Commit 9e107d67 authored by Tim-Philipp Müller's avatar Tim-Philipp Müller
Browse files

Printf format fixes.

Original commit message from CVS:
* ext/alsa/gstalsadeviceprobe.c:
(gst_alsa_device_property_probe_get_values):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
(gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
(gst_ogg_mux_process_best_pad):
* ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
(gst_ogg_parse_chain):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
(gst_vorbis_enc_buffer_check_discontinuous):
* ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_track_seek):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_push_full):
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
* gst/audioresample/resample.c: (resample_input_pushthrough):
* gst/playback/gstplaybasebin.c: (queue_out_of_data):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(wavpack_type_find):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
* tests/check/elements/volume.c: (GST_START_TEST):
Printf format fixes.
parent d73263cd
2006-10-05 Tim-Philipp Müller <tim at centricular dot net>
* ext/alsa/gstalsadeviceprobe.c:
(gst_alsa_device_property_probe_get_values):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
(gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
(gst_ogg_mux_process_best_pad):
* ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
(gst_ogg_parse_chain):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
(gst_vorbis_enc_buffer_check_discontinuous):
* ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_track_seek):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_push_full):
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
* gst/audioresample/resample.c: (resample_input_pushthrough):
* gst/playback/gstplaybasebin.c: (queue_out_of_data):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(wavpack_type_find):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
* tests/check/elements/volume.c: (GST_START_TEST):
Printf format fixes.
2006-10-04 Thomas Vander Stichele <thomas at apestaart dot org>
 
* gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
common @ a4af1b85
Subproject commit 9991f6fa61ee11475c390dd6675ef7952f079e43
Subproject commit a4af1b8542911352e29d53fba47e2c3d7231ebdb
......@@ -171,7 +171,7 @@ gst_alsa_device_property_probe_get_values (GstPropertyProbe * probe,
array = g_value_array_new (g_list_length (list));
g_value_init (&value, G_TYPE_STRING);
for (l = list; l != NULL; l = l->next) {
GST_LOG_OBJECT (probe, "Found device: %s", l->data);
GST_LOG_OBJECT (probe, "Found device: %s", (gchar *) l->data);
g_value_take_string (&value, (gchar *) l->data);
l->data = NULL;
g_value_array_append (array, &value);
......
......@@ -423,7 +423,7 @@ retry:
CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, &dir),
period_size);
GST_DEBUG_OBJECT (alsa, "buffer size %u, period size %u", alsa->buffer_size,
GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,
alsa->period_size);
return 0;
......
......@@ -926,7 +926,7 @@ gst_ogg_demux_chain_elem_pad (GstOggPad * pad, ogg_packet * packet)
no_decoder:
{
GST_WARNING_OBJECT (ogg,
"pad %08x does not have elem_pad, no decoder ?", pad);
"pad %p does not have elem_pad, no decoder ?", pad);
return GST_FLOW_ERROR;
}
decoder_error:
......@@ -1284,7 +1284,7 @@ gst_ogg_chain_new_stream (GstOggChain * chain, glong serialno)
GstTagList *list;
gchar *name;
GST_DEBUG_OBJECT (chain->ogg, "creating new stream %08x in chain %p",
GST_DEBUG_OBJECT (chain->ogg, "creating new stream %08lx in chain %p",
serialno, chain);
ret = g_object_new (GST_TYPE_OGG_PAD, NULL);
......@@ -1304,7 +1304,7 @@ gst_ogg_chain_new_stream (GstOggChain * chain, glong serialno)
ret->serialno = serialno;
if (ogg_stream_init (&ret->stream, serialno) != 0) {
GST_ERROR ("Could not initialize ogg_stream struct for serial %08x.",
GST_ERROR ("Could not initialize ogg_stream struct for serial %08lx.",
serialno);
gst_object_unref (ret);
return NULL;
......@@ -1316,7 +1316,7 @@ gst_ogg_chain_new_stream (GstOggChain * chain, glong serialno)
gst_tag_list_free (list);
GST_DEBUG_OBJECT (chain->ogg,
"created new ogg src %p for stream with serial %08x", ret, serialno);
"created new ogg src %p for stream with serial %08lx", ret, serialno);
g_array_append_val (chain->streams, ret);
......@@ -2258,7 +2258,7 @@ gst_ogg_demux_read_chain (GstOggDemux * ogg)
serial = ogg_page_serialno (&op);
if (gst_ogg_chain_get_stream (chain, serial) != NULL) {
GST_WARNING_OBJECT (ogg, "found serial %08x BOS page twice, ignoring",
GST_WARNING_OBJECT (ogg, "found serial %08lx BOS page twice, ignoring",
serial);
continue;
}
......@@ -2309,7 +2309,7 @@ gst_ogg_demux_read_chain (GstOggDemux * ogg)
}
/* the timestamp will be filled in when we submit the pages */
done &= (pad->start_time != GST_CLOCK_TIME_NONE);
GST_LOG_OBJECT (ogg, "done %08x now %d", serial, done);
GST_LOG_OBJECT (ogg, "done %08lx now %d", serial, done);
}
/* we read a page not belonging to the current chain: seek back to the
......
