[Swfdec] 6 commits - configure.ac libswfdec/swfdec_codec_audio.c libswfdec/swfdec_codec_gst.c libswfdec/swfdec_codec_video.c player/swfplay.c
Benjamin Otte
company at kemper.freedesktop.org
Fri Oct 26 12:16:30 PDT 2007
configure.ac | 12
libswfdec/swfdec_codec_audio.c | 2
libswfdec/swfdec_codec_gst.c | 525 +++++++++++++++++++++++++----------------
libswfdec/swfdec_codec_video.c | 2
player/swfplay.c | 2
5 files changed, 336 insertions(+), 207 deletions(-)
New commits:
commit 2aa4f7f59de874bab2a74398fda09d3d387fcd30
Merge: efc1a32... e6964fd...
Author: Benjamin Otte <otte at gnome.org>
Date: Fri Oct 26 21:15:44 2007 +0200
Merge branch 'master' of ssh://company@git.freedesktop.org/git/swfdec/swfdec
commit efc1a326a7f9b9eda40f4d6d346ca221da7087ba
Author: Benjamin Otte <otte at gnome.org>
Date: Fri Oct 26 21:14:54 2007 +0200
set max-runtime to 0 in swfplay
This is so the player doesn't abort when run in gdb after having set a
breakpoint.
diff --git a/player/swfplay.c b/player/swfplay.c
index b3fcea0..5dd5f61 100644
--- a/player/swfplay.c
+++ b/player/swfplay.c
@@ -155,6 +155,8 @@ main (int argc, char *argv[])
}
loop = g_main_loop_new (NULL, TRUE);
player = swfdec_gtk_player_new (NULL);
+ /* this allows the player to continue fine when running in gdb */
+ swfdec_player_set_maximum_runtime (player, 0);
if (gc)
g_object_set (player, "memory-until-gc", (gulong) 0, NULL);
if (trace)
commit 05acb78a01ca9262744f34981b5ef54fd7b3ba28
Author: Benjamin Otte <otte at gnome.org>
Date: Fri Oct 26 21:13:20 2007 +0200
rewrite GStreamer backend to directly use the element and not create a pipeline
That was Wim's suggested method, and it works fine. The only issue right now is
that GStreamer also does the resampling, and a) audioresample is slow and
b) the whole process takes a lot of time. This causes the app to stop for half a
second when a long sample gets decoded. Not good.
diff --git a/libswfdec/swfdec_codec_gst.c b/libswfdec/swfdec_codec_gst.c
index 08cf154..bd8f95a 100644
--- a/libswfdec/swfdec_codec_gst.c
+++ b/libswfdec/swfdec_codec_gst.c
@@ -28,273 +28,400 @@
#include "swfdec_debug.h"
#include "swfdec_internal.h"
-#if 0
-#define swfdec_cond_wait(cond, mutex) G_STMT_START { \
- g_print ("waiting at %s\n", G_STRLOC); \
- g_cond_wait (cond, mutex); \
- g_print (" done at %s\n", G_STRLOC); \
-}G_STMT_END
-#else
-#define swfdec_cond_wait g_cond_wait
-#endif
+/*** BUFFER ***/
-/*** AUDIO ***/
+static void
+swfdec_gst_buffer_free (unsigned char *data, gpointer priv)
+{
+ gst_buffer_unref (priv);
+}
-typedef struct _SwfdecGstAudio SwfdecGstAudio;
-struct _SwfdecGstAudio {
- SwfdecAudioDecoder decoder;
+static SwfdecBuffer *
+swfdec_buffer_new_from_gst (GstBuffer *buffer)
+{
+ SwfdecBuffer *ret;
- GMutex * mutex; /* mutex that blocks everything below */
- GCond * cond; /* cond used to signal when stuff below changes */
- volatile int refcount; /* refcount (d'oh) */
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), NULL);
- GstElement * pipeline; /* pipeline that is playing or NULL when done */
