static gboolean
event_func (GstPad * pad, GstObject * noparent, GstEvent * event)
{
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
GstCaps **caps2 = g_object_get_data (G_OBJECT (pad), "caps");
GstCaps *caps_no_ssrc;
GstCaps *caps2_no_ssrc;
gst_event_parse_caps (event, &caps);
caps_no_ssrc = remove_ssrc_from_caps (caps);
caps2_no_ssrc = remove_ssrc_from_caps (*caps2);
fail_unless (caps2 != NULL && *caps2 != NULL);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_caps_is_fixed (*caps2));
fail_unless (gst_caps_is_equal_fixed (caps_no_ssrc, caps2_no_ssrc));
gst_caps_unref (caps_no_ssrc);
gst_caps_unref (caps2_no_ssrc);
break;
}
default:
break;
}
gst_event_unref (event);
return TRUE;
}
static gboolean
_rtpbin_pad_have_data_callback (GstPad *pad, GstMiniObject *miniobj,
gpointer user_data)
{
FsRtpSubStream *self = FS_RTP_SUB_STREAM (user_data);
gboolean ret = TRUE;
gboolean remove = FALSE;
FsRtpSession *session;
if (fs_rtp_session_has_disposed_enter (self->priv->session, NULL))
return FALSE;
if (fs_rtp_sub_stream_has_stopped_enter (self))
{
fs_rtp_session_has_disposed_exit (self->priv->session);
return FALSE;
}
g_object_ref (self);
session = g_object_ref (self->priv->session);
FS_RTP_SESSION_LOCK (self->priv->session);
if (!self->priv->codecbin || !self->codec || !self->priv->caps)
{
ret = FALSE;
}
else if (GST_IS_BUFFER (miniobj))
{
if (!gst_caps_is_equal_fixed (GST_BUFFER_CAPS (miniobj), self->priv->caps))
{
if (!gst_caps_can_intersect (GST_BUFFER_CAPS (miniobj),
self->priv->caps))
ret = FALSE;
}
else
{
remove = TRUE;
}
}
if (remove && self->priv->blocking_id)
{
gst_pad_remove_data_probe (pad, self->priv->blocking_id);
self->priv->blocking_id = 0;
}
FS_RTP_SESSION_UNLOCK (self->priv->session);
fs_rtp_sub_stream_has_stopped_exit (self);
fs_rtp_session_has_disposed_exit (self->priv->session);
g_object_unref (self);
g_object_unref (session);
return ret;
}
static gboolean
gst_vdp_output_src_pad_acceptcaps (GstPad * pad, GstCaps * caps)
{
GstVdpOutputSrcPad *vdp_pad = GST_VDP_OUTPUT_SRC_PAD (pad);
if (!vdp_pad->lock_caps)
return TRUE;
return gst_caps_is_equal_fixed (caps, GST_PAD_CAPS (pad));
}
gboolean
gst_validate_media_info_compare (GstValidateMediaInfo * expected,
GstValidateMediaInfo * extracted)
{
gboolean ret = TRUE;
if (expected->duration != extracted->duration) {
g_print ("Duration changed: %" GST_TIME_FORMAT " -> %" GST_TIME_FORMAT "\n",
GST_TIME_ARGS (expected->duration),
GST_TIME_ARGS (extracted->duration));
ret = FALSE;
}
if (expected->file_size != extracted->file_size) {
g_print ("File size changed: %" G_GUINT64_FORMAT " -> %" G_GUINT64_FORMAT
"\n", expected->file_size, extracted->file_size);
ret = FALSE;
}
if (expected->seekable && !extracted->seekable) {
g_print ("File isn't seekable anymore\n");
ret = FALSE;
}
if (extracted->discover_only == FALSE) {
if (expected->playback_error == NULL && extracted->playback_error) {
g_print ("Playback is now failing with: %s\n", extracted->playback_error);
ret = FALSE;
}
if (expected->reverse_playback_error == NULL
&& extracted->reverse_playback_error) {