......@@ -1118,8 +1118,8 @@ gst_ogg_mux_send_headers (GstOggMux * mux)
/* if last header, flush page */
if (hwalk == NULL) {
GST_LOG_OBJECT (mux,
"flushing page as packet %d is first or last packet",
pad->packetno);
"flushing page as packet %" G_GUINT64_FORMAT " is first or "
"last packet", pad->packetno);
while (ogg_stream_flush (&pad->stream, &page)) {
GstBuffer *hbuf = gst_ogg_mux_buffer_from_page (mux, &page, FALSE);
......@@ -1400,7 +1400,7 @@ gst_ogg_mux_process_best_pad (GstOggMux * ogg_mux, GstOggPad * best)
GST_TIME_FORMAT ") caused new page",
granulepos, packet.packetno, GST_TIME_ARGS (timestamp));
GST_LOG_OBJECT (pad->collect.pad,
GST_GP_FORMAT " new page %d", ogg_page_granulepos (&page),
GST_GP_FORMAT " new page %ld", ogg_page_granulepos (&page),
pad->stream.pageno);
if (ogg_page_granulepos (&page) == granulepos) {
......
......@@ -147,7 +147,7 @@ gst_ogg_parse_new_stream (GstOggParse * parser, guint32 serialno)
{
GstOggStream *ret;
GST_DEBUG_OBJECT (parser, "creating new stream %08lx", serialno);
GST_DEBUG_OBJECT (parser, "creating new stream %08x", serialno);
ret = g_new0 (GstOggStream, 1);
......@@ -410,8 +410,10 @@ gst_ogg_parse_chain (GstPad * pad, GstBuffer * buffer)
serialno = ogg_page_serialno (&page);
GST_LOG_OBJECT (ogg,
"processing ogg page (serial %08lx, pageno %ld, granule pos %llu, "
"bos %d, offset %llu-%llu)", serialno, ogg_page_pageno (&page),
"processing ogg page (serial %08x, pageno %ld, "
"granule pos %" G_GUINT64_FORMAT ", bos %d, offset %"
G_GUINT64_FORMAT "-%" G_GUINT64_FORMAT ")",
serialno, ogg_page_pageno (&page),
granule, bos, startoffset, ogg->offset);
if (ogg_page_bos (&page)) {
......
......@@ -570,7 +570,7 @@ gst_ogm_parse_stream_header (GstOgmParse * ogm, const guint8 * data, guint size)
GST_LOG_OBJECT (ogm, "Type: %s, subtype: 0x%04x, channels: %d, "
"samplerate: %d, blockalign: %d, bps: %d, caps = %" GST_PTR_FORMAT,
ogm->hdr.streamtype, codec_id, ogm->hdr.s.audio.channels,
ogm->hdr.samples_per_unit, ogm->hdr.s.audio.blockalign,
(gint) ogm->hdr.samples_per_unit, ogm->hdr.s.audio.blockalign,
ogm->hdr.s.audio.avgbytespersec, caps);
break;
}
......
......@@ -997,7 +997,7 @@ vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet)
if (vd->cur_timestamp != GST_CLOCK_TIME_NONE) {
GST_BUFFER_TIMESTAMP (out) = vd->cur_timestamp;
GST_DEBUG_OBJECT (vd,
"cur_timestamp: %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT " = % "
"cur_timestamp: %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT " = %"
GST_TIME_FORMAT, GST_TIME_ARGS (vd->cur_timestamp),
GST_TIME_ARGS (GST_BUFFER_DURATION (out)),
GST_TIME_ARGS (vd->cur_timestamp + GST_BUFFER_DURATION (out)));
......