- SwfdecBuffer * in; /* next input buffer or NULL */
- SwfdecBufferQueue * out; /* all the stored output buffers */
- GstCaps * srccaps; /* caps to set on buffers */
- gboolean eof; /* we've pushed EOF */
- gboolean done; /* TRUE after decoding stopped (error or EOF) */
-};
+ ret = swfdec_buffer_new ();
+ ret->data = GST_BUFFER_DATA (buffer);
+ ret->length = GST_BUFFER_SIZE (buffer);
+ ret->free = swfdec_gst_buffer_free;
+ ret->priv = buffer;
-static void
-swfdec_gst_audio_unref (gpointer data, GObject *unused)
+ return ret;
+}
+
+static GstBuffer *
+swfdec_gst_buffer_new (SwfdecBuffer *buffer)
{
- SwfdecGstAudio *player = data;
+ /* FIXME: make this a zero-copy operation */
+ GstBuffer *ret;
- if (!g_atomic_int_dec_and_test (&player->refcount))
- return;
- g_cond_free (player->cond);
- g_mutex_free (player->mutex);
- gst_caps_unref (player->srccaps);
- if (player->in)
- swfdec_buffer_unref (player->in);
- swfdec_buffer_queue_unref (player->out);
- g_slice_free (SwfdecGstAudio, player);
+ g_return_val_if_fail (buffer != NULL , NULL);
+
+ ret = gst_buffer_new_and_alloc (buffer->length);
+ memcpy (GST_BUFFER_DATA (ret), buffer->data, buffer->length);
+
+ return ret;
}
-static void
-swfdec_audio_decoder_gst_free (SwfdecAudioDecoder *dec)
+/*** TYPEFINDING ***/
+
+/* NB: try to mirror decodebin behavior */
+static gboolean
+swfdec_gst_feature_filter (GstPluginFeature *feature, gpointer caps)
{
- SwfdecGstAudio *player = (SwfdecGstAudio *) dec;
- GstElement *pipeline;
+ const GList *walk;
+ const gchar *klass;
- g_mutex_lock (player->mutex);
- pipeline = player->pipeline;
- player->pipeline = NULL;
- g_cond_signal (player->cond);
- g_mutex_unlock (player->mutex);
- gst_element_set_state (pipeline, GST_STATE_NULL);
- g_object_unref (pipeline);
+ /* we only care about element factories */
+ if (!GST_IS_ELEMENT_FACTORY (feature))
+ return FALSE;
+
+ /* only decoders are interesting */
+ klass = gst_element_factory_get_klass (GST_ELEMENT_FACTORY (feature));
+ if (strstr (klass, "Decoder") == NULL)
+ return FALSE;
+
+ /* only select elements with autoplugging rank */
+ if (gst_plugin_feature_get_rank (feature) < GST_RANK_MARGINAL)
+ return FALSE;
+
+ /* only care about the right sink caps */
+ for (walk = gst_element_factory_get_static_pad_templates (GST_ELEMENT_FACTORY (feature));
+ walk; walk = walk->next) {
+ GstStaticPadTemplate *template = walk->data;
+ GstCaps *intersect;
+ GstCaps *template_caps;
+
+ if (template->direction != GST_PAD_SINK)
+ continue;
+
+ template_caps = gst_static_caps_get (&template->static_caps);
+ intersect = gst_caps_intersect (caps, template_caps);
+
+ gst_caps_unref (template_caps);
+ if (gst_caps_is_empty (intersect)) {
+ gst_caps_unref (intersect);
+ } else {
+ gst_caps_unref (intersect);
+ return TRUE;
+ }
+ }
+ return FALSE;
+}
+
+static int
+swfdec_gst_compare_features (gconstpointer a_, gconstpointer b_)
+{
+ int diff;
+ GstPluginFeature *a = GST_PLUGIN_FEATURE (a_);
+ GstPluginFeature *b = GST_PLUGIN_FEATURE (b_);
- swfdec_gst_audio_unref (player, NULL);
