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b3bcddf96b7fd6219506a8650a7f7b8192ec066b
oak-editor/playback/cacher.cpp
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itsmattkc b3bcddf96b fixed broken includes
2018-11-13 02:56:31 +11:00

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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 "project/effect.h"
#include "panels/timeline.h"
#include "panels/project.h"
#include "playback/audio.h"
#include "panels/panels.h"
#include "panels/viewer.h"
#include "debug.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 <QOpenGLFramebufferObject>
#include <QtMath>
#include <QAudioOutput>
#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) {
return ((double) (nb_bytes >> 1) / nb_channels / sample_rate);
}
void apply_audio_effects(Clip* c, double timecode_start, AVFrame* frame, int nb_bytes, QVector<Clip*> nests) {
// perform all aud io 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 = ((c->clip_in + 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, TRAN_TYPE_OPEN);
}
}
}
if (c->closing_transition != NULL) {
if (c->media_type == MEDIA_TYPE_FOOTAGE) {
double transition_start = (c->clip_in + c->getLength() - c->closing_transition->length) / c->sequence->frame_rate;
double transition_end = (c->clip_in + c->getLength()) / 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, TRAN_TYPE_CLOSE);
}
}
}
if (!nests.isEmpty()) {
Clip* next_nest = nests.last();
nests.removeLast();
apply_audio_effects(next_nest, timecode_start + (((double)c->timeline_in-c->clip_in)/c->sequence->frame_rate), frame, nb_bytes, nests);
}
}
#define AUDIO_BUFFER_PADDING 2048
void cache_audio_worker(Clip* c, bool scrubbing, QVector<Clip*>& nests) {
long timeline_in = c->timeline_in;
long timeline_out = c->timeline_out;
long target_frame = c->audio_target_frame;
long frame_skip = 0;
double last_fr = c->sequence->frame_rate;
if (!nests.isEmpty()) {
for (int i=nests.size()-1;i>=0;i--) {
timeline_in = refactor_frame_number(timeline_in, last_fr, nests.at(i)->sequence->frame_rate) + nests.at(i)->timeline_in - nests.at(i)->clip_in;
timeline_out = refactor_frame_number(timeline_out, last_fr, nests.at(i)->sequence->frame_rate) + nests.at(i)->timeline_in - nests.at(i)->clip_in;
target_frame = refactor_frame_number(target_frame, last_fr, nests.at(i)->sequence->frame_rate) + nests.at(i)->timeline_in - nests.at(i)->clip_in;
timeline_out = qMin(timeline_out, nests.at(i)->timeline_out);
frame_skip = refactor_frame_number(frame_skip, last_fr, nests.at(i)->sequence->frame_rate);
long validator = nests.at(i)->timeline_in - timeline_in;
if (validator > 0) {
frame_skip += validator;
//timeline_in = nests.at(i)->timeline_in;
}
last_fr = nests.at(i)->sequence->frame_rate;
}
}
while (true) {
AVFrame* frame;
int nb_bytes = INT_MAX;
switch (c->media_type) {
case MEDIA_TYPE_FOOTAGE:
{
double timebase = av_q2d(c->stream->time_base);
frame = c->queue.at(0);
// retrieve frame
bool new_frame = false;
while ((c->frame_sample_index == -1 || c->frame_sample_index >= nb_bytes) && nb_bytes > 0) {
// no more audio left in frame, get a new one
if (!c->reached_end) {
int loop = 0;
if (c->reverse && !c->audio_just_reset) {
avcodec_flush_buffers(c->codecCtx);
c->reached_end = false;
int64_t backtrack_seek = qMax(c->reverse_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) {
dout << "backtracked to 0";
}
#endif
}
do {
av_frame_unref(frame);
int ret;
while ((ret = av_buffersink_get_frame(c->buffersink_ctx, frame)) == 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) {
dout << "[ERROR] Could not feed filtergraph -" << ret;
break;
