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oak-editor/playback/cacher.cpp
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#include "cacher.h"
#include "project/clip.h"
#include "project/sequence.h"
#include "io/media.h"
#include "playback/audio.h"
#include "playback/playback.h"
#include "effects/effect.h"
#include "panels/timeline.h"
#include "panels/project.h"
#include "effects/transition.h"
extern "C" {
#include <libavformat/avformat.h>
#include <libavcodec/avcodec.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
#include <libavfilter/avfilter.h>
#include <libavfilter/buffersrc.h>
#include <libavfilter/buffersink.h>
#include <libavutil/opt.h>
}
#include <QDebug>
#include <QOpenGLFramebufferObject>
#include <QtMath>
#include <math.h>
int dest_format = AV_PIX_FMT_RGBA;
double bytes_to_seconds(int nb_bytes, int nb_channels, int sample_rate) {
return ((double) (nb_bytes >> 1) / nb_channels / sample_rate);
}
void apply_audio_effects(Clip* c, double timecode_start, AVFrame* frame, int nb_bytes) {
// perform all audio effects
double timecode_end;
timecode_end = timecode_start + bytes_to_seconds(nb_bytes, frame->channels, frame->sample_rate);
for (int j=0;j<c->effects.size();j++) {
Effect* e = c->effects.at(j);
if (e->is_enabled()) e->process_audio(timecode_start, timecode_end, frame->data[0], nb_bytes, 2);
}
if (c->opening_transition != NULL) {
if (c->media_type == MEDIA_TYPE_FOOTAGE) {
double transition_start = (c->clip_in / c->sequence->frame_rate);
double transition_end = transition_start + (c->opening_transition->length / c->sequence->frame_rate);
if (timecode_end < transition_end) {
double adjustment = transition_end - transition_start;
double adjusted_range_start = (timecode_start - transition_start) / adjustment;
double adjusted_range_end = (timecode_end - transition_start) / adjustment;
c->opening_transition->process_audio(adjusted_range_start, adjusted_range_end, frame->data[0], nb_bytes, false);
}
}
}
if (c->closing_transition != NULL) {
if (c->media_type == MEDIA_TYPE_FOOTAGE) {
double transition_end = (c->clip_in + c->getLength()) / c->sequence->frame_rate;
double transition_start = transition_end - (c->closing_transition->length / c->sequence->frame_rate);
if (timecode_start > transition_start) {
double adjustment = transition_end - transition_start;
double adjusted_range_start = (timecode_start - transition_start) / adjustment;
double adjusted_range_end = (timecode_end - transition_start) / adjustment;
c->closing_transition->process_audio(adjusted_range_start, adjusted_range_end, frame->data[0], nb_bytes, true);
}
}
}
}
void cache_audio_worker(Clip* c, Clip* nest) {
int written = 0;
int max_write = 16384; // TODO totally arbitrary, should probably remove
long timeline_in = c->timeline_in;
long timeline_out = c->timeline_out;
if (nest != NULL) {
timeline_in = refactor_frame_number(timeline_in, c->sequence->frame_rate, sequence->frame_rate) + nest->timeline_in;
timeline_out = refactor_frame_number(timeline_out, c->sequence->frame_rate, sequence->frame_rate) + nest->timeline_in;
}
while (written < max_write) { // TODO max_write is totally arbitrary, should probably just be an infinite loop or something
// gets one frame worth of audio and sends it to the audio buffer
AVFrame* frame;
int nb_bytes = INT_MAX;
switch (c->media_type) {
case MEDIA_TYPE_FOOTAGE:
{
frame = c->cache_A.frames[0];
// retrieve frame
bool new_frame = false;
while ((c->frame_sample_index < 0 || c->frame_sample_index >= nb_bytes) && nb_bytes > 0) {
// no more audio left in frame, get a new one
if (!c->reached_end) {
av_frame_unref(frame);
int ret;
while ((ret = av_buffersink_get_frame(c->buffersink_ctx, frame)) == AVERROR(EAGAIN)) {
if (c->reverse) {
/* TODO i have a feeling that here we're going to have to:
* - seek to previous frame
* - read it
* - convert it to 16-bit PCM LE
* - reverse it
*
* ...good luck :|
*/
if (!c->audio_just_reset) {
// get previous frame?