+ diff = gst_plugin_feature_get_rank (b) - gst_plugin_feature_get_rank (a);
+ if (diff != 0)
+ return diff;
+
+ return strcmp (gst_plugin_feature_get_name (a), gst_plugin_feature_get_name (b));
}
-static void
-swfdec_audio_decoder_gst_push (SwfdecAudioDecoder *dec, SwfdecBuffer *buffer)
+static GstElement *
+swfdec_gst_get_element (GstCaps *caps)
{
- SwfdecGstAudio *player = (SwfdecGstAudio *) dec;
+ GstElement *element;
+ GList *list;
- g_mutex_lock (player->mutex);
- g_return_if_fail (!player->eof);
- while (player->in != NULL && !player->done) {
- swfdec_cond_wait (player->cond, player->mutex);
- }
- if (buffer) {
- player->in = swfdec_buffer_ref (buffer);
- } else {
- player->eof = TRUE;
- }
- g_cond_signal (player->cond);
- g_mutex_unlock (player->mutex);
+ list = gst_registry_feature_filter (gst_registry_get_default (),
+ swfdec_gst_feature_filter, FALSE, caps);
+ if (list == NULL)
+ return NULL;
+
+ list = g_list_sort (list, swfdec_gst_compare_features);
+ element = gst_element_factory_create (list->data, "decoder");
+ gst_plugin_feature_list_free (list);
+ return element;
}
-static SwfdecBuffer *
-swfdec_audio_decoder_gst_pull (SwfdecAudioDecoder *dec)
+/*** PADS ***/
+
+static GstPad *
+swfdec_gst_connect_srcpad (GstElement *element, GstCaps *caps)
{
- SwfdecGstAudio *player = (SwfdecGstAudio *) dec;
- SwfdecBuffer *buffer;
+ GstPad *srcpad, *sinkpad;
- g_mutex_lock (player->mutex);
- if (player->eof) {
- while (!player->done)
- swfdec_cond_wait (player->cond, player->mutex);
- }
- buffer = swfdec_buffer_queue_pull_buffer (player->out);
- g_mutex_unlock (player->mutex);
- return buffer;
+ sinkpad = gst_element_get_pad (element, "sink");
+ if (sinkpad == NULL)
+ return NULL;
+ srcpad = gst_pad_new ("src", GST_PAD_SRC);
+ if (!gst_pad_set_caps (srcpad, caps))
+ goto error;
+ if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
+ goto error;
+
+ gst_object_unref (sinkpad);
+ gst_pad_set_active (srcpad, TRUE);
+ return srcpad;
+
+error:
+ SWFDEC_ERROR ("failed to create or link srcpad");
+ gst_object_unref (sinkpad);
+ gst_object_unref (srcpad);
+ return NULL;
}
-static void
-swfdec_audio_decoder_gst_fakesrc_handoff (GstElement *fakesrc, GstBuffer *buf,
- GstPad *pad, SwfdecGstAudio *player)
+static GstPad *
+swfdec_gst_connect_sinkpad (GstElement *element, GstCaps *caps)
{
- g_mutex_lock (player->mutex);
- while (player->pipeline != NULL && player->in == NULL && player->eof == FALSE)
- swfdec_cond_wait (player->cond, player->mutex);
- if (player->pipeline == NULL) {
- g_mutex_unlock (player->mutex);
- return;
+ GstPad *srcpad, *sinkpad;
+
+ srcpad = gst_element_get_pad (element, "src");
+ if (srcpad == NULL)
+ return NULL;
+ sinkpad = gst_pad_new ("sink", GST_PAD_SINK);
+ if (!gst_pad_set_caps (sinkpad, caps))
+ goto error;
+ if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
+ goto error;
+
+ gst_object_unref (srcpad);
+ gst_pad_set_active (sinkpad, TRUE);
+ return sinkpad;
+
+error:
+ SWFDEC_ERROR ("failed to create or link sinkpad");
+ gst_object_unref (srcpad);
+ gst_object_unref (sinkpad);
+ return NULL;
