static void
gst_navseek_change_playback_rate (GstNavSeek * navseek, gdouble rate)
{
gboolean ret;
GstPad *peer_pad;
gint64 current_position;
peer_pad = gst_pad_get_peer (GST_BASE_TRANSFORM (navseek)->sinkpad);
ret = gst_pad_query_position (peer_pad, GST_FORMAT_TIME, ¤t_position);
if (ret) {
GstEvent *event;
gint64 start;
gint64 stop;
if (rate > 0.0) {
start = current_position;
stop = -1;
} else {
/* negative rate: we play from stop to start */
start = 0;
stop = current_position;
}
event = gst_event_new_seek (rate, GST_FORMAT_TIME,
GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_SKIP,
GST_SEEK_TYPE_SET, start, GST_SEEK_TYPE_SET, stop);
gst_pad_send_event (peer_pad, event);
}
gst_object_unref (peer_pad);
}
static void
gst_navseek_seek (GstNavSeek * navseek, gint64 offset)
{
gboolean ret;
GstPad *peer_pad;
gint64 peer_value;
/* Query for the current time then attempt to set to time + offset */
peer_pad = gst_pad_get_peer (GST_BASE_TRANSFORM (navseek)->sinkpad);
ret = gst_pad_query_position (peer_pad, GST_FORMAT_TIME, &peer_value);
if (ret) {
GstEvent *event;
peer_value += offset;
if (peer_value < 0)
peer_value = 0;
event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_FLUSH,
GST_SEEK_TYPE_SET, peer_value, GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE);
gst_pad_send_event (peer_pad, event);
}
gst_object_unref (peer_pad);
}
static gboolean
gst_vorbis_enc_src_query (GstPad * pad, GstQuery * query)
{
gboolean res = TRUE;
GstVorbisEnc *vorbisenc;
GstPad *peerpad;
vorbisenc = GST_VORBISENC (gst_pad_get_parent (pad));
peerpad = gst_pad_get_peer (GST_PAD (vorbisenc->sinkpad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
GstFormat fmt, req_fmt;
gint64 pos, val;
gst_query_parse_position (query, &req_fmt, NULL);
if ((res = gst_pad_query_position (peerpad, &req_fmt, &val))) {
gst_query_set_position (query, req_fmt, val);
break;
}
fmt = GST_FORMAT_TIME;
if (!(res = gst_pad_query_position (peerpad, &fmt, &pos)))
break;
if ((res = gst_pad_query_convert (peerpad, fmt, pos, &req_fmt, &val))) {
gst_query_set_position (query, req_fmt, val);
}
break;
}
case GST_QUERY_DURATION:
{
GstFormat fmt, req_fmt;
gint64 dur, val;
gst_query_parse_duration (query, &req_fmt, NULL);
if ((res = gst_pad_query_duration (peerpad, &req_fmt, &val))) {
gst_query_set_duration (query, req_fmt, val);
break;
}
fmt = GST_FORMAT_TIME;
if (!(res = gst_pad_query_duration (peerpad, &fmt, &dur)))
break;
if ((res = gst_pad_query_convert (peerpad, fmt, dur, &req_fmt, &val))) {
gst_query_set_duration (query, req_fmt, val);
}
break;
}
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
if (!(res =
gst_vorbis_enc_convert_src (pad, src_fmt, src_val, &dest_fmt,
&dest_val)))
goto error;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
error:
gst_object_unref (peerpad);
gst_object_unref (vorbisenc);
return res;
}
static gboolean
xing_mp3_encoder_src_query(GstPad * pad, GstQuery * query)
{
gboolean res = TRUE;
XingMp3Encoder *encoder;
GstPad *peerpad;
encoder = XING_MP3_ENCODER(gst_pad_get_parent(pad));
peerpad = gst_pad_get_peer(GST_PAD(encoder->sinkpad));
switch(GST_QUERY_TYPE(query)) {
case GST_QUERY_DURATION: {
GstFormat fmt, req_fmt;
gint64 dur, val;
gst_query_parse_duration(query, &req_fmt, NULL);
if((res = gst_pad_query_duration(peerpad, &req_fmt, &val))) {
gst_query_set_duration(query, req_fmt, val);
break;
}
fmt = GST_FORMAT_TIME;
if(!(res = gst_pad_query_duration (peerpad, &fmt, &dur))) {
break;
}
if((res = gst_pad_query_convert(peerpad, fmt, dur, &req_fmt, &val))) {
gst_query_set_duration(query, req_fmt, val);
}
break;
}
case GST_QUERY_POSITION: {
GstFormat fmt, req_fmt;
gint64 pos, val;
gst_query_parse_position(query, &req_fmt, NULL);
if((res = gst_pad_query_position(peerpad, &req_fmt, &val))) {
gst_query_set_position (query, req_fmt, val);
break;
}
fmt = GST_FORMAT_TIME;
if(!(res = gst_pad_query_position(peerpad, &fmt, &pos))) {
break;
}
if((res = gst_pad_query_convert(peerpad, fmt, pos, &req_fmt, &val))) {
gst_query_set_position(query, req_fmt, val);
}
break;
}
case GST_QUERY_CONVERT: {
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert(query, &src_fmt, &src_val, &dest_fmt, &dest_val);
if(!(res = xing_mp3_encoder_convert_src(pad, src_fmt, src_val, &dest_fmt, &dest_val))) {
gst_object_unref(peerpad);
gst_object_unref(encoder);
}
