static gboolean
audio_trim_event (GstPad * pad, GstEvent *event)
{
if (GST_EVENT_TYPE(event) == GST_EVENT_EOS) {
GList *b;
gint64 last_offset;
AudioTrim *filter;
g_debug("Got EOS");
filter = AUDIO_TRIM (GST_OBJECT_PARENT (pad));
b = g_list_last(filter->buffers);
if (b) {
filter->accumulator = 0.0;
last_offset = GST_BUFFER_OFFSET_END((GstBuffer*)b->data);
last_offset -= time_to_sample(filter, filter->end_skip);
while(b) {
gint64 pos = find_not_silence_rev(filter, b->data, last_offset);
if (pos != GST_BUFFER_OFFSET_NONE) {
pos += time_to_sample(filter, filter->post_silence);
send_buffers_before(filter,pos);
filter->sound_duration=sample_to_time(filter, pos - filter->ref_time);
break;
}
b = g_list_previous(b);
}
}
}
return gst_pad_event_default (pad, event);
}
/* chain function
* this function does the actual processing
*/
static GstFlowReturn
audio_trim_chain (GstPad * pad, GstBuffer * buf)
{
AudioTrim *filter;
g_assert(GST_BUFFER_OFFSET(buf) != GST_BUFFER_OFFSET_NONE);
g_assert(GST_BUFFER_OFFSET_END(buf) != GST_BUFFER_OFFSET_NONE);
filter = AUDIO_TRIM (GST_OBJECT_PARENT (pad));
while(buf) {
g_assert(GST_IS_BUFFER(buf));
switch(filter->trim_state) {
case AUDIO_TRIM_NOT_STARTED:
filter->ref_time = (GST_BUFFER_OFFSET(buf)
+ time_to_sample(filter, filter->start_skip));
if (filter->empty_start_packet) {
GstFlowReturn ret;
GstBuffer *first;
first = gst_buffer_new_and_alloc (sizeof(gfloat));
*(gfloat*)GST_BUFFER_DATA(first) = 0.0;
GST_BUFFER_SIZE(first) = 4;
GST_BUFFER_OFFSET(first) = GST_BUFFER_OFFSET(buf);
GST_BUFFER_OFFSET_END(first) = GST_BUFFER_OFFSET(buf);
GST_BUFFER_TIMESTAMP(first) = GST_BUFFER_TIMESTAMP(buf);
GST_BUFFER_DURATION(first) = 0;
GST_BUFFER_CAPS(first) = gst_caps_ref(GST_BUFFER_CAPS(buf));
ret = gst_pad_push(filter->srcpad, first);
if (ret != GST_FLOW_OK) {
gst_buffer_unref(buf);
return ret;
}
}
filter->trim_state = AUDIO_TRIM_START_SKIP;
break;
case AUDIO_TRIM_START_SKIP:
if (GST_BUFFER_OFFSET_END(buf) <= filter->ref_time) {
gst_buffer_unref(buf); /* Ignore buffer completely */
} else {
GstBuffer *tail = buffer_tail(filter, buf, filter->ref_time);
if (buf) gst_buffer_unref(buf);
buf = tail;
filter->trim_state = AUDIO_TRIM_START_SILENCE;
}
break;
case AUDIO_TRIM_START_SILENCE:
{
guint64 offset = find_not_silence(filter, buf);
if (offset == GST_BUFFER_OFFSET_NONE) {
while(filter->buffered > filter->pre_silence) {
GstBuffer *old = filter->buffers->data;
filter->buffered -= GST_BUFFER_DURATION(old);
gst_buffer_unref(old);
filter->buffers =
g_list_delete_link(filter->buffers, filter->buffers);
}
save_buffer(filter, buf);
buf = NULL;
} else {
GstBuffer *head;
GstBuffer *tail;
GstFlowReturn ret;
gint64 clip_start;
clip_start = offset - time_to_sample(filter, filter->pre_silence);
ret = send_buffers_after(filter, clip_start);
if (ret != GST_FLOW_OK) {
gst_buffer_unref(buf);
return ret;
}
head = buffer_slice(filter, buf, clip_start, offset);
if (head) {
ret = gst_pad_push(filter->srcpad, head);
if (ret != GST_FLOW_OK) {
gst_buffer_unref(buf);
return ret;
}
}