+}
+
+/*** DECODER ***/
+
+typedef struct {
+ GstElement * decoder;
+ GstPad * src;
+ GstPad * sink;
+ GQueue * queue; /* all the stored output GstBuffers */
+} SwfdecGstDecoder;
+
+static GstFlowReturn
+swfdec_gst_chain_func (GstPad *pad, GstBuffer *buffer)
+{
+ GQueue *queue = g_object_get_data (G_OBJECT (pad), "swfdec-queue");
+
+ g_queue_push_tail (queue, buffer);
+
+ return GST_FLOW_OK;
+}
+
+static gboolean
+swfdec_gst_decoder_init (SwfdecGstDecoder *dec, const char *name, GstCaps *srccaps, GstCaps *sinkcaps)
+{
+ if (name) {
+ dec->decoder = gst_element_factory_make (name, "decoder");
+ } else {
+ dec->decoder = swfdec_gst_get_element (srccaps);
}
- if (player->eof) {
- //doesn't work: g_object_set (fakesrc, "num-buffers", 1, NULL);
- /* HACK: just tell everyone we're done, that'll probably lose data in the
- * gst stream, since we can't properly push EOF, but that's life... */
- player->done = TRUE;
+ if (dec->decoder == NULL) {
+ SWFDEC_ERROR ("failed to create decoder");
+ return FALSE;
}
- if (player->in) {
- buf->data = g_memdup (player->in->data, player->in->length);
- buf->malloc_data = buf->data;
- buf->size = player->in->length;
+ dec->src = swfdec_gst_connect_srcpad (dec->decoder, srccaps);
+ if (dec->src == NULL)
+ return FALSE;
+ dec->sink = swfdec_gst_connect_sinkpad (dec->decoder, sinkcaps);
+ if (dec->sink == NULL)
+ return FALSE;
+ gst_pad_set_chain_function (dec->sink, swfdec_gst_chain_func);
+ dec->queue = g_queue_new ();
+ g_object_set_data (G_OBJECT (dec->sink), "swfdec-queue", dec->queue);
+ if (!gst_element_set_state (dec->decoder, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS) {
+ SWFDEC_ERROR ("could not change element state");
+ return FALSE;
}
- gst_buffer_set_caps (buf, player->srccaps);
- swfdec_buffer_unref (player->in);
- player->in = NULL;
- g_cond_signal (player->cond);
- g_mutex_unlock (player->mutex);
+ return TRUE;
}
static void
-swfdec_audio_decoder_gst_fakesink_handoff (GstElement *fakesrc, GstBuffer *buf,
- GstPad *pad, SwfdecGstAudio *player)
+swfdec_gst_decoder_finish (SwfdecGstDecoder *dec)
{
- SwfdecBuffer *buffer;
+ if (dec->decoder) {
+ gst_element_set_state (dec->decoder, GST_STATE_NULL);
+ g_object_unref (dec->decoder);
+ dec->decoder = NULL;
+ }
+ if (dec->src) {
+ g_object_unref (dec->src);
+ dec->src = NULL;
+ }
+ if (dec->sink) {
+ g_object_unref (dec->sink);
+ dec->sink = NULL;
+ }
+ if (dec->queue) {
+ GstBuffer *buffer;
+ while ((buffer = g_queue_pop_head (dec->queue)) != NULL) {
+ gst_buffer_unref (buffer);
+ }
+ g_queue_free (dec->queue);
+ dec->queue = NULL;
+ }
+}
- g_mutex_lock (player->mutex);
+static gboolean
+swfdec_gst_decoder_push (SwfdecGstDecoder *dec, GstBuffer *buffer)
+{
+ return GST_FLOW_IS_SUCCESS (gst_pad_push (dec->src, buffer));
+}
- while (player->pipeline == NULL && player->out != NULL)
- swfdec_cond_wait (player->cond, player->mutex);
- buffer = swfdec_buffer_new_for_data (
- g_memdup (buf->data, buf->size), buf->size);
- swfdec_buffer_queue_push (player->out, buffer);
- g_cond_signal (player->cond);