}
} else {
if (ret == AVERROR_EOF) {
#ifdef AUDIOWARNINGS
dout << "reached EOF while reading";
#endif
// TODO revise usage of reached_end in audio
if (!c->reverse) {
c->reached_end = true;
} else {
}
} else {
dout << "[WARNING] Raw audio frame data could not be retrieved." << ret;
c->reached_end = true;
}
break;
}
}
if (ret < 0) {
if (ret != AVERROR_EOF) {
dout << "[ERROR] Could not pull from filtergraph";
c->reached_end = true;
break;
} else {
#ifdef AUDIOWARNINGS
dout << "reached EOF while pulling from filtergraph";
#endif
if (!c->reverse) break;
}
}
if (c->reverse) {
if (loop > 1) {
AVFrame* rev_frame = c->queue.at(1);
if (ret != AVERROR_EOF) {
if (loop == 2) {
#ifdef AUDIOWARNINGS
dout << "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
dout << "offset 1:" << offset;
dout << "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
dout << "pts:" << c->frame->pts << "dur:" << c->frame->pkt_duration << "rev_target:" << c->reverse_target << "offset:" << offset << "limit:" << rev_frame->linesize[0];
#endif
}
rev_frame->nb_samples += frame->nb_samples;
if ((c->frame->pts >= c->reverse_target) || (ret == AVERROR_EOF)) {
/*
#ifdef AUDIOWARNINGS
dout << "time for the end of rev cache" << rev_frame->nb_samples << c->rev_target << c->frame->pts << c->frame->pkt_duration << c->frame->nb_samples;
dout << "diff:" << (c->frame->pkt_pts + c->frame->pkt_duration) - c->rev_target;
#endif
int cutoff = qRound64((((c->frame->pkt_pts + c->frame->pkt_duration) - c->reverse_target) * timebase) * audio_output->format().sampleRate());
if (cutoff > 0) {
#ifdef AUDIOWARNINGS
dout << "cut off" << cutoff << "samples (rate:" << audio_output->format().sampleRate() << ")";
#endif
rev_frame->nb_samples -= cutoff;
}
*/
#ifdef AUDIOWARNINGS
dout << "pre cutoff deets::: rev_frame.pts:" << rev_frame->pts << "rev_frame.nb_samples" << rev_frame->nb_samples << "rev_target:" << c->reverse_target;
#endif
rev_frame->nb_samples = qRound64(static_cast<double>(c->reverse_target - rev_frame->pts) / c->stream->codecpar->sample_rate * (audio_output->format().sampleRate() / c->speed));
#ifdef AUDIOWARNINGS
dout << "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;
int sample_size = rev_frame->channels*av_get_bytes_per_sample(static_cast<AVSampleFormat>(rev_frame->format));
char* temp_chars = new char[sample_size];
for (int i=0;i<half_frame_size;i+=sample_size) {
for (int j=0;j<sample_size;j++) {
temp_chars[j] = rev_frame->data[0][i+j];
}
for (int j=0;j<sample_size;j++) {
rev_frame->data[0][i+j] = rev_frame->data[0][frame_size-i-sample_size+j];
}
for (int j=0;j<sample_size;j++) {
rev_frame->data[0][frame_size-i-sample_size+j] = temp_chars[j];
}
}
delete [] temp_chars;
c->reverse_target = rev_frame->pts;
frame = rev_frame;
break;
}
}
loop++;
#ifdef AUDIOWARNINGS
dout << "loop" << loop;
#endif
} else {
frame->pts = c->frame->pts;
break;
}
} while (true);
} else {
// if there is no more data in the file, we flush the remainder out of swresample
break;
}
new_frame = true;
if (c->frame_sample_index < 0) {
c->frame_sample_index = 0;
} else {
c->frame_sample_index -= nb_bytes;
}
nb_bytes = frame->nb_samples * av_get_bytes_per_sample(static_cast<AVSampleFormat>(frame->format)) * frame->channels;
if (c->audio_just_reset) {
// get precise sample offset for the elected clip_in from this audio frame
double target_sts = playhead_to_clip_seconds(c, c->audio_target_frame);
double frame_sts = ((frame->pts - c->stream->start_time) * timebase);
int nb_samples = qRound64((target_sts - frame_sts)*audio_output->format().sampleRate());
c->frame_sample_index = nb_samples * 4;
#ifdef AUDIOWARNINGS