avcodec_flush_buffers(c->codecCtx);
av_seek_frame(c->formatCtx, c->stream->index, c->frame->pts - c->frame->pkt_duration - c->frame->pkt_duration, AVSEEK_FLAG_BACKWARD);
}
ret = retrieve_next_frame(c, c->frame);
qDebug() << c->frame->nb_samples << c->frame->linesize[0];
ret = retrieve_next_frame(c, c->frame);
qDebug() << c->frame->nb_samples << c->frame->linesize[0];
// reverse it
AVSampleFormat sample_fmt = static_cast<AVSampleFormat>(c->frame->format);
if (av_sample_fmt_is_planar(sample_fmt)) {
int sample_size = av_get_bytes_per_sample(sample_fmt);
char* temp_chars = new char[sample_size];
for (int j=0;j<c->frame->channels;j++) {
int frame_bytes = c->frame->linesize[0]/2;
int half_frame_bytes = frame_bytes >> 1;
for (int i=0;i<half_frame_bytes;i+=sample_size) {
for (int k=0;k<sample_size;k++) {
temp_chars[k] = c->frame->data[j][i+k];
}
for (int k=0;k<sample_size;k++) {
c->frame->data[j][i+k] = c->frame->data[j][frame_bytes-i-sample_size+k];
}
for (int k=0;k<sample_size;k++) {
c->frame->data[j][frame_bytes-i-sample_size+k] = temp_chars[k];
}
}
}
delete [] temp_chars;
} else {
int frame_bytes = av_samples_get_buffer_size(NULL, c->frame->channels, c->frame->nb_samples, sample_fmt, 1);
int half_frame_bytes = frame_bytes >> 1;
int sample_size = c->frame->channels*av_get_bytes_per_sample(sample_fmt);
char* temp_chars = new char[sample_size];
for (int i=0;i<half_frame_bytes;i+=sample_size) {
for (int j=0;j<sample_size;j++) {
temp_chars[j] = c->frame->data[0][i+j];
}
for (int j=0;j<sample_size;j++) {
c->frame->data[0][i+j] = c->frame->data[0][frame_bytes-i-sample_size+j];
}
for (int j=0;j<sample_size;j++) {
c->frame->data[0][frame_bytes-i-sample_size+j] = temp_chars[j];
}
}
delete [] temp_chars;
}
} else {
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;
break;
}
} else {
if (ret == AVERROR_EOF) {
// TODO likewise, I'm not sure if this if statement breaks anything (see equivalent section in cache_video_worker)
if (c->reverse) c->reached_end = true;
} else {
qDebug() << "[WARNING] Raw audio frame data could not be retrieved." << ret;
c->reached_end = true;
}
break;
}
}
if (ret < 0) {
if (ret != AVERROR_EOF) {
qDebug() << "[ERROR] Could not pull from filtergraph";
c->reached_end = true;
}
break;
}
} else {
// if there is no more data in the file, we flush the remainder out of swresample
// swr_convert_frame(c->swr_ctx, frame, NULL);
}
new_frame = true;
if (c->frame_sample_index < 0) {
c->frame_sample_index = 0;
} else {
c->frame_sample_index -= nb_bytes;
}
nb_bytes = av_samples_get_buffer_size(NULL, frame->channels, frame->nb_samples, static_cast<AVSampleFormat>(frame->format), 1);
if (c->audio_just_reset) {
// get precise sample offset for the elected clip_in from this audio frame
double timebase = av_q2d(c->stream->time_base);
double target_sts = playhead_to_seconds(c, c->audio_target_frame);
double frame_sts = (c->frame->pts * timebase);
int nb_samples = qRound((target_sts - frame_sts)*c->sequence->audio_frequency);
c->frame_sample_index = (nb_samples == 0) ? 0 : av_samples_get_buffer_size(NULL, av_get_channel_layout_nb_channels(c->sequence->audio_layout), nb_samples, AV_SAMPLE_FMT_S16, 1);
if (c->reverse) c->frame_sample_index = nb_bytes - c->frame_sample_index;
c->audio_just_reset = false;
}
if (c->audio_buffer_write == 0) c->audio_buffer_write = get_buffer_offset_from_frame(qMax(timeline_in, c->audio_target_frame));
int offset = (audio_ibuffer_read + 2048) - c->audio_buffer_write;