- g_mutex_unlock (player->mutex);
+static void
+swfdec_gst_decoder_push_eos (SwfdecGstDecoder *dec)
+{
+ gst_pad_push_event (dec->src, gst_event_new_eos ());
}
+static GstBuffer *
+swfdec_gst_decoder_pull (SwfdecGstDecoder *dec)
+{
+ return g_queue_pop_head (dec->queue);
+}
+
+/*** AUDIO ***/
+
+typedef struct _SwfdecGstAudio SwfdecGstAudio;
+struct _SwfdecGstAudio {
+ SwfdecAudioDecoder decoder;
+
+ gboolean error;
+ SwfdecGstDecoder dec;
+ SwfdecGstDecoder convert;
+ SwfdecGstDecoder resample;
+};
+
static void
-swfdec_audio_decoder_gst_link (GstElement *src, GstPad *pad, GstElement *sink)
+swfdec_audio_decoder_gst_free (SwfdecAudioDecoder *dec)
{
- if (!gst_element_link (src, sink)) {
- SWFDEC_ERROR ("no delayed link");
- }
+ SwfdecGstAudio *player = (SwfdecGstAudio *) dec;
+
+ swfdec_gst_decoder_finish (&player->dec);
+ swfdec_gst_decoder_finish (&player->convert);
+ swfdec_gst_decoder_finish (&player->resample);
+
+ g_slice_free (SwfdecGstAudio, player);
}
-static GstBusSyncReply
-swfdec_audio_decoder_gst_handle_bus (GstBus *bus, GstMessage *message, gpointer data)
+static void
+swfdec_audio_decoder_gst_push (SwfdecAudioDecoder *dec, SwfdecBuffer *buffer)
{
- SwfdecGstAudio *player = data;
-
- switch (message->type) {
- case GST_MESSAGE_EOS:
- case GST_MESSAGE_ERROR:
- g_mutex_lock (player->mutex);
- g_cond_signal (player->cond);
- player->done = TRUE;
- g_mutex_unlock (player->mutex);
- break;
- default:
- break;
+ SwfdecGstAudio *player = (SwfdecGstAudio *) dec;
+ GstBuffer *buf;
+
+ if (player->error)
+ return;
+ if (buffer == NULL) {
+ swfdec_gst_decoder_push_eos (&player->dec);
+ } else {
+ swfdec_buffer_ref (buffer);
+ buf = swfdec_gst_buffer_new (buffer);
+ if (!swfdec_gst_decoder_push (&player->dec, buf))
+ goto error;
}
- return GST_BUS_PASS;
+ while ((buf = swfdec_gst_decoder_pull (&player->dec))) {
+ if (!swfdec_gst_decoder_push (&player->convert, buf))
+ goto error;
+ }
+ while ((buf = swfdec_gst_decoder_pull (&player->convert))) {
+ if (!swfdec_gst_decoder_push (&player->resample, buf))
+ goto error;
+ }
+ return;
+
+error:
+ SWFDEC_ERROR ("error pushing");
+ player->error = TRUE;
+}
+
+static SwfdecBuffer *
+swfdec_audio_decoder_gst_pull (SwfdecAudioDecoder *dec)
+{
+ SwfdecGstAudio *player = (SwfdecGstAudio *) dec;
+ GstBuffer *buf;
+
+ if (player->error)
+ return NULL;
+ buf = swfdec_gst_decoder_pull (&player->resample);
+ if (buf == NULL)
+ return NULL;
+ return swfdec_buffer_new_from_gst (buf);
}
SwfdecAudioDecoder *
swfdec_audio_decoder_gst_new (SwfdecAudioCodec type, SwfdecAudioFormat format)
{
SwfdecGstAudio *player;
- GstElement *fakesrc, *fakesink, *decoder, *convert;
- GstBus *bus;
- GstCaps *caps;
+ GstCaps *srccaps, *sinkcaps;
if (!gst_init_check (NULL, NULL, NULL))
return NULL;
switch (type) {
case SWFDEC_AUDIO_CODEC_MP3:
- caps = gst_caps_from_string ("audio/mpeg, mpegversion=(int)1, layer=(int)3");
+ srccaps = gst_caps_from_string ("audio/mpeg, mpegversion=(int)1, layer=(int)3");
break;
default:
return NULL;
}
- g_assert (caps);
+ g_assert (srccaps);
