Commit 97426a1c authored by Edward Hervey's avatar Edward Hervey

all: Fix for GST_DISABLE_GST_DEBUG

Where applicable, remove methods that don't do anything different than
the default implementation.
parent bc84cd77
......@@ -227,11 +227,13 @@ static void gst_dash_demux_remove_streams (GstDashDemux * demux,
GSList * streams);
static void gst_dash_demux_stream_free (GstDashDemuxStream * stream);
static void gst_dash_demux_reset (GstDashDemux * demux, gboolean dispose);
#ifndef GST_DISABLE_GST_DEBUG
static GstClockTime gst_dash_demux_get_buffering_time (GstDashDemux * demux);
static GstCaps *gst_dash_demux_get_input_caps (GstDashDemux * demux,
GstActiveStream * stream);
static GstClockTime gst_dash_demux_stream_get_buffering_time (GstDashDemuxStream
* stream);
#endif
static GstCaps *gst_dash_demux_get_input_caps (GstDashDemux * demux,
GstActiveStream * stream);
static GstPad *gst_dash_demux_create_pad (GstDashDemux * demux);
#define gst_dash_demux_parent_class parent_class
......@@ -1409,6 +1411,7 @@ gst_dash_demux_reset (GstDashDemux * demux, gboolean dispose)
demux->cancelled = FALSE;
}
#ifndef GST_DISABLE_GST_DEBUG
static GstClockTime
gst_dash_demux_get_buffering_time (GstDashDemux * demux)
{
......@@ -1436,6 +1439,7 @@ gst_dash_demux_stream_get_buffering_time (GstDashDemuxStream * stream)
return (GstClockTime) level.time;
}
#endif
static gboolean
gst_dash_demux_all_streams_have_data (GstDashDemux * demux)
......@@ -2186,12 +2190,16 @@ gst_dash_demux_get_next_fragment (GstDashDemux * demux,
/* Wake the download task up */
GST_TASK_SIGNAL (demux->download_task);
if (selected_stream) {
#ifndef GST_DISABLE_GST_DEBUG
guint64 brate;
#endif
diff = (GST_TIMEVAL_TO_TIME (now) - GST_TIMEVAL_TO_TIME (start));
gst_download_rate_add_rate (&selected_stream->dnl_rate, size_buffer, diff);
#ifndef GST_DISABLE_GST_DEBUG
brate = (size_buffer * 8) / ((double) diff / GST_SECOND);
#endif
GST_INFO_OBJECT (demux,
"Stream: %d Download rate = %" PRIu64 " Kbits/s (%" PRIu64
" Ko in %.2f s)", selected_stream->index,
......
......@@ -699,15 +699,14 @@ gst_mpdparser_get_xml_prop_dateTime (xmlNode * a_node,
{
xmlChar *prop_string;
gchar *str;
gint ret, len, pos;
gint ret, pos;
gint year, month, day, hour, minute, second;
gboolean exists = FALSE;
prop_string = xmlGetProp (a_node, (const xmlChar *) property_name);
if (prop_string) {
len = xmlStrlen (prop_string);
str = (gchar *) prop_string;
GST_TRACE ("dateTime: %s, len %d", str, len);
GST_TRACE ("dateTime: %s, len %d", str, xmlStrlen (prop_string));
/* parse year */
ret = sscanf (str, "%d", &year);
if (ret != 1)
......@@ -2023,10 +2022,12 @@ gst_mpdparser_get_first_adapt_set_with_mimeType_and_lang (GList *
gchar *this_mimeType = NULL;
rep =
gst_mpdparser_get_lowest_representation (adapt_set->Representations);
#ifndef GST_DISABLE_GST_DEBUG
if (rep && rep->BaseURLs) {
GstBaseURL *url = rep->BaseURLs->data;
GST_DEBUG ("%s", url->baseURL);
}
#endif
if (rep->RepresentationBase)
this_mimeType = rep->RepresentationBase->mimeType;
if (!this_mimeType && adapt_set->RepresentationBase) {
......
......@@ -106,6 +106,7 @@ got_egl_error (const char *wtf)
void
gst_egl_adaptation_init_egl_exts (GstEglAdaptationContext * ctx)
{
#ifndef GST_DISABLE_GST_DEBUG
const char *eglexts;
unsigned const char *glexts;
......@@ -116,7 +117,7 @@ gst_egl_adaptation_init_egl_exts (GstEglAdaptationContext * ctx)
GST_STR_NULL (eglexts));
GST_DEBUG_OBJECT (ctx->element, "Available GLES extensions: %s\n",
GST_STR_NULL ((const char *) glexts));
#endif
return;
}
......