g_print ("Reverse playback is now failing with: %s\n",
extracted->reverse_playback_error);
ret = FALSE;
}
if (expected->track_switch_error == NULL && extracted->track_switch_error) {
g_print ("Track switching is now failing with: %s\n",
extracted->track_switch_error);
ret = FALSE;
}
}
if (extracted->stream_info == NULL || expected->stream_info == NULL) {
g_print ("Stream infos could not be retrieved, an error occured\n");
ret = FALSE;
} else if (expected->stream_info
&& !gst_caps_is_equal_fixed (expected->stream_info->caps,
extracted->stream_info->caps)) {
gchar *caps1 = gst_caps_to_string (expected->stream_info->caps);
gchar *caps2 = gst_caps_to_string (extracted->stream_info->caps);
g_print ("Media caps changed: '%s' -> '%s'\n", caps1, caps2);
g_free (caps1);
g_free (caps2);
ret = FALSE;
}
return ret;
}
gboolean
gtk_gst_widget_set_caps (GtkGstWidget * widget, GstCaps * caps)
{
GMainContext *main_context = g_main_context_default ();
GstVideoInfo v_info;
g_return_val_if_fail (GTK_IS_GST_WIDGET (widget), FALSE);
g_return_val_if_fail (GST_IS_CAPS (caps), FALSE);
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
if (widget->priv->caps && gst_caps_is_equal_fixed (widget->priv->caps, caps))
return TRUE;
if (!gst_video_info_from_caps (&v_info, caps))
return FALSE;
/* FIXME: support other formats */
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
g_return_val_if_fail (GST_VIDEO_INFO_FORMAT (&v_info) ==
GST_VIDEO_FORMAT_BGRA || GST_VIDEO_INFO_FORMAT (&v_info) ==
GST_VIDEO_FORMAT_BGRx, FALSE);
#else
g_return_val_if_fail (GST_VIDEO_INFO_FORMAT (&v_info) ==
GST_VIDEO_FORMAT_ARGB || GST_VIDEO_INFO_FORMAT (&v_info) ==
GST_VIDEO_FORMAT_xRGB, FALSE);
#endif
g_mutex_lock (&widget->priv->lock);
if (!_calculate_par (widget, &v_info)) {
g_mutex_unlock (&widget->priv->lock);
return FALSE;
}
gst_caps_replace (&widget->priv->caps, caps);
widget->priv->v_info = v_info;
widget->priv->negotiated = TRUE;
g_mutex_unlock (&widget->priv->lock);
gtk_widget_queue_resize (GTK_WIDGET (widget));
g_main_context_invoke (main_context, (GSourceFunc) _queue_resize, widget);
return TRUE;
}
static gboolean
_rtpbin_pad_have_data_callback (GstPad *pad, GstMiniObject *miniobj,
gpointer user_data)
{
FsRtpSubStream *self = FS_RTP_SUB_STREAM (user_data);
gboolean ret = TRUE;
gboolean remove = FALSE;
FS_RTP_SESSION_LOCK (self->priv->session);
if (!self->priv->codecbin || !self->codec || !self->priv->caps)
{
ret = FALSE;
}
else if (GST_IS_BUFFER (miniobj))
{
if (!gst_caps_is_equal_fixed (GST_BUFFER_CAPS (miniobj), self->priv->caps))
{
GstCaps *intersect = gst_caps_intersect (GST_BUFFER_CAPS (miniobj),
self->priv->caps);
if (gst_caps_is_empty (intersect))
ret = FALSE;
else
gst_buffer_set_caps (GST_BUFFER (miniobj), self->priv->caps);
gst_caps_unref (intersect);
}
else
{
remove = TRUE;
}
}
if (remove && self->priv->blocking_id)
{
gst_pad_remove_data_probe (pad, self->priv->blocking_id);
self->priv->blocking_id = 0;
}
FS_RTP_SESSION_UNLOCK (self->priv->session);
return ret;
}
static void
run_decoding_test (GstElement * mpg123audiodec, gchar const *filename)