dout << "fsts:" << frame_sts << "tsts:" << target_sts << "nbs:" << nb_samples << "nbb:" << nb_bytes << "rev_targetToSec:" << (c->reverse_target * timebase);
dout << "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
dout << "fsi-post-post:" << c->frame_sample_index;
#endif
if (c->audio_buffer_write == 0) {
c->audio_buffer_write = get_buffer_offset_from_frame(last_fr, qMax(timeline_in, target_frame));
if (frame_skip > 0) {
int target = get_buffer_offset_from_frame(last_fr, qMax(timeline_in + frame_skip, target_frame));
c->frame_sample_index += (target - c->audio_buffer_write);
c->audio_buffer_write = target;
}
}
int offset = audio_ibuffer_read - c->audio_buffer_write;
if (offset > 0) {
c->audio_buffer_write += offset;
c->frame_sample_index += offset;
}
// try to correct negative fsi
if (c->frame_sample_index < 0) {
c->audio_buffer_write -= c->frame_sample_index;
c->frame_sample_index = 0;
}
}
if (c->reverse) frame = c->queue.at(1);
#ifdef AUDIOWARNINGS
dout << "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;
if (new_frame) {
apply_audio_effects(c, bytes_to_seconds(c->audio_buffer_write, 2, audio_output->format().sampleRate()) + audio_ibuffer_timecode + ((double)c->clip_in/c->sequence->frame_rate) - ((double)timeline_in/last_fr), frame, nb_bytes, nests);
}
}
break;
case MEDIA_TYPE_TONE:
frame = c->frame;
nb_bytes = frame->nb_samples * av_get_bytes_per_sample(static_cast<AVSampleFormat>(frame->format)) * frame->channels;
while ((c->frame_sample_index == -1 || c->frame_sample_index >= nb_bytes) && nb_bytes > 0) {
// 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, nests);
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(last_fr, qMax(timeline_in, 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
dout << "[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(c->sequence->frame_rate, timeline_out);
audio_write_lock.lock();
while (c->frame_sample_index < nb_bytes
&& c->audio_buffer_write < audio_ibuffer_read+(audio_ibuffer_size>>1)
&& 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;
}
#ifdef AUDIOWARNINGS
if (c->audio_buffer_write >= buffer_timeline_out) dout << "timeline out at fsi" << c->frame_sample_index << "of frame ts" << c->frame->pts;
#endif
audio_write_lock.unlock();
if (scrubbing) {
audio_thread->notifyReceiver();
}
if (c->frame_sample_index == nb_bytes) {
c->frame_sample_index = -1;
} else {
// assume we have no more data to send
break;
}
// dout << "ended" << c->frame_sample_index << nb_bytes;
}
if (c->reached_end) {
frame->nb_samples = 0;
}
if (scrubbing) {
break;
}
}
QMetaObject::invokeMethod(panel_footage_viewer, "play_wake", Qt::QueuedConnection);
QMetaObject::invokeMethod(panel_sequence_viewer, "play_wake", Qt::QueuedConnection);
}
void cache_video_worker(Clip* c, long playhead) {
int read_ret, send_ret, retr_ret;
int64_t target_pts = seconds_to_timestamp(c, playhead_to_clip_seconds(c, playhead));
int limit = c->max_queue_size;
if (c->ignore_reverse) {
// waiting for one frame
limit = c->queue.size() + 1;
} else if (c->reverse) {
limit *= 2;
}
if (c->queue.size() < limit) {
bool reverse = (c->reverse && !c->ignore_reverse);
c->ignore_reverse = false;
int64_t eighth_second = av_q2d(av_inv_q(c->stream->time_base))*0.125;
int64_t smallest_pts = INT64_MAX;
if (reverse && c->queue.size() > 0) {
int64_t quarter_sec = qRound64(av_q2d(av_inv_q(c->stream->time_base))) >> 2;
for (int i=0;i<c->queue.size();i++) {
smallest_pts = qMin(smallest_pts, c->queue.at(i)->pts);
}
avcodec_flush_buffers(c->codecCtx);
c->reached_end = false;
int64_t seek_ts = qMax(static_cast<int64_t>(0), smallest_pts - quarter_sec);