......@@ -1052,7 +1052,10 @@ static gboolean
gst_eglglessink_fill_texture (GstEglGlesSink * eglglessink, GstBuffer * buf)
{
GstVideoFrame vframe;
gint w, h;
#ifndef GST_DISABLE_GST_DEBUG
gint w;
#endif
gint h;
memset (&vframe, 0, sizeof (vframe));
......@@ -1061,8 +1064,9 @@ gst_eglglessink_fill_texture (GstEglGlesSink * eglglessink, GstBuffer * buf)
GST_ERROR_OBJECT (eglglessink, "Couldn't map frame");
goto HANDLE_ERROR;
}
#ifndef GST_DISABLE_GST_DEBUG
w = GST_VIDEO_FRAME_WIDTH (&vframe);
#endif
h = GST_VIDEO_FRAME_HEIGHT (&vframe);
GST_DEBUG_OBJECT (eglglessink,
......
......@@ -126,7 +126,6 @@ static gboolean gst_faac_configure_source_pad (GstFaac * faac,
GstAudioInfo * info);
static GstCaps *gst_faac_getcaps (GstAudioEncoder * enc, GstCaps * filter);
static gboolean gst_faac_start (GstAudioEncoder * enc);
static gboolean gst_faac_stop (GstAudioEncoder * enc);
static gboolean gst_faac_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
......@@ -207,7 +206,6 @@ gst_faac_class_init (GstFaacClass * klass)
"Free MPEG-2/4 AAC encoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
base_class->start = GST_DEBUG_FUNCPTR (gst_faac_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_faac_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_faac_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_faac_handle_frame);
......@@ -259,15 +257,6 @@ gst_faac_close_encoder (GstFaac * faac)
faac->handle = NULL;
}
static gboolean
gst_faac_start (GstAudioEncoder * enc)
{
GstFaac *faac = GST_FAAC (enc);
GST_DEBUG_OBJECT (faac, "start");
return TRUE;
}
static gboolean
gst_faac_stop (GstAudioEncoder * enc)
{
......
......@@ -460,7 +460,6 @@ gst_flups_demux_send_data (GstFluPSDemux * demux, GstFluPSStream * stream,
{
GstFlowReturn result;
GstClockTime pts = GST_CLOCK_TIME_NONE, dts = GST_CLOCK_TIME_NONE;
guint size;
if (stream == NULL)
goto no_stream;
......@@ -497,16 +496,15 @@ gst_flups_demux_send_data (GstFluPSDemux * demux, GstFluPSStream * stream,
stream->discont = FALSE;
}
size = gst_buffer_get_size (buf);
demux->next_pts = G_MAXUINT64;
demux->next_dts = G_MAXUINT64;
result = gst_pad_push (stream->pad, buf);
GST_DEBUG_OBJECT (demux, "pushed stream id 0x%02x type 0x%02x, pts time: %"
GST_TIME_FORMAT ", size %d. result: %s",
GST_TIME_FORMAT ", size %" G_GSIZE_FORMAT ". result: %s",
stream->id, stream->type, GST_TIME_ARGS (pts),
size, gst_flow_get_name (result));
gst_buffer_get_size (buf), gst_flow_get_name (result));
return result;
......@@ -1525,7 +1523,9 @@ gst_flups_demux_parse_sys_head (GstFluPSDemux * demux)
{
guint16 length;
const guint8 *data;
#ifndef GST_DISABLE_GST_DEBUG
gboolean csps;
#endif
if (gst_adapter_available (demux->adapter) < 6)
goto need_more_data;
......@@ -1572,6 +1572,7 @@ gst_flups_demux_parse_sys_head (GstFluPSDemux * demux)
/* audio_bound:6==1 ! fixed:1 | constrained:1 */
{
#ifndef GST_DISABLE_GST_DEBUG
guint8 audio_bound;
gboolean fixed;
......@@ -1584,36 +1585,40 @@ gst_flups_demux_parse_sys_head (GstFluPSDemux * demux)
GST_DEBUG_OBJECT (demux, "audio_bound %d, fixed %d, constrained %d",
audio_bound, fixed, csps);
#endif
data += 1;
}
/* audio_lock:1 | video_lock:1 | marker:1==1 | video_bound:5 */
{
#ifndef GST_DISABLE_GST_DEBUG
gboolean audio_lock;
gboolean video_lock;
guint8 video_bound;
audio_lock = (data[0] & 0x80) == 0x80;
video_lock = (data[0] & 0x40) == 0x40;