{
GstBus *bus;
unsigned int num_input_buffers, num_decoded_buffers;
gint expected_size;
GstCaps *out_caps, *caps;
GstAudioInfo audioinfo;
GstElement *input_pipeline, *input_appsink;
int i;
GstBuffer *outbuffer;
/* 440 Hz = frequency of sine wave in audio data
* 44100 Hz = sample rate
* (44100 / 2) Hz = Nyquist frequency */
static double const expected_frequency_spot = 440.0 / (44100.0 / 2.0);
fail_unless (gst_element_set_state (mpg123audiodec,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
bus = gst_bus_new ();
gst_element_set_bus (mpg123audiodec, bus);
setup_input_pipeline (filename, &input_pipeline, &input_appsink);
num_input_buffers = 0;
while (TRUE) {
GstSample *sample;
GstBuffer *input_buffer;
sample = gst_app_sink_pull_sample (GST_APP_SINK (input_appsink));
if (sample == NULL)
break;
fail_unless (GST_IS_SAMPLE (sample));
input_buffer = gst_sample_get_buffer (sample);
fail_if (input_buffer == NULL);
/* This is done to be on the safe side - docs say lifetime of the input buffer
* depends *solely* on the sample */
input_buffer = gst_buffer_copy (input_buffer);
fail_unless_equals_int (gst_pad_push (mysrcpad, input_buffer), GST_FLOW_OK);
++num_input_buffers;
gst_sample_unref (sample);
}
num_decoded_buffers = g_list_length (buffers);
/* check number of decoded buffers */
fail_unless_equals_int (num_decoded_buffers, num_input_buffers - 2);
caps = gst_pad_get_current_caps (mysinkpad);
GST_LOG ("output caps %" GST_PTR_FORMAT, caps);
fail_unless (gst_audio_info_from_caps (&audioinfo, caps),
"Getting audio info from caps failed");
/* check caps */
out_caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, "S32LE",
"layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1, NULL);
fail_unless (gst_caps_is_equal_fixed (caps, out_caps), "Incorrect out caps");
gst_caps_unref (out_caps);
gst_caps_unref (caps);
/* here, test if decoded data is a sine tone, and if the sine frequency is at the
* right spot in the spectrum */
for (i = 0; i < num_decoded_buffers; ++i) {
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL, "Invalid buffer retrieved");
/* MPEG 1 layer 2 uses 1152 samples per frame */
expected_size = 1152 * GST_AUDIO_INFO_BPF (&audioinfo);
fail_unless_equals_int (gst_buffer_get_size (outbuffer), expected_size);
check_main_frequency_spot_S32 (outbuffer, expected_frequency_spot);
buffers = g_list_remove (buffers, outbuffer);
gst_buffer_unref (outbuffer);
outbuffer = NULL;
}
g_list_free (buffers);
buffers = NULL;
cleanup_input_pipeline (input_pipeline);
gst_bus_set_flushing (bus, TRUE);
gst_element_set_bus (mpg123audiodec, NULL);
gst_object_unref (GST_OBJECT (bus));
}
static gboolean
gst_droidadec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
GstDroidADec *dec = GST_DROIDADEC (decoder);
GstStructure *str = gst_caps_get_structure (caps, 0);
const GValue *value = gst_structure_get_value (str, "codec_data");
GstBuffer *codec_data = value ? gst_value_get_buffer (value) : NULL;
/*
* destroying the droidmedia codec here will cause stagefright to call abort.