tail = buffer_tail(filter, buf, offset);
filter->sound_duration =
sample_to_time(filter, GST_BUFFER_OFFSET_END(buf) - clip_start);
filter->ref_time = clip_start;
gst_buffer_unref(buf);
buf = tail;
filter->trim_state = AUDIO_TRIM_NOT_SILENCE;
g_debug("Got sound");
}
}
break;
case AUDIO_TRIM_NOT_SILENCE:
{
GstFlowReturn ret;
filter->sound_duration += GST_BUFFER_DURATION(buf);
while(filter->buffered > filter->max_silence_duration) {
GstBuffer *old = filter->buffers->data;
filter->buffered -= GST_BUFFER_DURATION(old);
filter->buffers = g_list_delete_link(filter->buffers,filter->buffers);
ret = gst_pad_push(filter->srcpad, old);
if (ret != GST_FLOW_OK) {
gst_buffer_unref(buf);
return ret;
}
}
save_buffer(filter, buf);
buf = 0;
}
break;
default:
gst_buffer_unref(buf);
buf = NULL;
}
}
return GST_FLOW_OK;
}
// reported by everwanna:
// av out of sync because:
// audio track 0 without stss, seek to the exact time.
// video track 1 with stss, seek to the nearest key frame time.
//
// fixed:
// first pass we get the new aligned times for traks with an stss present
// second pass is for traks without an stss
static int get_aligned_start_and_end(mp4_context_t const* mp4_context,
int64_t start, int64_t end,
unsigned int* trak_sample_start,
unsigned int* trak_sample_end)
{
unsigned int pass;
moov_t const* moov = mp4_context->moov;
long moov_time_scale = moov->mvhd_->timescale_;
for(pass = 0; pass != 2; ++pass)
{
unsigned int i;
for(i = 0; i != moov->tracks_; ++i)
{
trak_t const* trak = moov->traks_[i];
long trak_time_scale = trak->mdia_->mdhd_->timescale_;
if(trak->samples_size_ == 0)
{
trak_sample_start[i] = 0;
trak_sample_end[i] = 0;
continue;
}
// get start
{
unsigned int sample_index = time_to_sample(trak,
moov_time_to_trak_time(start, moov_time_scale, trak_time_scale));
// backtrack to nearest keyframe
if(!trak->samples_[sample_index].is_ss_)
{
while(sample_index && !trak->samples_[sample_index].is_ss_)
{
--sample_index;
}
start = trak_time_to_moov_time(trak->samples_[sample_index].pts_,
moov_time_scale, trak_time_scale);
}
trak_sample_start[i] = sample_index;
// MP4_INFO("ts=%"PRId64" (moov time)\n", start);
MP4_INFO("ts=%.2f (seconds)\n",
trak->samples_[sample_index].pts_ / (float)trak_time_scale);
}
// get end
{
unsigned int sample_index;
if(end == 0)
{
// The default is till-the-end of the track
sample_index = trak->samples_size_;
}
else
{
sample_index = time_to_sample(trak,
moov_time_to_trak_time(end, moov_time_scale, trak_time_scale));
}
// backtrack to nearest keyframe
if(!trak->samples_[sample_index].is_ss_)
{
while(sample_index && !trak->samples_[sample_index].is_ss_)
{
--sample_index;
}
end = trak_time_to_moov_time(trak->samples_[sample_index].pts_,
moov_time_scale, trak_time_scale);
}
trak_sample_end[i] = sample_index;
// MP4_INFO("te=%"PRId64" (moov time)\n", end);
MP4_INFO("te=%.2f (seconds)\n",
trak->samples_[sample_index].pts_ / (float)trak_time_scale);
}
}
}
MP4_INFO("final start=%"PRId64"\n", start);
MP4_INFO("final end=%"PRId64"\n", end);
if(end && start >= end)
{
return 0;
}
return 1;
}
