extremely limited speed support implemented #120

This commit is contained in:
itsmattkc
2018-10-01 14:51:36 +10:00
parent c21a21c219
commit ac153e4cff
14 changed files with 143 additions and 91 deletions
+21 -16
View File
@@ -170,7 +170,6 @@ void cache_audio_worker(Clip* c, Clip* nest) {
} else {
long buffer_timeline_out = get_buffer_offset_from_frame(timeline_out);
audio_write_lock.lock();
qDebug() << "starting to write:" << audio_ibuffer_read;
while (c->frame_sample_index < nb_bytes
&& c->audio_buffer_write < audio_ibuffer_read+audio_ibuffer_size
&& c->audio_buffer_write < buffer_timeline_out) {
@@ -207,6 +206,8 @@ void cache_video_worker(Clip* c, long playhead, ClipCache* cache) {
int i = 0;
double fr_ratio = qRound((c->sequence->frame_rate / c->frame_rate)*100)*0.01;
/* old swscale solution
if (!c->reached_end) {
while (i < c->cache_size) {
@@ -227,11 +228,19 @@ void cache_video_worker(Clip* c, long playhead, ClipCache* cache) {
if (ret < 0) {
if (ret == AVERROR(EAGAIN)) {
ret = retrieve_next_frame(c, c->frame);
if (ret >= 0) {
if ((ret = av_buffersrc_add_frame_flags(c->buffersrc_ctx, c->frame, AV_BUFFERSRC_FLAG_KEEP_REF)) < 0) {
qDebug() << "[ERROR] Could not feed filtergraph -" << ret;
error = true;
break;
// optimization: mathematically determine based on the sequence and clips' frame rates whether this frame will actually be shown
double timeline_frame = (i+cache->offset) * fr_ratio;
if (qFloor(timeline_frame) == timeline_frame) {
if ((ret = av_buffersrc_add_frame_flags(c->buffersrc_ctx, c->frame, AV_BUFFERSRC_FLAG_KEEP_REF)) < 0) {
qDebug() << "[ERROR] Could not feed filtergraph -" << ret;
error = true;
break;
}
} else {
i++;
}
} else {
if (ret == AVERROR_EOF) {
@@ -442,29 +451,25 @@ void open_clip_worker(Clip* clip) {
clip->stream->codecpar->sample_aspect_ratio.den
);
char errormsg[100];
avfilter_graph_create_filter(&clip->buffersrc_ctx, avfilter_get_by_name("buffer"), "in", args, NULL, clip->filter_graph);
AVFilter* buffersrc = avfilter_get_by_name("buffer");
AVFilter* buffersink = avfilter_get_by_name("buffersink");
AVFilterContext* negate_ctx;
avfilter_graph_create_filter(&negate_ctx, avfilter_get_by_name("setpts"), "rev", "0.5*PTS", NULL, clip->filter_graph);
if (buffersrc == NULL || buffersink == NULL) {
qDebug() << "lol";
}
avfilter_graph_create_filter(&clip->buffersrc_ctx, buffersrc, "in", args, NULL, clip->filter_graph);
avfilter_graph_create_filter(&clip->buffersink_ctx, buffersink, "out", NULL, NULL, clip->filter_graph);
avfilter_graph_create_filter(&clip->buffersink_ctx, avfilter_get_by_name("buffersink"), "out", NULL, NULL, clip->filter_graph);
enum AVPixelFormat pix_fmts[] = { static_cast<AVPixelFormat>(dest_format), AV_PIX_FMT_NONE };
av_opt_set_int_list(clip->buffersink_ctx, "pix_fmts", pix_fmts, AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
avfilter_link(clip->buffersrc_ctx, 0, clip->buffersink_ctx, 0);
avfilter_link(clip->buffersrc_ctx, 0, negate_ctx, 0);
avfilter_link(negate_ctx, 0, clip->buffersink_ctx, 0);
avfilter_graph_config(clip->filter_graph, NULL);
} else if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
// if FFmpeg can't pick up the channel layout (usually WAV), assume
// based on channel count (doesn't support surround sound sources yet)
if (clip->codecCtx->channel_layout == 0) {
clip->codecCtx->channel_layout = guess_layout_from_channels(clip->stream->codecpar->channels);
clip->codecCtx->channel_layout = av_get_default_channel_layout(clip->stream->codecpar->channels);
}
// init resampling context