_buffer:
{
GST_ERROR_OBJECT (src, "Can't push buffer to %s:%s",
GST_ELEMENT_NAME (src), GST_PAD_NAME (pad));
goto pause;
}
pause:
{
if (src->mode == MODE_LOOP)
goto loop;
GstEvent *event;
if (ret == GST_FLOW_EOS) {
event = gst_event_new_eos ();
gst_pad_push_event (GST_PAD (pad), event);
} else if (ret == GST_FLOW_NOT_LINKED || ret <= GST_FLOW_EOS) {
event = gst_event_new_eos ();
GST_ELEMENT_ERROR (src, STREAM, FAILED,
("Internal data flow error."),
("streaming task paused (%s (%d))", gst_flow_get_name (ret), ret));
gst_pad_push_event (GST_PAD (pad), event);
}
/*
GST_DEBUG_OBJECT (pad, "Paused %s.%s (%s)",
GST_ELEMENT_NAME (src), GST_PAD_NAME (pad),
gst_flow_get_name (ret));
*/
gst_pad_pause_task (GST_PAD (pad));
return;
}
loop:
{
if (ret == GST_FLOW_NOT_LINKED || ret <= GST_FLOW_EOS) {
GST_ELEMENT_ERROR (src, STREAM, FAILED,
("Internal data flow error."),
("streaming task paused (%s (%d))", gst_flow_get_name (ret), ret));
}
return;
}
}
static void
gst_musepackdec_loop (GstPad * sinkpad)
{
GstMusepackDec *musepackdec;
GstFlowReturn flow;
GstBuffer *out;
GstMapInfo info;
mpc_frame_info frame;
mpc_status err;
gint num_samples, samplerate, bitspersample;
musepackdec = GST_MUSEPACK_DEC (GST_PAD_PARENT (sinkpad));
samplerate = g_atomic_int_get (&musepackdec->rate);
if (samplerate == 0) {
if (!gst_musepack_stream_init (musepackdec))
goto pause_task;
gst_musepackdec_send_newsegment (musepackdec);
samplerate = g_atomic_int_get (&musepackdec->rate);
}
bitspersample = g_atomic_int_get (&musepackdec->bps);
out = gst_buffer_new_allocate (NULL, MPC_DECODER_BUFFER_LENGTH * 4, NULL);
gst_buffer_map (out, &info, GST_MAP_READWRITE);
frame.buffer = (MPC_SAMPLE_FORMAT *) info.data;
err = mpc_demux_decode (musepackdec->d, &frame);
gst_buffer_unmap (out, &info);
if (err != MPC_STATUS_OK) {
GST_ERROR_OBJECT (musepackdec, "Failed to decode sample");
GST_ELEMENT_ERROR (musepackdec, STREAM, DECODE, (NULL), (NULL));
goto pause_task;
} else if (frame.bits == -1) {
goto eos_and_pause;
}
num_samples = frame.samples;
gst_buffer_set_size (out, num_samples * bitspersample);
GST_BUFFER_OFFSET (out) = musepackdec->segment.position;
GST_BUFFER_PTS (out) =
gst_util_uint64_scale_int (musepackdec->segment.position,
GST_SECOND, samplerate);
GST_BUFFER_DURATION (out) =
gst_util_uint64_scale_int (num_samples, GST_SECOND, samplerate);
musepackdec->segment.position += num_samples;
GST_LOG_OBJECT (musepackdec, "Pushing buffer, timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out)));
flow = gst_pad_push (musepackdec->srcpad, out);
if (flow != GST_FLOW_OK) {
GST_DEBUG_OBJECT (musepackdec, "Flow: %s", gst_flow_get_name (flow));
goto pause_task;
}
/* check if we're at the end of a configured segment */
if (musepackdec->segment.stop != -1 &&
musepackdec->segment.position >= musepackdec->segment.stop) {
gint64 stop_time;