if (offset > 0) {
c->audio_buffer_write += offset;
c->frame_sample_index += offset;
}
}
// apply any audio effects to the data
if (nb_bytes == INT_MAX) nb_bytes = av_samples_get_buffer_size(NULL, frame->channels, frame->nb_samples, static_cast<AVSampleFormat>(frame->format), 1);
if (new_frame) {
apply_audio_effects(c, bytes_to_seconds(c->audio_buffer_write, 2, sequence->audio_frequency) + audio_ibuffer_timecode, frame, nb_bytes);
}
}
break;
case MEDIA_TYPE_TONE:
frame = c->frame;
nb_bytes = av_samples_get_buffer_size(NULL, frame->channels, frame->nb_samples, static_cast<AVSampleFormat>(frame->format), 1);
if (c->frame_sample_index < 0 || c->frame_sample_index >= nb_bytes) {
// create "new frame"
memset(c->frame->data[0], 0, nb_bytes);
apply_audio_effects(c, bytes_to_seconds(frame->pts, frame->channels, frame->sample_rate), frame, nb_bytes);
c->frame->pts += nb_bytes;
c->frame_sample_index = 0;
if (c->audio_buffer_write == 0) c->audio_buffer_write = get_buffer_offset_from_frame(qMax(timeline_in, c->audio_target_frame));
int offset = audio_ibuffer_read - c->audio_buffer_write;
if (offset > 0) {
c->audio_buffer_write += offset;
c->frame_sample_index += offset;
}
}
break;
default: // shouldn't ever get here
qDebug() << "[ERROR] Tried to cache a non-footage/tone clip";
return;
}
// mix audio into internal buffer
if (frame->nb_samples == 0) {
break;
} else {
long buffer_timeline_out = get_buffer_offset_from_frame(timeline_out);
audio_write_lock.lock();
while (c->frame_sample_index < nb_bytes
&& c->audio_buffer_write < audio_ibuffer_read+audio_ibuffer_size
&& c->audio_buffer_write < buffer_timeline_out) {
int upper_byte_index = (c->audio_buffer_write+1)%audio_ibuffer_size;
int lower_byte_index = (c->audio_buffer_write)%audio_ibuffer_size;
qint16 old_sample = static_cast<qint16>((audio_ibuffer[upper_byte_index] & 0xFF) << 8 | (audio_ibuffer[lower_byte_index] & 0xFF));
qint16 new_sample = static_cast<qint16>((frame->data[0][c->frame_sample_index+1] & 0xFF) << 8 | (frame->data[0][c->frame_sample_index] & 0xFF));
qint16 mixed_sample = mix_audio_sample(old_sample, new_sample);
audio_ibuffer[upper_byte_index] = static_cast<quint8>((mixed_sample >> 8) & 0xFF);
audio_ibuffer[lower_byte_index] = static_cast<quint8>(mixed_sample & 0xFF);
c->audio_buffer_write+=2;
c->frame_sample_index+=2;
written+=2;
}
audio_write_lock.unlock();
if (c->frame_sample_index == nb_bytes) {
c->frame_sample_index = -1;
} else {
// assume we have no more data to send
break;
}
}
}
}
void cache_video_worker(Clip* c, long playhead, ClipCache* cache) {
cache->mutex.lock();
cache->offset = playhead;
bool error = false;
int i = 0;
// double fr_ratio = c->sequence->frame_rate / c->frame_rate;
/* old swscale solution
if (!c->reached_end) {
while (i < c->cache_size) {
retrieve_next_frame_raw_data(c, cache->frames[i]);
if (c->reached_end) break;
i++;
}
}*/
/* new AVFilter solution */
if (!c->reached_end) {
while (i < c->cache_size) {
av_frame_unref(cache->frames[i]);
int ret = (c->filter_graph == NULL) ? AVERROR(EAGAIN) : av_buffersink_get_frame(c->buffersink_ctx, cache->frames[i]);
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;
}
} else {
if (ret == AVERROR_EOF) {
// TODO sorta hacky? not even sure if "reached_end" serves a proper purpose anymore
if (!c->reverse) c->reached_end = true;
} else {
qDebug() << "[WARNING] Raw frame data could not be retrieved." << ret;