#endif
if ((data[0] & 0x20) != 0x20)
goto marker_expected;
#ifndef GST_DISABLE_GST_DEBUG
/* max number of simultaneous video streams active */
video_bound = (data[0] & 0x1f);
GST_DEBUG_OBJECT (demux, "audio_lock %d, video_lock %d, video_bound %d",
audio_lock, video_lock, video_bound);
#endif
data += 1;
}
/* packet_rate_restriction:1 | reserved:7==0x7F */
{
#ifndef GST_DISABLE_GST_DEBUG
gboolean packet_rate_restriction;
#endif
if ((data[0] & 0x7f) != 0x7f)
goto marker_expected;
#ifndef GST_DISABLE_GST_DEBUG
/* only valid if csps is set */
if (csps) {
packet_rate_restriction = (data[0] & 0x80) == 0x80;
......@@ -1621,6 +1626,7 @@ gst_flups_demux_parse_sys_head (GstFluPSDemux * demux)
GST_DEBUG_OBJECT (demux, "packet_rate_restriction %d",
packet_rate_restriction);
}
#endif
}
data += 1;
......@@ -1632,10 +1638,11 @@ gst_flups_demux_parse_sys_head (GstFluPSDemux * demux)
for (i = 0; i < stream_count; i++) {
guint8 stream_id;
#ifndef GST_DISABLE_GST_DEBUG
gboolean STD_buffer_bound_scale;
guint16 STD_buffer_size_bound;
guint32 buf_byte_size_bound;
#endif
stream_id = *data++;
if (!(stream_id & 0x80))
goto sys_len_error;
......@@ -1643,7 +1650,7 @@ gst_flups_demux_parse_sys_head (GstFluPSDemux * demux)
/* check marker bits */
if ((*data & 0xC0) != 0xC0)
goto no_placeholder_bits;
#ifndef GST_DISABLE_GST_DEBUG
STD_buffer_bound_scale = *data & 0x20;
STD_buffer_size_bound = ((guint16) (*data++ & 0x1F)) << 8;
STD_buffer_size_bound |= *data++;
......@@ -1653,7 +1660,7 @@ gst_flups_demux_parse_sys_head (GstFluPSDemux * demux)
} else {
buf_byte_size_bound = STD_buffer_size_bound * 1024;
}
#endif
GST_DEBUG_OBJECT (demux, "STD_buffer_bound_scale %d",
STD_buffer_bound_scale);
GST_DEBUG_OBJECT (demux, "STD_buffer_size_bound %d or %d bytes",
......@@ -1700,7 +1707,9 @@ gst_flups_demux_parse_psm (GstFluPSDemux * demux)
guint16 length = 0, info_length = 0, es_map_length = 0;
guint8 psm_version = 0;
const guint8 *data, *es_map_base;
#ifndef GST_DISABLE_GST_DEBUG
gboolean applicable;
#endif
if (gst_adapter_available (demux->adapter) < 6)
goto need_more_data;
......@@ -1731,7 +1740,9 @@ gst_flups_demux_parse_psm (GstFluPSDemux * demux)
/* Read PSM applicable bit together with version */
psm_version = GST_READ_UINT8 (data);
#ifndef GST_DISABLE_GST_DEBUG
applicable = (psm_version & 0x80) >> 7;
#endif
psm_version &= 0x1F;
GST_DEBUG_OBJECT (demux, "PSM version %u (applicable now %u)", psm_version,
applicable);
......@@ -1835,8 +1846,6 @@ gst_flups_demux_data_cb (GstPESFilter * filter, gboolean first,
if (stream_type == -1) {
/* no stream type, if PS1, get the new id */
if (start_code == ID_PRIVATE_STREAM_1 && datalen >= 2) {
guint8 nframes;
/* VDR writes A52 streams without any header bytes
* (see ftp://ftp.mplayerhq.hu/MPlayer/samples/MPEG-VOB/vdr-AC3) */
if (datalen >= 4) {
......@@ -1861,8 +1870,13 @@ gst_flups_demux_data_cb (GstPESFilter * filter, gboolean first,
* take the first byte too, since it's the frame count in audio
* streams and our backwards compat convention is to strip it off */
if (stream_type != ST_PS_DVD_SUBPICTURE) {
#ifndef GST_DISABLE_GST_DEBUG
guint8 nframes;
/* Number of audio frames in this packet */
nframes = map.data[offset++];
nframes = map.data[offset];
#endif
offset++;
datalen--;
GST_DEBUG_OBJECT (demux, "private type 0x%02x, %d frames", id,
nframes);
......