GST_DEBUG_OBJECT (musepackdec, "Reached end of configured segment");
if ((musepackdec->segment.flags & GST_SEEK_FLAG_SEGMENT) == 0)
goto eos_and_pause;
GST_DEBUG_OBJECT (musepackdec, "Posting SEGMENT_DONE message");
stop_time = gst_util_uint64_scale_int (musepackdec->segment.stop,
GST_SECOND, samplerate);
gst_element_post_message (GST_ELEMENT (musepackdec),
gst_message_new_segment_done (GST_OBJECT (musepackdec),
GST_FORMAT_TIME, stop_time));
gst_pad_push_event (musepackdec->srcpad,
gst_event_new_segment_done (GST_FORMAT_TIME, stop_time));
goto pause_task;
}
return;
eos_and_pause:
{
GST_DEBUG_OBJECT (musepackdec, "sending EOS event");
gst_pad_push_event (musepackdec->srcpad, gst_event_new_eos ());
/* fall through to pause */
}
pause_task:
{
GST_DEBUG_OBJECT (musepackdec, "Pausing task");
gst_pad_pause_task (sinkpad);
return;
}
}
static void
gst_wildmidi_loop (GstPad * sinkpad)
{
GstWildmidi *wildmidi = GST_WILDMIDI (GST_PAD_PARENT (sinkpad));
GstFlowReturn ret;
switch (wildmidi->state) {
case GST_WILDMIDI_STATE_LOAD:
{
GstBuffer *buffer = NULL;
GST_DEBUG_OBJECT (wildmidi, "loading song");
ret =
gst_pad_pull_range (wildmidi->sinkpad, wildmidi->offset, -1, &buffer);
if (ret == GST_FLOW_EOS) {
GST_DEBUG_OBJECT (wildmidi, "Song loaded");
wildmidi->state = GST_WILDMIDI_STATE_PARSE;
} else if (ret != GST_FLOW_OK) {
GST_ELEMENT_ERROR (wildmidi, STREAM, DECODE, (NULL),
("Unable to read song"));
goto pause;
} else {
GST_DEBUG_OBJECT (wildmidi, "pushing buffer");
gst_adapter_push (wildmidi->adapter, buffer);
wildmidi->offset += gst_buffer_get_size (buffer);
}
break;
}
case GST_WILDMIDI_STATE_PARSE:
ret = gst_wildmidi_parse_song (wildmidi);
if (ret != GST_FLOW_OK)
goto pause;
wildmidi->state = GST_WILDMIDI_STATE_PLAY;
break;
case GST_WILDMIDI_STATE_PLAY:
ret = gst_wildmidi_do_play (wildmidi);
if (ret != GST_FLOW_OK)
goto pause;
break;
default:
break;
}
return;
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GstEvent *event;
GST_DEBUG_OBJECT (wildmidi, "pausing task, reason %s", reason);
gst_pad_pause_task (sinkpad);
if (ret == GST_FLOW_EOS) {
/* perform EOS logic */
event = gst_event_new_eos ();
gst_pad_push_event (wildmidi->srcpad, event);
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
event = gst_event_new_eos ();
/* for fatal errors we post an error message, post the error
* first so the app knows about the error first. */
GST_ELEMENT_ERROR (wildmidi, STREAM, FAILED,
("Internal data flow error."),
("streaming task paused, reason %s (%d)", reason, ret));
gst_pad_push_event (wildmidi->srcpad, event);
}
}
}
/* This function is used to perform seeks on the element in
* pull mode.
*
* It also works when event is NULL, in which case it will just
* start from the last configured segment. This technique is
* used when activating the element and to perform the seek in
* READY.