error = true;
}
break;
}
} else {
if (ret != AVERROR_EOF) {
qDebug() << "[ERROR] Could not pull from filtergraph";
error = true;
}
break;
}
} else {
i++;
}
}
}
cache->write_count = i;
if (!error) {
// setting the cache to written even if it hasn't reached_end prevents playback from
// signaling a seek/reset because it wasn't able to find the frame
cache->written = true;
cache->unread = true;
}
cache->mutex.unlock();
}
void reset_cache(Clip* c, long target_frame) {
// if we seek to a whole other place in the timeline, we'll need to reset the cache with new values
switch (c->media_type) {
case MEDIA_TYPE_FOOTAGE:
{
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
avcodec_flush_buffers(c->codecCtx);
double timebase = av_q2d(c->stream->time_base);
if (c->stream->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
// seeks to nearest keyframe (target_frame represents internal clip frame)
av_seek_frame(c->formatCtx, ms->file_index, (int64_t) qFloor(clip_frame_to_seconds(c, target_frame) / timebase), AVSEEK_FLAG_BACKWARD);
// play up to the frame we actually want
long retrieved_frame = 0;
AVFrame* temp = av_frame_alloc();
do {
retrieve_next_frame(c, temp);
if (retrieved_frame == 0) {
if (target_frame != 0) retrieved_frame = floor(temp->pts * timebase * av_q2d(c->stream->avg_frame_rate));
} else {
retrieved_frame++;
}
} while (retrieved_frame < target_frame);
av_frame_free(&temp);
} else if (c->stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
// seek (target_frame represents timeline timecode in frames, not clip timecode)
int64_t timestamp = qRound(playhead_to_seconds(c, target_frame) / timebase); // TODO qRound here might lead to clicking? or might fix it... who knows
if (c->reverse) {
timestamp--;
}
av_seek_frame(c->formatCtx, ms->file_index, timestamp, AVSEEK_FLAG_BACKWARD);
c->audio_target_frame = target_frame;
c->frame_sample_index = -1;
c->audio_just_reset = true;
}
}
}
break;
case MEDIA_TYPE_TONE:
c->audio_target_frame = target_frame;
c->frame_sample_index = -1;
c->frame->pts = 0;
break;
}
}
Cacher::Cacher(Clip* c) : clip(c) {}
AVSampleFormat sample_format = AV_SAMPLE_FMT_S16;
void open_clip_worker(Clip* clip) {
switch (clip->media_type) {
case MEDIA_TYPE_FOOTAGE:
{
// opens file resource for FFmpeg and prepares Clip struct for playback
Media* m = static_cast<Media*>(clip->media);
QByteArray ba = m->url.toUtf8();
const char* filename = ba.constData();
MediaStream* ms = m->get_stream_from_file_index(clip->track < 0, clip->media_stream);
int errCode = avformat_open_input(
&clip->formatCtx,
filename,
NULL,
NULL
);
if (errCode != 0) {
char err[1024];
av_strerror(errCode, err, 1024);
qDebug() << "[ERROR] Could not open" << filename << "-" << err;
}
errCode = avformat_find_stream_info(clip->formatCtx, NULL);
if (errCode < 0) {
char err[1024];
av_strerror(errCode, err, 1024);
qDebug() << "[ERROR] Could not open" << filename << "-" << err;
}
av_dump_format(clip->formatCtx, 0, filename, 0);
clip->stream = clip->formatCtx->streams[ms->file_index];
clip->codec = avcodec_find_decoder(clip->stream->codecpar->codec_id);
clip->codecCtx = avcodec_alloc_context3(clip->codec);
avcodec_parameters_to_context(clip->codecCtx, clip->stream->codecpar);
AVDictionary* opts = NULL;
// optimized decoding settings
if (clip->stream->codecpar->codec_id != AV_CODEC_ID_PNG &&
clip->stream->codecpar->codec_id != AV_CODEC_ID_APNG &&
clip->stream->codecpar->codec_id != AV_CODEC_ID_TIFF &&