player = g_slice_new0 (SwfdecGstAudio);
player->decoder.format = swfdec_audio_format_new (44100, 2, TRUE);
player->decoder.pull = swfdec_audio_decoder_gst_pull;
player->decoder.push = swfdec_audio_decoder_gst_push;
player->decoder.free = swfdec_audio_decoder_gst_free;
- player->pipeline = gst_pipeline_new ("pipeline");
- player->refcount = 1;
- g_assert (player->pipeline);
- bus = gst_element_get_bus (player->pipeline);
- g_atomic_int_inc (&player->refcount);
- g_object_weak_ref (G_OBJECT (bus), swfdec_gst_audio_unref, player);
- gst_bus_set_sync_handler (bus, swfdec_audio_decoder_gst_handle_bus, player);
- player->mutex = g_mutex_new ();
- player->cond = g_cond_new ();
- player->out = swfdec_buffer_queue_new ();
- player->srccaps = caps;
- fakesrc = gst_element_factory_make ("fakesrc", NULL);
- if (fakesrc == NULL) {
- SWFDEC_ERROR ("failed to create fakesrc");
- swfdec_audio_decoder_gst_free (&player->decoder);
- return NULL;
- }
- g_object_set (fakesrc, "signal-handoffs", TRUE,
- "can-activate-pull", FALSE, NULL);
- g_signal_connect (fakesrc, "handoff",
- G_CALLBACK (swfdec_audio_decoder_gst_fakesrc_handoff), player);
- g_atomic_int_inc (&player->refcount);
- g_object_weak_ref (G_OBJECT (fakesrc), swfdec_gst_audio_unref, player);
- gst_bin_add (GST_BIN (player->pipeline), fakesrc);
- fakesink = gst_element_factory_make ("fakesink", NULL);
- if (fakesink == NULL) {
- SWFDEC_ERROR ("failed to create fakesink");
- swfdec_audio_decoder_gst_free (&player->decoder);
- return NULL;
- }
- g_object_set (fakesink, "signal-handoffs", TRUE, NULL);
- g_signal_connect (fakesink, "handoff",
- G_CALLBACK (swfdec_audio_decoder_gst_fakesink_handoff), player);
- g_atomic_int_inc (&player->refcount);
- g_object_weak_ref (G_OBJECT (fakesink), swfdec_gst_audio_unref, player);
- gst_bin_add (GST_BIN (player->pipeline), fakesink);
- decoder = gst_element_factory_make ("decodebin", NULL);
- if (decoder == NULL) {
- SWFDEC_ERROR ("failed to create decoder");
- swfdec_audio_decoder_gst_free (&player->decoder);
- return NULL;
- }
- gst_bin_add (GST_BIN (player->pipeline), decoder);
- convert = gst_element_factory_make ("audioconvert", NULL);
- if (convert == NULL) {
- SWFDEC_ERROR ("failed to create audioconvert");
- swfdec_audio_decoder_gst_free (&player->decoder);
- return NULL;
- }
- gst_bin_add (GST_BIN (player->pipeline), convert);
- g_signal_connect (decoder, "pad-added",
- G_CALLBACK (swfdec_audio_decoder_gst_link), convert);
-
- caps = gst_caps_from_string ("audio/x-raw-int, endianness=byte_order, signed=(boolean)true, width=16, depth=16, rate=44100, channels=2");
- g_assert (caps);
- if (!gst_element_link_filtered (fakesrc, decoder, player->srccaps) ||
- !gst_element_link_filtered (convert, fakesink, caps)) {
- SWFDEC_ERROR ("linking failed");
- swfdec_audio_decoder_gst_free (&player->decoder);
- return NULL;
- }
- gst_caps_unref (caps);
- if (gst_element_set_state (player->pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
- SWFDEC_ERROR ("failed to change sate");
- swfdec_audio_decoder_gst_free (&player->decoder);