*/
static gboolean
gst_aiff_parse_perform_seek (GstAiffParse * aiff, GstEvent * event)
{
gboolean res;
gdouble rate;
GstFormat format, bformat;
GstSeekFlags flags;
GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
gint64 cur, stop, upstream_size;
gboolean flush;
gboolean update;
GstSegment seeksegment = { 0, };
gint64 last_stop;
if (event) {
GST_DEBUG_OBJECT (aiff, "doing seek with event");
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
/* no negative rates yet */
if (rate < 0.0)
goto negative_rate;
if (format != aiff->segment.format) {
GST_INFO_OBJECT (aiff, "converting seek-event from %s to %s",
gst_format_get_name (format),
gst_format_get_name (aiff->segment.format));
res = TRUE;
if (cur_type != GST_SEEK_TYPE_NONE)
res =
gst_pad_query_convert (aiff->srcpad, format, cur,
&aiff->segment.format, &cur);
if (res && stop_type != GST_SEEK_TYPE_NONE)
res =
gst_pad_query_convert (aiff->srcpad, format, stop,
&aiff->segment.format, &stop);
if (!res)
goto no_format;
format = aiff->segment.format;
}
} else {
GST_DEBUG_OBJECT (aiff, "doing seek without event");
flags = 0;
rate = 1.0;
cur_type = GST_SEEK_TYPE_SET;
stop_type = GST_SEEK_TYPE_SET;
}
/* get flush flag */
flush = flags & GST_SEEK_FLAG_FLUSH;
/* now we need to make sure the streaming thread is stopped. We do this by
* either sending a FLUSH_START event downstream which will cause the
* streaming thread to stop with a WRONG_STATE.
* For a non-flushing seek we simply pause the task, which will happen as soon
* as it completes one iteration (and thus might block when the sink is
* blocking in preroll). */
if (flush) {
GST_DEBUG_OBJECT (aiff, "sending flush start");
gst_pad_push_event (aiff->srcpad, gst_event_new_flush_start ());
} else {
gst_pad_pause_task (aiff->sinkpad);
}
/* we should now be able to grab the streaming thread because we stopped it
* with the above flush/pause code */
GST_PAD_STREAM_LOCK (aiff->sinkpad);
/* save current position */
last_stop = aiff->segment.last_stop;
GST_DEBUG_OBJECT (aiff, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
/* copy segment, we need this because we still need the old
* segment when we close the current segment. */
memcpy (&seeksegment, &aiff->segment, sizeof (GstSegment));
/* configure the seek parameters in the seeksegment. We will then have the
* right values in the segment to perform the seek */
if (event) {
GST_DEBUG_OBJECT (aiff, "configuring seek");
gst_segment_set_seek (&seeksegment, rate, format, flags,
cur_type, cur, stop_type, stop, &update);
}
/* figure out the last position we need to play. If it's configured (stop !=
* -1), use that, else we play until the total duration of the file */
if ((stop = seeksegment.stop) == -1)
stop = seeksegment.duration;
GST_DEBUG_OBJECT (aiff, "cur_type =%d", cur_type);
if ((cur_type != GST_SEEK_TYPE_NONE)) {
/* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
* we can just copy the last_stop. If not, we use the bps to convert TIME to
* bytes. */
if (aiff->bps > 0)
aiff->offset =
uint64_ceiling_scale (seeksegment.last_stop, (guint64) aiff->bps,
GST_SECOND);
else
aiff->offset = seeksegment.last_stop;
GST_LOG_OBJECT (aiff, "offset=%" G_GUINT64_FORMAT, aiff->offset);
aiff->offset -= (aiff->offset % aiff->bytes_per_sample);
GST_LOG_OBJECT (aiff, "offset=%" G_GUINT64_FORMAT, aiff->offset);
aiff->offset += aiff->datastart;
GST_LOG_OBJECT (aiff, "offset=%" G_GUINT64_FORMAT, aiff->offset);
} else {
GST_LOG_OBJECT (aiff, "continue from offset=%" G_GUINT64_FORMAT,
aiff->offset);
}
if (stop_type != GST_SEEK_TYPE_NONE) {
if (aiff->bps > 0)
aiff->end_offset =
uint64_ceiling_scale (stop, (guint64) aiff->bps, GST_SECOND);
else
aiff->end_offset = stop;