......@@ -787,7 +787,7 @@ gst_vorbis_enc_setup (GstVorbisEnc * vorbisenc)
max_bitrate, vorbisenc->bitrate, min_bitrate) != 0) {
GST_ERROR_OBJECT (vorbisenc,
"vorbis_encode_setup_managed "
"(c %d, rate %d, max br %ld, br %ld, min br %ld) failed",
"(c %d, rate %d, max br %ld, br %d, min br %ld) failed",
vorbisenc->channels, vorbisenc->frequency, max_bitrate,
vorbisenc->bitrate, min_bitrate);
vorbis_info_clear (&vorbisenc->vi);
......@@ -1012,8 +1012,8 @@ gst_vorbis_enc_buffer_check_discontinuous (GstVorbisEnc * vorbisenc,
if ((GstClockTimeDiff) (GST_BUFFER_TIMESTAMP (buffer) -
vorbisenc->expected_ts) > halfsample) {
GST_DEBUG_OBJECT (vorbisenc, "Expected TS % " GST_TIME_FORMAT
", buffer TS % " GST_TIME_FORMAT,
GST_DEBUG_OBJECT (vorbisenc, "Expected TS %" GST_TIME_FORMAT
", buffer TS %" GST_TIME_FORMAT,
GST_TIME_ARGS (vorbisenc->expected_ts),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
ret = TRUE;
......
......@@ -580,8 +580,8 @@ vorbis_parse_src_query (GstPad * pad, GstQuery * query)
gst_query_set_position (query, format, value);
GST_LOG_OBJECT (parse,
"query %u: peer returned granulepos: %llu - we return %llu (format %u)",
GST_LOG_OBJECT (parse, "query %p: peer returned granulepos: %"
G_GUINT64_FORMAT " - we return %" G_GUINT64_FORMAT " (format %u)",
query, granulepos, value, format);
break;
......
......@@ -618,7 +618,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
gst_util_uint64_scale_int (render_time, ringbuf->spec.rate, GST_SECOND);
GST_DEBUG_OBJECT (sink, "render time %" GST_TIME_FORMAT
", render offset %llu, samples %lu",
", render offset %llu, samples %u",
GST_TIME_ARGS (render_time), render_offset, samples);
/* never try to align samples when we are slaved to another clock, just
......@@ -651,7 +651,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
* non-discont should be aligned by definition. */
if (G_LIKELY (diff < ringbuf->spec.rate / DIFF_TOLERANCE)) {
GST_DEBUG_OBJECT (sink,
"align with prev sample, %" G_GINT64_FORMAT " < %lu", diff,
"align with prev sample, %" G_GINT64_FORMAT " < %d", diff,
ringbuf->spec.rate / DIFF_TOLERANCE);
/* just align with previous sample then */
render_offset = sink->next_sample;
......
......@@ -810,7 +810,7 @@ gst_cdda_base_src_handle_track_seek (GstCddaBaseSrc * src, gdouble rate,
return FALSE;
}
GST_DEBUG_OBJECT (src, "seeking to track %d", start + 1);
GST_DEBUG_OBJECT (src, "seeking to track %" G_GINT64_FORMAT, start + 1);
src->cur_sector = src->tracks[start].start;
GST_DEBUG_OBJECT (src, "starting at sector %d", src->cur_sector);
......
......@@ -359,8 +359,9 @@ gst_base_rtp_depayload_push_full (GstBaseRTPDepayload * filter,
bclass->set_gst_timestamp (filter, timestamp, out_buf);
/* push it */
GST_LOG_OBJECT (filter, "Pushing buffer size %d, timestamp %u",
GST_BUFFER_SIZE (out_buf), GST_BUFFER_TIMESTAMP (out_buf));
GST_LOG_OBJECT (filter, "Pushing buffer size %d, timestamp %" GST_TIME_FORMAT,
GST_BUFFER_SIZE (out_buf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out_buf)));
ret = gst_pad_push (filter->srcpad, out_buf);
GST_LOG_OBJECT (filter, "Pushed buffer: %s", gst_flow_get_name (ret));
......