- return NULL;
- }
+ /* create decoder */
+ sinkcaps = gst_caps_from_string ("audio/x-raw-int");
+ g_assert (sinkcaps);
+ if (!swfdec_gst_decoder_init (&player->dec, NULL, srccaps, sinkcaps))
+ goto error;
+ /* create audioconvert */
+ gst_caps_unref (srccaps);
+ srccaps = sinkcaps;
+ sinkcaps = gst_caps_from_string ("audio/x-raw-int, endianness=byte_order, signed=(boolean)true, width=16, depth=16, channels=2");
+ g_assert (sinkcaps);
+ if (!swfdec_gst_decoder_init (&player->convert, "audioconvert", srccaps, sinkcaps))
+ goto error;
+ /* create audiorate */
+ gst_caps_unref (srccaps);
+ srccaps = sinkcaps;
+ sinkcaps = gst_caps_from_string ("audio/x-raw-int, endianness=byte_order, signed=(boolean)true, width=16, depth=16, rate=44100, channels=2");
+ g_assert (sinkcaps);
+ if (!swfdec_gst_decoder_init (&player->resample, "audioresample", srccaps, sinkcaps))
+ goto error;
+
+ gst_caps_unref (srccaps);
+ gst_caps_unref (sinkcaps);
return &player->decoder;
+
+error:
+ swfdec_audio_decoder_gst_free (&player->decoder);
+ gst_caps_unref (srccaps);
+ gst_caps_unref (sinkcaps);
+ return NULL;
}
/*** VIDEO ***/
+#if 0
+#define swfdec_cond_wait(cond, mutex) G_STMT_START { \
+ g_print ("waiting at %s\n", G_STRLOC); \
+ g_cond_wait (cond, mutex); \
+ g_print (" done at %s\n", G_STRLOC); \
+}G_STMT_END
+#else
+#define swfdec_cond_wait g_cond_wait
+#endif
+
typedef struct _SwfdecGstVideo SwfdecGstVideo;
struct _SwfdecGstVideo {
SwfdecVideoDecoder decoder;
commit e34ac83cf6b15eccaf945203f0f0628d04649870
Author: Benjamin Otte <otte at gnome.org>
Date: Fri Oct 26 21:11:15 2007 +0200
use , as delimiter for SWFDEC_CODEC, not :
diff --git a/libswfdec/swfdec_codec_audio.c b/libswfdec/swfdec_codec_audio.c
index 7c4ab8a..a46b315 100644
--- a/libswfdec/swfdec_codec_audio.c
+++ b/libswfdec/swfdec_codec_audio.c
@@ -167,7 +167,7 @@ swfdec_audio_decoder_new (SwfdecAudioCodec codec, SwfdecAudioFormat format)
break;
}
} else {
- char **split = g_strsplit (list, ":", -1);
+ char **split = g_strsplit (list, ",", -1);
guint i, j;
ret = NULL;
SWFDEC_LOG ("codecs limited to \"%s\"", list);
diff --git a/libswfdec/swfdec_codec_video.c b/libswfdec/swfdec_codec_video.c
index 291c24d..93bc6dd 100644
--- a/libswfdec/swfdec_codec_video.c
+++ b/libswfdec/swfdec_codec_video.c
@@ -80,7 +80,7 @@ swfdec_video_decoder_new (SwfdecVideoCodec codec)
break;
}
} else {
- char **split = g_strsplit (list, ":", -1);
+ char **split = g_strsplit (list, ",", -1);
guint i, j;
ret = NULL;
SWFDEC_LOG ("codecs limited to \"%s\"", list);
commit 9552b2f74398a8e75ba78153b779a27455a47557
Author: Benjamin Otte <otte at gnome.org>
Date: Fri Oct 26 08:52:21 2007 +0200
don't scare them with GTK either
diff --git a/configure.ac b/configure.ac
index 2dc0eed..296b09b 100644
--- a/configure.ac
+++ b/configure.ac
@@ -119,7 +119,7 @@ if test "$enable_gtk" = "yes"; then
AC_MSG_ERROR([Couldn't find a suitable Gtk version. You need at least version $GTK_VER])
fi
else
- AC_MSG_WARN([*** Gtk support was not enabled. ***])
+ AC_MSG_NOTICE([Gtk support was not enabled.])