......@@ -36,14 +36,19 @@ gst_mpeg_descriptor_free (GstMPEGDescriptor * desc)
static guint
gst_mpeg_descriptor_parse_1 (guint8 * data, guint size)
{
#ifndef GST_DISABLE_GST_DEBUG
guint8 tag;
#endif
guint8 length;
/* need at least 2 bytes for tag and length */
if (size < 2)
return 0;
tag = *data++;
#ifndef GST_DISABLE_GST_DEBUG
tag = *data;
#endif
data += 1;
length = *data++;
size -= 2;
......
......@@ -458,9 +458,11 @@ gst_pes_filter_parse (GstPESFilter * filter)
push_out:
{
GstBuffer *out;
#ifndef GST_DISABLE_GST_DEBUG
guint16 consumed;
consumed = avail - 6 - datalen;
#endif
if (filter->unbounded_packet == FALSE) {
filter->length -= avail - 6;
......
......@@ -293,13 +293,12 @@ static GstFlowReturn
gst_rsvg_dec_parse (GstVideoDecoder * decoder, GstVideoCodecFrame * frame,
GstAdapter * adapter, gboolean at_eos)
{
GstRsvgDec *rsvg = GST_RSVG_DEC (decoder);
gboolean completed = FALSE;
const guint8 *data;
guint size;
guint i;
GST_LOG_OBJECT (rsvg, "parse start");
GST_LOG_OBJECT (decoder, "parse start");
size = gst_adapter_available (adapter);
/* "<svg></svg>" */
......@@ -308,7 +307,7 @@ gst_rsvg_dec_parse (GstVideoDecoder * decoder, GstVideoCodecFrame * frame,
data = gst_adapter_map (adapter, size);
if (data == NULL) {
GST_ERROR_OBJECT (rsvg, "Unable to map memory");
GST_ERROR_OBJECT (decoder, "Unable to map memory");
return GST_FLOW_ERROR;
}
for (i = 0; i < size - 4; i++) {
......@@ -320,7 +319,7 @@ gst_rsvg_dec_parse (GstVideoDecoder * decoder, GstVideoCodecFrame * frame,
return GST_VIDEO_DECODER_FLOW_NEED_DATA;
data = gst_adapter_map (adapter, size);
if (data == NULL) {
GST_ERROR_OBJECT (rsvg, "Unable to map memory");
GST_ERROR_OBJECT (decoder, "Unable to map memory");
return GST_FLOW_ERROR;
}
break;
......@@ -342,7 +341,7 @@ gst_rsvg_dec_parse (GstVideoDecoder * decoder, GstVideoCodecFrame * frame,
if (completed) {
GST_LOG_OBJECT (rsvg, "have complete svg of %u bytes", size);
GST_LOG_OBJECT (decoder, "have complete svg of %u bytes", size);
gst_video_decoder_add_to_frame (decoder, size);
return gst_video_decoder_have_frame (decoder);
......
......@@ -1106,8 +1106,10 @@ gst_mss_demux_stream_download_fragment (GstMssDemuxStream * stream,
after_download = g_get_real_time ();
if (_buffer) {
#ifndef GST_DISABLE_GST_DEBUG
guint64 bitrate = (8 * gst_buffer_get_size (_buffer) * 1000000LLU) /
(after_download - before_download);
#endif
GST_DEBUG_OBJECT (mssdemux,
"Measured download bitrate: %s %" G_GUINT64_FORMAT " bps",
......
