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several improvements

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This commit is contained in:
itsmattkc
2018-10-13 13:08:20 +11:00
parent 2e577330b9
commit a285f10e6a
24 changed files with 179 additions and 57 deletions
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playback/cacher.cpp
+38 -5
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@@ -26,6 +26,9 @@ extern "C" {
#include <QtMath>
#include <math.h>
// temp debug shit
//#define AUDIOWARNINGS
int dest_format = AV_PIX_FMT_RGBA;
double bytes_to_seconds(int nb_bytes, int nb_channels, int sample_rate) {
@@ -97,9 +100,11 @@ void cache_audio_worker(Clip* c, Clip* nest) {
avcodec_flush_buffers(c->codecCtx);
int64_t backtrack_seek = qMax(c->rev_target - static_cast<int64_t>(av_q2d(av_inv_q(c->stream->time_base))), static_cast<int64_t>(0));
av_seek_frame(c->formatCtx, c->stream->index, backtrack_seek, AVSEEK_FLAG_BACKWARD);
#ifdef AUDIOWARNINGS
if (backtrack_seek == 0) {
qDebug() << "backtracked to 0";
}
#endif
}
do {
@@ -120,7 +125,9 @@ void cache_audio_worker(Clip* c, Clip* nest) {
if (!c->reverse) {
c->reached_end = true;
} else {
#ifdef AUDIOWARNINGS
qDebug() << "reached EOF";
#endif
}
} else {
qDebug() << "[WARNING] Raw audio frame data could not be retrieved." << ret;
@@ -136,7 +143,9 @@ void cache_audio_worker(Clip* c, Clip* nest) {
c->reached_end = true;
break;
} else {
#ifdef AUDIOWARNINGS
qDebug() << "reached EOF";
#endif
if (!c->reverse) break;
}
}
@@ -146,19 +155,25 @@ void cache_audio_worker(Clip* c, Clip* nest) {
AVFrame* rev_frame = c->cache_A.frames[1];
if (ret != AVERROR_EOF) {
if (loop == 2) {
#ifdef AUDIOWARNINGS
qDebug() << "starting rev_frame";
#endif
rev_frame->nb_samples = 0;
rev_frame->pts = c->frame->pkt_pts;
}
int offset = rev_frame->nb_samples * av_get_bytes_per_sample(static_cast<AVSampleFormat>(rev_frame->format)) * rev_frame->channels;
#ifdef AUDIOWARNINGS
qDebug() << "offset 1:" << offset;
qDebug() << "retrieved samples:" << frame->nb_samples << "size:" << (frame->nb_samples * av_get_bytes_per_sample(static_cast<AVSampleFormat>(frame->format)) * frame->channels);
#endif
memcpy(
rev_frame->data[0]+offset,
frame->data[0],
(frame->nb_samples * av_get_bytes_per_sample(static_cast<AVSampleFormat>(frame->format)) * frame->channels)
);
#ifdef AUDIOWARNINGS
qDebug() << "pts:" << c->frame->pts << "dur:" << c->frame->pkt_duration << "rev_target:" << c->rev_target << "offset:" << offset << "limit:" << rev_frame->linesize[0];
#endif
}
rev_frame->nb_samples += frame->nb_samples;
@@ -173,9 +188,13 @@ void cache_audio_worker(Clip* c, Clip* nest) {
rev_frame->nb_samples -= cutoff;
}*/
#ifdef AUDIOWARNINGS
qDebug() << "pre cutoff deets::: rev_frame.pts:" << rev_frame->pts << "rev_frame.nb_samples" << rev_frame->nb_samples << "rev_target:" << c->rev_target;
#endif
rev_frame->nb_samples = qRound(static_cast<double>(c->rev_target - rev_frame->pts) / c->stream->codecpar->sample_rate * sequence->audio_frequency);
#ifdef AUDIOWARNINGS
qDebug() << "post cutoff deets::" << rev_frame->nb_samples;
#endif
int frame_size = rev_frame->nb_samples * rev_frame->channels * av_get_bytes_per_sample(static_cast<AVSampleFormat>(rev_frame->format));
int half_frame_size = frame_size >> 1;
@@ -203,7 +222,9 @@ void cache_audio_worker(Clip* c, Clip* nest) {
loop++;
#ifdef AUDIOWARNINGS
qDebug() << "loop" << loop;
#endif
} else {
frame->pts = c->frame->pts;
break;
@@ -228,15 +249,19 @@ void cache_audio_worker(Clip* c, Clip* nest) {
// get precise sample offset for the elected clip_in from this audio frame
double target_sts = playhead_to_seconds(c, c->audio_target_frame);