* That is why we create it after we are sure that everything is correct
*/
GST_DEBUG_OBJECT (dec, "set format %" GST_PTR_FORMAT, caps);
if (dec->codec) {
/* If we get a format change then we stop */
GstCaps *current =
gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (decoder));
gboolean equal = gst_caps_is_equal_fixed (caps, current);
gst_caps_unref (current);
GST_DEBUG_OBJECT (dec, "new format is similar to old format? %d", equal);
if (!equal) {
GST_ELEMENT_ERROR (dec, LIBRARY, SETTINGS, (NULL),
("codec already configured"));
}
return equal;
}
dec->codec_type =
gst_droid_codec_new_from_caps (caps, GST_DROID_CODEC_DECODER_AUDIO);
if (!dec->codec_type) {
GST_ELEMENT_ERROR (dec, LIBRARY, FAILED, (NULL),
("Unknown codec type for caps %" GST_PTR_FORMAT, caps));
return FALSE;
}
if (!gst_structure_get_int (str, "channels", &dec->channels) ||
!gst_structure_get_int (str, "rate", &dec->rate)) {
GST_ELEMENT_ERROR (dec, STREAM, FORMAT, (NULL),
("Failed to parse caps %" GST_PTR_FORMAT, caps));
return FALSE;
}
GST_INFO_OBJECT (dec, "configuring decoder. rate=%d, channels=%d", dec->rate,
dec->channels);
gst_buffer_replace (&dec->codec_data, codec_data);
/* handle_frame will create the codec */
dec->dirty = TRUE;
dec->spf = gst_droid_codec_get_samples_per_frane (caps);
GST_INFO_OBJECT (dec, "samples per frame: %d", dec->spf);
return TRUE;
}
static void
gst_hls_demux_stream_loop (GstHLSDemux * demux)
{
GstFragment *fragment;
GstBuffer *buf;
GstFlowReturn ret;
GstCaps *bufcaps, *srccaps = NULL;
/* Loop for the source pad task. The task is started when we have
* received the main playlist from the source element. It tries first to
* cache the first fragments and then it waits until it has more data in the
* queue. This task is woken up when we push a new fragment to the queue or
* when we reached the end of the playlist */
if (G_UNLIKELY (demux->need_cache)) {
if (!gst_hls_demux_cache_fragments (demux))
goto cache_error;
/* we can start now the updates thread (only if on playing) */
if (GST_STATE (demux) == GST_STATE_PLAYING)
gst_task_start (demux->updates_task);
GST_INFO_OBJECT (demux, "First fragments cached successfully");
}
if (g_queue_is_empty (demux->queue)) {
if (demux->end_of_playlist)
goto end_of_playlist;
goto pause_task;
}
fragment = g_queue_pop_head (demux->queue);
buf = gst_fragment_get_buffer (fragment);
/* Figure out if we need to create/switch pads */
if (G_LIKELY (demux->srcpad))
srccaps = gst_pad_get_current_caps (demux->srcpad);
bufcaps = gst_fragment_get_caps (fragment);
if (G_UNLIKELY (!srccaps || !gst_caps_is_equal_fixed (bufcaps, srccaps)
|| demux->need_segment)) {
switch_pads (demux, bufcaps);
demux->need_segment = TRUE;
}
gst_caps_unref (bufcaps);
if (G_LIKELY (srccaps))
gst_caps_unref (srccaps);
g_object_unref (fragment);
if (demux->need_segment) {
GstSegment segment;
GstClockTime start = GST_BUFFER_PTS (buf);
start += demux->position_shift;
/* And send a newsegment */
GST_DEBUG_OBJECT (demux, "Sending new-segment. segment start:%"
GST_TIME_FORMAT, GST_TIME_ARGS (start));
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.start = start;
segment.time = start;
gst_pad_push_event (demux->srcpad, gst_event_new_segment (&segment));
demux->need_segment = FALSE;
demux->position_shift = 0;
}
ret = gst_pad_push (demux->srcpad, buf);
if (ret != GST_FLOW_OK)
goto error_pushing;
return;
end_of_playlist:
{
GST_DEBUG_OBJECT (demux, "Reached end of playlist, sending EOS");
gst_pad_push_event (demux->srcpad, gst_event_new_eos ());
gst_hls_demux_stop (demux);
return;
}
cache_error:
{
gst_task_pause (demux->stream_task);
if (!demux->cancelled) {
GST_ELEMENT_ERROR (demux, RESOURCE, NOT_FOUND,
("Could not cache the first fragments"), (NULL));
gst_hls_demux_stop (demux);
}
return;
}
error_pushing:
{
/* FIXME: handle error */
GST_DEBUG_OBJECT (demux, "Error pushing buffer: %s... stopping task",
gst_flow_get_name (ret));
gst_hls_demux_stop (demux);
return;
}
pause_task:
{
gst_task_pause (demux->stream_task);
return;
}
}
/*
* Method: equal_fixed?(caps)
* caps: another Gst::Caps.
*
* Returns: whether the given fixed caps represent the same set of caps
than self
* (self must be fixed as well).
*/
static VALUE
rg_equal_fixed_p (VALUE self, VALUE caps)
{
return CBOOL2RVAL (gst_caps_is_equal_fixed (RGST_CAPS (self), RGST_CAPS
(caps)));
}