GST_LOG_OBJECT (aiff, "end_offset=%" G_GUINT64_FORMAT, aiff->end_offset);
aiff->end_offset -= (aiff->end_offset % aiff->bytes_per_sample);
GST_LOG_OBJECT (aiff, "end_offset=%" G_GUINT64_FORMAT, aiff->end_offset);
aiff->end_offset += aiff->datastart;
GST_LOG_OBJECT (aiff, "end_offset=%" G_GUINT64_FORMAT, aiff->end_offset);
} else {
GST_LOG_OBJECT (aiff, "continue to end_offset=%" G_GUINT64_FORMAT,
aiff->end_offset);
}
/* make sure filesize is not exceeded due to rounding errors or so,
* same precaution as in _stream_headers */
bformat = GST_FORMAT_BYTES;
if (gst_pad_query_peer_duration (aiff->sinkpad, &bformat, &upstream_size))
aiff->end_offset = MIN (aiff->end_offset, upstream_size);
/* this is the range of bytes we will use for playback */
aiff->offset = MIN (aiff->offset, aiff->end_offset);
aiff->dataleft = aiff->end_offset - aiff->offset;
GST_DEBUG_OBJECT (aiff,
"seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, aiff->offset,
aiff->end_offset, GST_TIME_ARGS (seeksegment.start),
GST_TIME_ARGS (stop));
/* prepare for streaming again */
if (flush) {
/* if we sent a FLUSH_START, we now send a FLUSH_STOP */
GST_DEBUG_OBJECT (aiff, "sending flush stop");
gst_pad_push_event (aiff->srcpad, gst_event_new_flush_stop ());
} else if (aiff->segment_running) {
/* we are running the current segment and doing a non-flushing seek,
* close the segment first based on the previous last_stop. */
GST_DEBUG_OBJECT (aiff, "closing running segment %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, aiff->segment.accum, aiff->segment.last_stop);
/* queue the segment for sending in the stream thread */
if (aiff->close_segment)
gst_event_unref (aiff->close_segment);
aiff->close_segment = gst_event_new_new_segment (TRUE,
aiff->segment.rate, aiff->segment.format,
aiff->segment.accum, aiff->segment.last_stop, aiff->segment.accum);
/* keep track of our last_stop */
seeksegment.accum = aiff->segment.last_stop;
}
/* now we did the seek and can activate the new segment values */
memcpy (&aiff->segment, &seeksegment, sizeof (GstSegment));
/* if we're doing a segment seek, post a SEGMENT_START message */
if (aiff->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (aiff),
gst_message_new_segment_start (GST_OBJECT_CAST (aiff),
aiff->segment.format, aiff->segment.last_stop));
}
/* now create the newsegment */
GST_DEBUG_OBJECT (aiff, "Creating newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, aiff->segment.last_stop, stop);
/* store the newsegment event so it can be sent from the streaming thread. */
if (aiff->start_segment)
gst_event_unref (aiff->start_segment);
aiff->start_segment =
gst_event_new_new_segment (FALSE, aiff->segment.rate,
aiff->segment.format, aiff->segment.last_stop, stop,
aiff->segment.last_stop);
/* mark discont if we are going to stream from another position. */
if (last_stop != aiff->segment.last_stop) {
GST_DEBUG_OBJECT (aiff, "mark DISCONT, we did a seek to another position");
aiff->discont = TRUE;
}
/* and start the streaming task again */
aiff->segment_running = TRUE;
if (!aiff->streaming) {
gst_pad_start_task (aiff->sinkpad, (GstTaskFunction) gst_aiff_parse_loop,
aiff->sinkpad);
}
GST_PAD_STREAM_UNLOCK (aiff->sinkpad);
return TRUE;
/* ERRORS */
negative_rate:
{
GST_DEBUG_OBJECT (aiff, "negative playback rates are not supported yet.");
return FALSE;
}
no_format:
{
GST_DEBUG_OBJECT (aiff, "unsupported format given, seek aborted.");
return FALSE;
}
}
static void
gst_amc_audio_dec_loop (GstAmcAudioDec * self)
{
GstFlowReturn flow_ret = GST_FLOW_OK;
gboolean is_eos;
GstAmcBufferInfo buffer_info;
gint idx;
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);
GST_AUDIO_DECODER_STREAM_LOCK (self);
/*} */
if (idx < 0) {