double frame_sts = (frame->pts * timebase);
int nb_samples = qRound((target_sts - frame_sts)*c->sequence->audio_frequency);
qDebug() << "fsts:" << frame_sts << "tsts:" << target_sts << "nbs:" << nb_samples << "nbb:" << nb_bytes << "rev_targetToSec:" << (c->rev_target * timebase);
int nb_samples = qRound((target_sts - frame_sts)*c->sequence->audio_frequency);
c->frame_sample_index = nb_samples * 4;
#ifdef AUDIOWARNINGS
qDebug() << "fsts:" << frame_sts << "tsts:" << target_sts << "nbs:" << nb_samples << "nbb:" << nb_bytes << "rev_targetToSec:" << (c->rev_target * timebase);
qDebug() << "fsi-calc:" << c->frame_sample_index;
#endif
if (c->reverse) c->frame_sample_index = nb_bytes - c->frame_sample_index;
c->audio_just_reset = false;
}
#ifdef AUDIOWARNINGS
qDebug() << "fsi-post-post:" << c->frame_sample_index;
#endif
if (c->audio_buffer_write == 0) c->audio_buffer_write = get_buffer_offset_from_frame(qMax(timeline_in, c->audio_target_frame));
@@ -255,7 +280,9 @@ void cache_audio_worker(Clip* c, Clip* nest) {
if (c->reverse) frame = c->cache_A.frames[1];
#ifdef AUDIOWARNINGS
qDebug() << "j" << c->frame_sample_index << nb_bytes;
#endif
// apply any audio effects to the data
if (nb_bytes == INT_MAX) nb_bytes = frame->nb_samples * av_get_bytes_per_sample(static_cast<AVSampleFormat>(frame->format)) * frame->channels;
@@ -290,7 +317,6 @@ void cache_audio_worker(Clip* c, Clip* nest) {
if (frame->nb_samples == 0) {
break;
} else {
qDebug() << "re:" << c->reached_end;
long buffer_timeline_out = get_buffer_offset_from_frame(timeline_out);
audio_write_lock.lock();
while (c->frame_sample_index < nb_bytes
@@ -308,7 +334,9 @@ void cache_audio_worker(Clip* c, Clip* nest) {
c->audio_buffer_write+=2;
c->frame_sample_index+=2;
}
#ifdef AUDIOWARNINGS
if (c->audio_buffer_write >= buffer_timeline_out) qDebug() << "timeline out at fsi" << c->frame_sample_index << "of frame ts" << c->frame->pts;
#endif
audio_write_lock.unlock();
if (c->frame_sample_index == nb_bytes) {
@@ -318,7 +346,7 @@ void cache_audio_worker(Clip* c, Clip* nest) {
break;
}
qDebug() << "ended" << c->frame_sample_index << nb_bytes;
// qDebug() << "ended" << c->frame_sample_index << nb_bytes;
}
if (c->reached_end) {
frame->nb_samples = 0;
@@ -386,6 +414,8 @@ void cache_video_worker(Clip* c, long playhead, ClipCache* cache) {
break;
}
} else {
qDebug() << "retrieved PTS:" << cache->frames[i]->pts << c->stream->time_base.num << "/" << c->stream->time_base.den;
cache->written = true;
i++;
cache->write_count = i;
@@ -412,6 +442,7 @@ void reset_cache(Clip* c, long target_frame) {
switch (c->media_type) {
case MEDIA_TYPE_FOOTAGE:
{
c->reached_end = false;
MediaStream* ms = static_cast<Media*>(c->media)->get_stream_from_file_index(c->track < 0, c->media_stream);
if (!ms->infinite_length) {
// flush ffmpeg codecs
@@ -445,7 +476,7 @@ void reset_cache(Clip* c, long target_frame) {
av_frame_unref(temp);
av_frame_free(&temp);
} else {
av_seek_frame(c->formatCtx, ms->file_index, 0, AVSEEK_FLAG_BACKWARD);
av_seek_frame(c->formatCtx, ms->file_index, c->stream->start_time, AVSEEK_FLAG_BACKWARD);
}
} else if (c->stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
// seek (target_frame represents timeline timecode in frames, not clip timecode)
@@ -453,7 +484,9 @@ void reset_cache(Clip* c, long target_frame) {
if (c->reverse) {
c->rev_target = timestamp;
timestamp -= av_q2d(av_inv_q(c->stream->time_base));
#ifdef AUDIOWARNINGS
qDebug() << "seeking to" << timestamp << "(originally" << c->rev_target << ")";
#endif
}
av_seek_frame(c->formatCtx, ms->file_index, timestamp, AVSEEK_FLAG_BACKWARD);
c->audio_target_frame = target_frame;