gst_query_set_convert(query, src_fmt, src_val, dest_fmt, dest_val);
break;
}
default:
res = gst_pad_query_default(pad, query);
break;
}
return res;
}
// ----------------------------------------------------------------------------
// Handle the "pad-added" message
void
GStreamerImportFileHandle::OnPadAdded(GstPad *pad)
{
// Retrieve the stream caps...skip stream if unavailable
GstCaps *caps = gst_pad_get_current_caps(pad);
if (!caps)
{
WARN(mPipeline, ("OnPadAdded: unable to retrieve stream caps"));
return;
}
// Get the caps structure...no need to release
GstStructure *str = gst_caps_get_structure(caps, 0);
if (!str)
{
WARN(mPipeline, ("OnPadAdded: unable to retrieve caps structure"));
gst_caps_unref(caps);
return;
}
// Only accept audio streams...no need to release
const gchar *name = gst_structure_get_name(str);
if (!g_strrstr(name, "audio"))
{
WARN(mPipeline, ("OnPadAdded: bypassing '%s' stream", name));
gst_caps_unref(caps);
return;
}
// Allocate a new stream context
GStreamContext *c = g_new0(GStreamContext, 1);
if (!c)
{
WARN(mPipeline, ("OnPadAdded: unable to allocate stream context"));
gst_caps_unref(caps);
return;
}
// Set initial state
c->mUse = true;
// Always add it to the context list to keep the number of contexts
// in sync with the number of streams
g_mutex_lock(&mStreamsLock);
g_ptr_array_add(mStreams, c);
g_mutex_unlock(&mStreamsLock);
// Need pointer to context during pad removal (pad-remove signal)
SETCTX(pad, c);
// Save the stream's start time and duration
gst_pad_query_position(pad, GST_FORMAT_TIME, &c->mPosition);
gst_pad_query_duration(pad, GST_FORMAT_TIME, &c->mDuration);
// Retrieve the number of channels and validate
gint channels = -1;
gst_structure_get_int(str, "channels", &channels);
if (channels <= 0)
{
WARN(mPipeline, ("OnPadAdded: channel count is invalid %d", channels));
gst_caps_unref(caps);
return;
}
c->mNumChannels = channels;
// Retrieve the sample rate and validate
gint rate = -1;
gst_structure_get_int(str, "rate", &rate);
if (rate <= 0)
{
WARN(mPipeline, ("OnPadAdded: sample rate is invalid %d", rate));
gst_caps_unref(caps);
return;
}
c->mSampleRate = (double) rate;
c->mType = g_strdup(name);
if (c->mType == NULL)
{
WARN(mPipeline, ("OnPadAdded: unable to allocate audio type"));
gst_caps_unref(caps);
return;
}
// Done with capabilities
gst_caps_unref(caps);
// Create audioconvert element
c->mConv = gst_element_factory_make("audioconvert", NULL);
if (!c->mConv)
{
WARN(mPipeline, ("OnPadAdded: failed to create audioconvert element"));
return;
}
// Create appsink element
c->mSink = gst_element_factory_make("appsink", NULL);
if (!c->mSink)
{
WARN(mPipeline, ("OnPadAdded: failed to create appsink element"));
return;
}
SETCTX(c->mSink, c);
// Set the appsink callbacks and add the context pointer
gst_app_sink_set_callbacks(GST_APP_SINK(c->mSink), &AppSinkCallbacks, this, NULL);
// Set the capabilities that we desire
caps = gst_static_caps_get(&supportedCaps);
if (!caps)
{
WARN(mPipeline, ("OnPadAdded: failed to create static caps"));
return;
}
gst_app_sink_set_caps(GST_APP_SINK(c->mSink), caps);
gst_caps_unref(caps);
// Do not sync to the clock...process as quickly as possible
gst_base_sink_set_sync(GST_BASE_SINK(c->mSink), FALSE);
// Don't drop buffers...allow queue to build unfettered
gst_app_sink_set_drop(GST_APP_SINK(c->mSink), FALSE);
// Add both elements to the pipeline
gst_bin_add_many(GST_BIN(mPipeline), c->mConv, c->mSink, NULL);
// Link them together
if (!gst_element_link(c->mConv, c->mSink))
{
WARN(mPipeline, ("OnPadAdded: failed to link autioconvert and appsink"));
return;
}
// Link the audiconvert sink pad to the src pad
GstPadLinkReturn ret = GST_PAD_LINK_OK;
GstPad *convsink = gst_element_get_static_pad(c->mConv, "sink");
if (convsink)
{
ret = gst_pad_link(pad, convsink);
gst_object_unref(convsink);
}
if (!convsink || ret != GST_PAD_LINK_OK)
{
WARN(mPipeline, ("OnPadAdded: failed to link uridecodebin to audioconvert - %d", ret));
return;
}
// Synchronize audioconvert state with parent
if (!gst_element_sync_state_with_parent(c->mConv))
{
WARN(mPipeline, ("OnPadAdded: unable to sync audioconvert state"));
return;
}
// Synchronize appsink state with parent
if (!gst_element_sync_state_with_parent(c->mSink))
{
WARN(mPipeline, ("OnPadAdded: unable to sync appaink state"));
return;
}
return;
}