fi
AM_CONDITIONAL(WITH_GTK, [test "x$HAVE_GTK" = xyes])
commit fb8cd3ca0a393dca4b9f5f278d16fdbeaec954eb
Author: Benjamin Otte <otte at gnome.org>
Date: Fri Oct 26 08:48:55 2007 +0200
Don't scare people about disabled dependencies
diff --git a/configure.ac b/configure.ac
index 97d474b..2dc0eed 100644
--- a/configure.ac
+++ b/configure.ac
@@ -201,7 +201,7 @@ if test "$enable_mad" = "yes"; then
AC_MSG_ERROR([Couldn't find mad. You might need to install the libmad0-dev package.])
fi
else
- AC_MSG_WARN([*** mad audio support was not enabled. ***])
+ AC_MSG_NOTICE([mad audio support was not enabled.])
fi
AM_CONDITIONAL(HAVE_MAD, [test "x$HAVE_MAD" = xyes])
@@ -225,7 +225,7 @@ if test "$enable_ffmpeg" = "yes"; then
AC_MSG_ERROR([Couldn't find ffmpeg. You might need to install the libavcodec-dev and libswscale-dev packages.])
fi
else
- AC_MSG_WARN([*** ffmpeg support was not enabled. ***])
+ AC_MSG_NOTICE([ffmpeg support was not enabled.])
fi
AM_CONDITIONAL(HAVE_FFMPEG, [test "x$HAVE_FFMPEG" = xyes])
AC_ARG_ENABLE(gstreamer,
@@ -243,7 +243,7 @@ if test "$enable_gstreamer" = "yes"; then
AC_MSG_ERROR([Couldn't find GStreamer $GST_REQUIRED.])
fi
else
- AC_MSG_WARN([*** GStreamer support was not enabled. ***])
+ AC_MSG_NOTICE([GStreamer support was not enabled.])
fi
AM_CONDITIONAL(HAVE_GST, [test "x$HAVE_GST" = xyes])
@@ -262,7 +262,7 @@ if test "$enable_libsoup" = "yes"; then
AC_MSG_ERROR([Couldn't find libsoup-2.2.])
fi
else
- AC_MSG_WARN([*** libsoup HTTP support was not enabled. ***])
+ AC_MSG_NOTICE([libsoup HTTP support was not enabled.])
fi
AM_CONDITIONAL(HAVE_HTTP, [test "x$HAVE_HTTP" = xyes])
@@ -281,7 +281,7 @@ if test "$enable_vivi" = "yes"; then
AC_MSG_ERROR([Vivified requirements not met. You need libming >= $MING_REQUIRED and Gtk+ >= $VIVI_GTK_REQUIRED.])
fi
else
- AC_MSG_WARN([*** Vivified was not enabled. ***])
+ AC_MSG_NOTICE([Vivified was not enabled.])
fi
AM_CONDITIONAL(HAVE_VIVI, [test "x$HAVE_VIVI" = xyes])
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