if (self->flushing)
goto flushing;
switch (idx) {
case INFO_OUTPUT_BUFFERS_CHANGED:{
GST_DEBUG_OBJECT (self, "Output buffers have changed");
if (self->output_buffers)
gst_amc_codec_free_buffers (self->output_buffers,
self->n_output_buffers);
self->output_buffers =
gst_amc_codec_get_output_buffers (self->codec,
&self->n_output_buffers);
if (!self->output_buffers)
goto get_output_buffers_error;
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);
if (!format)
goto format_error;
format_string = gst_amc_format_to_string (format);
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);
if (self->output_buffers)
gst_amc_codec_free_buffers (self->output_buffers,
self->n_output_buffers);
self->output_buffers =
gst_amc_codec_get_output_buffers (self->codec,
&self->n_output_buffers);
if (!self->output_buffers)
goto get_output_buffers_error;
goto retry;
break;
}
case INFO_TRY_AGAIN_LATER:
GST_DEBUG_OBJECT (self, "Dequeueing output buffer timed out");
goto retry;
break;
case G_MININT:
GST_ERROR_OBJECT (self, "Failure dequeueing output buffer");
goto dequeue_error;
break;
default:
g_assert_not_reached ();
break;
}
goto retry;
}
GST_DEBUG_OBJECT (self,
"Got output buffer at index %d: size %d time %" G_GINT64_FORMAT
" flags 0x%08x", idx, buffer_info.size, buffer_info.presentation_time_us,
buffer_info.flags);
is_eos = ! !(buffer_info.flags & BUFFER_FLAG_END_OF_STREAM);
self->n_buffers++;
if (buffer_info.size > 0) {
GstAmcAudioDecClass *klass = GST_AMC_AUDIO_DEC_GET_CLASS (self);
GstBuffer *outbuf;
GstAmcBuffer *buf;
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 (idx >= self->n_output_buffers)
goto invalid_buffer_index;
if (strcmp (klass->codec_info->name, "OMX.google.mp3.decoder") == 0) {
/* Google's MP3 decoder outputs garbage in the first output buffer
* so we just drop it here */
if (self->n_buffers == 1) {
GST_DEBUG_OBJECT (self,
"Skipping first buffer of Google MP3 decoder output");
goto done;
}
}
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);
buf = &self->output_buffers[idx];
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);
/* FIXME: We should get one decoded input frame here for
* every buffer. If this is not the case somewhere, we will
* error out at some point and will need to add workarounds
*/
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, 1);
}
done:
if (!gst_amc_codec_release_output_buffer (self->codec, idx))
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 (self, LIBRARY, FAILED, (NULL),
("Failed to dequeue output buffer"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
get_output_buffers_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Failed to get output buffers"));
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);
return;
}
format_error:
{
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);
return;
}
failed_release:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Failed to release output buffer index %d", idx));
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);
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_NOT_LINKED || flow_ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
("Internal data stream error."), ("stream stopped, reason %s",
gst_flow_get_name (flow_ret)));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
}
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
invalid_buffer_index:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Invalid input buffer index %d of %d", idx, self->n_input_buffers));
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);
return;
}
failed_allocate:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to allocate output buffer"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
}