......@@ -84,7 +84,6 @@ static GstCaps *gst_wayland_sink_get_caps (GstBaseSink * bsink,
GstCaps * filter);
static gboolean gst_wayland_sink_set_caps (GstBaseSink * bsink, GstCaps * caps);
static gboolean gst_wayland_sink_start (GstBaseSink * bsink);
static gboolean gst_wayland_sink_stop (GstBaseSink * bsink);
static gboolean gst_wayland_sink_preroll (GstBaseSink * bsink,
GstBuffer * buffer);
static gboolean
......@@ -130,6 +129,7 @@ gst_wayland_format_to_wl_format (GstVideoFormat format)
return -1;
}
#ifndef GST_DISABLE_GST_DEBUG
static const gchar *
gst_wayland_format_to_string (uint32_t wl_format)
{
......@@ -142,6 +142,7 @@ gst_wayland_format_to_string (uint32_t wl_format)
return gst_video_format_to_string (format);
}
#endif
static void
gst_wayland_sink_class_init (GstWaylandSinkClass * klass)
......@@ -169,7 +170,6 @@ gst_wayland_sink_class_init (GstWaylandSinkClass * klass)
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_wayland_sink_get_caps);
gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_wayland_sink_set_caps);
gstbasesink_class->start = GST_DEBUG_FUNCPTR (gst_wayland_sink_start);
gstbasesink_class->stop = GST_DEBUG_FUNCPTR (gst_wayland_sink_stop);
gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_wayland_sink_preroll);
gstbasesink_class->propose_allocation =
GST_DEBUG_FUNCPTR (gst_wayland_sink_propose_allocation);
......@@ -527,16 +527,6 @@ gst_wayland_sink_start (GstBaseSink * bsink)
return result;
}
static gboolean
gst_wayland_sink_stop (GstBaseSink * bsink)
{
GstWaylandSink *sink = (GstWaylandSink *) bsink;
GST_DEBUG_OBJECT (sink, "stop");
return TRUE;
}
static gboolean
gst_wayland_sink_propose_allocation (GstBaseSink * bsink, GstQuery * query)
{
......
......@@ -1968,7 +1968,9 @@ gst_h264_parser_parse_sei (GstH264NalParser * nalparser, GstH264NalUnit * nalu,
guint32 payloadSize;
guint8 payload_type_byte, payload_size_byte;
#ifndef GST_DISABLE_GST_DEBUG
guint remaining, payload_size;
#endif
GstH264ParserResult res;
GST_DEBUG ("parsing \"Sei message\"");
......@@ -1991,11 +1993,13 @@ gst_h264_parser_parse_sei (GstH264NalParser * nalparser, GstH264NalUnit * nalu,
}
while (payload_size_byte == 0xff);
#ifndef GST_DISABLE_GST_DEBUG
remaining = nal_reader_get_remaining (&nr) * 8;
payload_size = payloadSize < remaining ? payloadSize : remaining;
GST_DEBUG ("SEI message received: payloadType %u, payloadSize = %u bytes",
sei->payloadType, payload_size);
#endif
if (sei->payloadType == GST_H264_SEI_BUF_PERIOD) {
/* size not set; might depend on emulation_prevention_three_byte */
......
......@@ -51,8 +51,6 @@ static void gst_audio_channel_mix_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_channel_mix_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_audio_channel_mix_dispose (GObject * object);
static void gst_audio_channel_mix_finalize (GObject * object);
static gboolean gst_audio_channel_mix_setup (GstAudioFilter * filter,
const GstAudioInfo * info);
......@@ -115,8 +113,6 @@ gst_audio_channel_mix_class_init (GstAudioChannelMixClass * klass)
gobject_class->set_property = gst_audio_channel_mix_set_property;
gobject_class->get_property = gst_audio_channel_mix_get_property;
gobject_class->dispose = gst_audio_channel_mix_dispose;
gobject_class->finalize = gst_audio_channel_mix_finalize;
audio_filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_channel_mix_setup);
base_transform_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_channel_mix_transform_ip);
......@@ -209,36 +205,14 @@ gst_audio_channel_mix_get_property (GObject * object, guint property_id,
}
}
void
gst_audio_channel_mix_dispose (GObject * object)
{
GstAudioChannelMix *audiochannelmix = GST_AUDIO_CHANNEL_MIX (object);
GST_DEBUG_OBJECT (audiochannelmix, "dispose");
/* clean up as possible. may be called multiple times */
G_OBJECT_CLASS (gst_audio_channel_mix_parent_class)->dispose (object);
}
void
gst_audio_channel_mix_finalize (GObject * object)
{
GstAudioChannelMix *audiochannelmix = GST_AUDIO_CHANNEL_MIX (object);
GST_DEBUG_OBJECT (audiochannelmix, "finalize");
/* clean up object here */
G_OBJECT_CLASS (gst_audio_channel_mix_parent_class)->finalize (object);
}
static gboolean
gst_audio_channel_mix_setup (GstAudioFilter * filter, const GstAudioInfo * info)
{
#ifndef GST_DISABLE_GST_DEBUG
GstAudioChannelMix *audiochannelmix = GST_AUDIO_CHANNEL_MIX (filter);
GST_DEBUG_OBJECT (audiochannelmix, "setup");
#endif
return TRUE;
}
......