clip->stream->codecpar->codec_id != AV_CODEC_ID_PSD) {
av_dict_set(&opts, "threads", "auto", 0);
}
if (clip->stream->codecpar->codec_id == AV_CODEC_ID_H264) {
av_dict_set(&opts, "tune", "fastdecode", 0);
av_dict_set(&opts, "tune", "zerolatency", 0);
}
// Open codec
if (avcodec_open2(clip->codecCtx, clip->codec, &opts) < 0) {
qDebug() << "[ERROR] Could not open codec";
}
// allocate filtergraph
clip->filter_graph = avfilter_graph_alloc();
if (clip->filter_graph == NULL) {
qDebug() << "couldn't create filtergraph";
}
char filter_args[512];
if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
// create memory cache for video
clip->cache_size = (ms->infinite_length) ? 1 : ceil(av_q2d(clip->stream->avg_frame_rate)/4); // cache is half a second in total
clip->cache_A.frames = new AVFrame* [clip->cache_size];
clip->cache_B.frames = new AVFrame* [clip->cache_size];
for (int i=0;i<clip->cache_size;i++) {
clip->cache_A.frames[i] = av_frame_alloc();
clip->cache_B.frames[i] = av_frame_alloc();
}
snprintf(filter_args, sizeof(filter_args), "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
clip->stream->codecpar->width,
clip->stream->codecpar->height,
clip->stream->codecpar->format,
clip->stream->time_base.num,
clip->stream->time_base.den,
clip->stream->codecpar->sample_aspect_ratio.num,
clip->stream->codecpar->sample_aspect_ratio.den
);
avfilter_graph_create_filter(&clip->buffersrc_ctx, avfilter_get_by_name("buffer"), "in", filter_args, 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 };
if (av_opt_set_int_list(clip->buffersink_ctx, "pix_fmts", pix_fmts, AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN) < 0) {
qDebug() << "[ERROR] Could not set output pixel format";
}
avfilter_link(clip->buffersrc_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 = av_get_default_channel_layout(clip->stream->codecpar->channels);
// init resampling context
/*
clip->swr_ctx = swr_alloc_set_opts(
NULL,
sequence->audio_layout,
static_cast<AVSampleFormat>(sample_format),
sequence->audio_frequency / clip->speed,
clip->codecCtx->channel_layout,
static_cast<AVSampleFormat>(clip->stream->codecpar->format),
clip->stream->codecpar->sample_rate,
0,
NULL
);
swr_init(clip->swr_ctx);
*/
// set up cache
clip->cache_A.frames = new AVFrame* [1];
clip->cache_A.frames[0] = av_frame_alloc();
snprintf(filter_args, sizeof(filter_args), "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%" PRIx64,
clip->stream->time_base.num,
clip->stream->time_base.den,
clip->stream->codecpar->sample_rate,
av_get_sample_fmt_name(clip->codecCtx->sample_fmt),
clip->codecCtx->channel_layout
);
avfilter_graph_create_filter(&clip->buffersrc_ctx, avfilter_get_by_name("abuffer"), "in", filter_args, NULL, clip->filter_graph);
avfilter_graph_create_filter(&clip->buffersink_ctx, avfilter_get_by_name("abuffersink"), "out", NULL, NULL, clip->filter_graph);
enum AVSampleFormat sample_fmts[] = { sample_format, static_cast<AVSampleFormat>(-1) };
if (av_opt_set_int_list(clip->buffersink_ctx, "sample_fmts", sample_fmts, -1, AV_OPT_SEARCH_CHILDREN) < 0) {
qDebug() << "[ERROR] Could not set output sample format";
}
int target_sample_rate = sequence->audio_frequency;
if (qFuzzyCompare(clip->speed, 1.0)) {
avfilter_link(clip->buffersrc_ctx, 0, clip->buffersink_ctx, 0);
} else if (clip->maintain_audio_pitch) {
char speed_param[10];
snprintf(speed_param, sizeof(speed_param), "%f", clip->speed);
AVFilterContext* tempo_filter;