......@@ -404,7 +404,7 @@ gst_basertppayload_push (GstBaseRTPPayload * payload, GstBuffer * buffer)
ts += rtime;
}
GST_LOG_OBJECT (payload, "setting RTP timestamp %lu", ts);
GST_LOG_OBJECT (payload, "setting RTP timestamp %u", (guint) ts);
gst_rtp_buffer_set_timestamp (buffer, ts);
payload->timestamp = ts;
......
......@@ -155,7 +155,7 @@ resample_input_pushthrough (ResampleState * r)
buffer = audioresample_buffer_new_and_alloc (filter_bytes / 2);
memset (buffer->data, 0, buffer->length);
RESAMPLE_DEBUG ("pushthrough", buffer->length);
RESAMPLE_DEBUG ("pushthrough %u", buffer->length);
audioresample_buffer_queue_push (r->queue, buffer);
}
......
......@@ -656,7 +656,7 @@ queue_out_of_data (GstElement * queue, GstPlayBaseBin * play_base_bin)
id = gst_pad_add_buffer_probe (sinkpad, G_CALLBACK (check_queue), queue);
g_object_set_data (G_OBJECT (queue), "probe", GINT_TO_POINTER (id));
GST_DEBUG_OBJECT (play_base_bin,
"Re-attaching buffering probe to pad %s:%s",
"Re-attaching buffering probe to pad %s:%s %p",
GST_DEBUG_PAD_NAME (sinkpad), sinkpad);
gst_object_unref (sinkpad);
......
......@@ -2248,7 +2248,7 @@ gst_multi_fd_sink_handle_clients (GstMultiFdSink * sink)
continue;
}
if (gst_fdset_fd_has_error (sink->fdset, &client->fd)) {
GST_WARNING_OBJECT (sink, "gst_fdset_fd_has_error for %d", client->fd);
GST_WARNING_OBJECT (sink, "gst_fdset_fd_has_error for %d", client->fd.fd);
client->status = GST_CLIENT_STATUS_ERROR;
gst_multi_fd_sink_remove_client_link (sink, clients);
continue;
......
......@@ -727,8 +727,8 @@ mp3_type_find_at_offset (GstTypeFind * tf, guint64 start_off,
GST_INFO
("audio/mpeg calculated %u = %u * %u / %u * (%u - %u) / %u",
probability, GST_TYPE_FIND_MAXIMUM, found,
GST_MP3_TYPEFIND_TRY_HEADERS, GST_MP3_TYPEFIND_TRY_SYNC, skipped,
GST_MP3_TYPEFIND_TRY_SYNC);
GST_MP3_TYPEFIND_TRY_HEADERS, GST_MP3_TYPEFIND_TRY_SYNC,
(guint) skipped, GST_MP3_TYPEFIND_TRY_SYNC);
/* make sure we're not id3 tagged */
head_data = gst_type_find_peek (tf, -128, 3);
if (head_data && (memcmp (head_data, "TAG", 3) == 0)) {
......@@ -906,7 +906,8 @@ wavpack_type_find (GstTypeFind * tf, gpointer unused)
sublen += 1 + 1; /* id + length */
}
if (sublen > blocksize - offset + 32) {
GST_LOG ("chunk length too big (%u > %u)", sublen, blocksize - offset);
GST_LOG ("chunk length too big (%u > %" G_GUINT64_FORMAT ")", sublen,
blocksize - offset);
break;
}
if ((data[0] & 0x20) == 0) {
......
......@@ -529,8 +529,8 @@ gst_video_test_src_create (GstPushSrc * psrc, GstBuffer ** buffer)
g_return_val_if_fail (newsize > 0, GST_FLOW_ERROR);
GST_LOG_OBJECT (src,
"creating buffer of %ld bytes with %dx%d image for frame %d", newsize,
src->width, src->height, src->n_frames);
"creating buffer of %lu bytes with %dx%d image for frame %d", newsize,
src->width, src->height, (gint) src->n_frames);
#ifdef USE_PEER_BUFFERALLOC
res =
......@@ -575,7 +575,7 @@ not_negotiated:
}
eos:
{
GST_DEBUG_OBJECT (src, "eos: 0 framerate and frame %d", src->n_frames);
GST_DEBUG_OBJECT (src, "eos: 0 framerate, frame %d", (gint) src->n_frames);
return GST_FLOW_UNEXPECTED;
}
no_buffer:
......
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