......@@ -59,11 +59,7 @@ static void gst_watchdog_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static void gst_watchdog_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_watchdog_dispose (GObject * object);
static void gst_watchdog_finalize (GObject * object);
static GstCaps *gst_watchdog_transform_caps (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * caps, GstCaps * filter);
static gboolean gst_watchdog_start (GstBaseTransform * trans);
static gboolean gst_watchdog_stop (GstBaseTransform * trans);
static gboolean gst_watchdog_sink_event (GstBaseTransform * trans,
......@@ -105,10 +101,6 @@ gst_watchdog_class_init (GstWatchdogClass * klass)
gobject_class->set_property = gst_watchdog_set_property;
gobject_class->get_property = gst_watchdog_get_property;
gobject_class->dispose = gst_watchdog_dispose;
gobject_class->finalize = gst_watchdog_finalize;
base_transform_class->transform_caps =
GST_DEBUG_FUNCPTR (gst_watchdog_transform_caps);
base_transform_class->start = GST_DEBUG_FUNCPTR (gst_watchdog_start);
base_transform_class->stop = GST_DEBUG_FUNCPTR (gst_watchdog_stop);
base_transform_class->sink_event =
......@@ -166,41 +158,6 @@ gst_watchdog_get_property (GObject * object, guint property_id,
}
}
void
gst_watchdog_dispose (GObject * object)
{
GstWatchdog *watchdog = GST_WATCHDOG (object);
GST_DEBUG_OBJECT (watchdog, "dispose");
/* clean up as possible. may be called multiple times */
G_OBJECT_CLASS (gst_watchdog_parent_class)->dispose (object);
}
void
gst_watchdog_finalize (GObject * object)
{
GstWatchdog *watchdog = GST_WATCHDOG (object);
GST_DEBUG_OBJECT (watchdog, "finalize");
/* clean up object here */
G_OBJECT_CLASS (gst_watchdog_parent_class)->finalize (object);
}
static GstCaps *
gst_watchdog_transform_caps (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * caps, GstCaps * filter)
{
GstWatchdog *watchdog = GST_WATCHDOG (trans);
GST_DEBUG_OBJECT (watchdog, "transform_caps");
return gst_caps_ref (caps);
}
static gpointer
gst_watchdog_thread (gpointer user_data)
{
......
......@@ -788,22 +788,19 @@ gst_dvd_spu_advance_spu (GstDVDSpu * dvdspu, GstClockTime new_ts)
GST_TIME_ARGS (state->next_ts), GST_TIME_ARGS (new_ts));
if (!gstspu_execute_event (dvdspu)) {
GstClockTime vid_run_ts;
/* No current command buffer, try and get one */
SpuPacket *packet = (SpuPacket *) g_queue_pop_head (dvdspu->pending_spus);
if (packet == NULL)
return; /* No SPU packets available */
vid_run_ts =
gst_segment_to_running_time (&dvdspu->video_seg, GST_FORMAT_TIME,
dvdspu->video_seg.position);
GST_LOG_OBJECT (dvdspu,
"Popped new SPU packet with TS %" GST_TIME_FORMAT
". Video position=%" GST_TIME_FORMAT " (%" GST_TIME_FORMAT
") type %s",
GST_TIME_ARGS (packet->event_ts), GST_TIME_ARGS (vid_run_ts),
GST_TIME_ARGS (packet->event_ts),
GST_TIME_ARGS (gst_segment_to_running_time (&dvdspu->video_seg,
GST_FORMAT_TIME, dvdspu->video_seg.position)),
GST_TIME_ARGS (dvdspu->video_seg.position),
packet->buf ? "buffer" : "event");
......
......@@ -234,9 +234,7 @@ static void
gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "get_times");
GST_DEBUG_OBJECT (src, "get_times");
/* for live sources, sync on the timestamp of the buffer */
if (gst_base_src_is_live (src)) {
......@@ -325,10 +323,9 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
static gboolean
gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
gboolean ret;
GST_DEBUG_OBJECT (interaudiosrc, "query");
GST_DEBUG_OBJECT (src,