avfilter_graph_create_filter(&tempo_filter, avfilter_get_by_name("atempo"), "atempo", speed_param, NULL, clip->filter_graph);
avfilter_link(clip->buffersrc_ctx, 0, tempo_filter, 0);
avfilter_link(tempo_filter, 0, clip->buffersink_ctx, 0);
} else {
target_sample_rate = qRound(sequence->audio_frequency / clip->speed);
avfilter_link(clip->buffersrc_ctx, 0, clip->buffersink_ctx, 0);
}
int sample_rates[] = { target_sample_rate, 0 };
if (av_opt_set_int_list(clip->buffersink_ctx, "sample_rates", sample_rates, 0, AV_OPT_SEARCH_CHILDREN) < 0) {
qDebug() << "[ERROR] Could not set output sample rates";
}
avfilter_graph_config(clip->filter_graph, NULL);
clip->audio_reset = true;
}
clip->frame = av_frame_alloc();
}
break;
case MEDIA_TYPE_TONE:
clip->frame = av_frame_alloc();
clip->frame->format = sample_format;
clip->frame->channel_layout = sequence->audio_layout;
clip->frame->channels = av_get_channel_layout_nb_channels(clip->frame->channel_layout);
clip->frame->sample_rate = sequence->audio_frequency;
clip->frame->nb_samples = 2048;
av_frame_make_writable(clip->frame);
if (av_frame_get_buffer(clip->frame, 0)) {
qDebug() << "[ERROR] Could not allocate buffer for tone clip";
}
clip->audio_reset = true;
break;
}
for (int i=0;i<clip->effects.size();i++) {
clip->effects.at(i)->open();
}
clip->finished_opening = true;
qDebug() << "[INFO] Clip opened on track" << clip->track;
}
void cache_clip_worker(Clip* clip, long playhead, bool write_A, bool write_B, bool reset, Clip* nest) {
if (reset) {
// note: for video, playhead is in "internal clip" frames - for audio, it's the timeline playhead
reset_cache(clip, playhead);
clip->audio_reset = false;
}
switch (clip->media_type) {
case MEDIA_TYPE_FOOTAGE:
if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
if (write_A) {
cache_video_worker(clip, playhead, &clip->cache_A);
playhead += clip->cache_size;
}
if (write_B) {
cache_video_worker(clip, playhead, &clip->cache_B);
}
} else if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
cache_audio_worker(clip, nest);
}
break;
case MEDIA_TYPE_TONE:
cache_audio_worker(clip, nest);
break;
}
}
void close_clip_worker(Clip* clip) {
clip->finished_opening = false;
if (clip->media_type == MEDIA_TYPE_FOOTAGE) {
// closes ffmpeg file handle and frees any memory used for caching
MediaStream* ms = static_cast<Media*>(clip->media)->get_stream_from_file_index(clip->track < 0, clip->media_stream);
avfilter_graph_free(&clip->filter_graph);
avcodec_close(clip->codecCtx);
avcodec_free_context(&clip->codecCtx);
avformat_close_input(&clip->formatCtx);
for (int i=0;i<clip->cache_size;i++) {
av_frame_free(&clip->cache_A.frames[i]);
if (!ms->infinite_length) av_frame_free(&clip->cache_B.frames[i]);
}
delete [] clip->cache_A.frames;
if (!ms->infinite_length) delete [] clip->cache_B.frames;
}
av_frame_free(&clip->frame);
clip->reset();
qDebug() << "[INFO] Clip closed on track" << clip->track;
}
void Cacher::run() {
// open_lock is used to prevent the clip from being destroyed before the cacher has closed it properly
clip->lock.lock();
clip->finished_opening = false;
clip->open = true;
caching = true;
open_clip_worker(clip);
while (caching) {
clip->can_cache.wait(&clip->lock);
if (!caching) {
break;
} else {
cache_clip_worker(clip, playhead, write_A, write_B, reset, nest);
}
}
close_clip_worker(clip);
clip->lock.unlock();
clip->open_lock.unlock();
}
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