av_seek_frame(c->formatCtx, c->stream->index, seek_ts, AVSEEK_FLAG_BACKWARD);
} else {
smallest_pts = target_pts;
}
while (true) {
AVFrame* frame = av_frame_alloc();
Media* media = static_cast<Media*>(c->media);
MediaStream* ms = media->get_stream_from_file_index(true, c->media_stream);
while ((retr_ret = av_buffersink_get_frame(c->buffersink_ctx, frame)) == AVERROR(EAGAIN)) {
AVFrame* send_frame = c->frame;
read_ret = (c->use_existing_frame) ? 0 : retrieve_next_frame(c, send_frame);
c->use_existing_frame = false;
if (read_ret >= 0) {
bool send_it = false;
if (reverse) {
send_it = true;
} else if (send_frame->pts > target_pts - eighth_second) {
send_it = true;
} else if (media->get_stream_from_file_index(true, c->media_stream)->infinite_length) {
send_it = true;
} else {
dout << "skipped adding a frame to the queue - fpts:" << send_frame->pts << "target:" << target_pts;
}
if (send_it) {
if ((send_ret = av_buffersrc_add_frame_flags(c->buffersrc_ctx, send_frame, AV_BUFFERSRC_FLAG_KEEP_REF)) < 0) {
dout << "[ERROR] Failed to add frame to buffer source." << send_ret;
break;
}
}
av_frame_unref(c->frame);
} else {
if (read_ret == AVERROR_EOF) {
c->reached_end = true;
} else {
dout << "[ERROR] Failed to read frame." << read_ret;
}
break;
}
}
if (retr_ret < 0) {
if (retr_ret == AVERROR_EOF) {
c->reached_end = true;
} else {
dout << "[ERROR] Failed to retrieve frame from buffersink." << retr_ret;
}
av_frame_free(&frame);
break;
} else {
if (reverse && ((smallest_pts == target_pts && frame->pts >= smallest_pts) || (smallest_pts != target_pts && frame->pts > smallest_pts))) {
av_frame_free(&frame);
break;
} else {
// thread-safety while adding frame to the queue
c->queue_lock.lock();
c->queue.append(frame);
if (!ms->infinite_length && !reverse && c->queue.size() == limit) {
// see if we got the frame we needed (used for speed ups primarily)
bool found = false;
for (int i=0;i<c->queue.size();i++) {
// TODO/NOTE: this will not work on clips that are sped up AND reversed
if (c->queue.at(i)->pts >= target_pts) {
found = true;
break;
}
}
if (found) {
c->queue_lock.unlock();
break;
} else {
// remove earliest frame and loop to store another
c->queue_remove_earliest();
}
}
c->queue_lock.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) {
/*avcodec_flush_buffers(c->codecCtx);
av_seek_frame(c->formatCtx, ms->file_index, 0, AVSEEK_FLAG_BACKWARD);*/
c->use_existing_frame = false;
} else {
if (c->stream->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
// clear current queue
c->queue_clear();
// seeks to nearest keyframe (target_frame represents internal clip frame)
int64_t target_ts = seconds_to_timestamp(c, playhead_to_clip_seconds(c, target_frame));
int64_t seek_ts = target_ts;
int64_t timebase_half_second = qRound64(av_q2d(av_inv_q(c->stream->time_base)));
if (c->reverse) seek_ts -= timebase_half_second;
while (true) {
// flush ffmpeg codecs
avcodec_flush_buffers(c->codecCtx);
c->reached_end = false;
if (seek_ts > 0) {
av_seek_frame(c->formatCtx, ms->file_index, seek_ts, AVSEEK_FLAG_BACKWARD);
av_frame_unref(c->frame);
int ret = retrieve_next_frame(c, c->frame);
if (ret < 0) {
dout << "[WARNING] Seeking terminated prematurely";
break;
}
if (c->frame->pts <= target_ts) {
c->use_existing_frame = true;
break;
} else {
seek_ts -= timebase_half_second;
}
} else {
av_frame_unref(c->frame);
av_seek_frame(c->formatCtx, ms->file_index, 0, AVSEEK_FLAG_BACKWARD);
c->use_existing_frame = false;
break;
}
}
} else if (c->stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
// flush ffmpeg codecs
avcodec_flush_buffers(c->codecCtx);
c->reached_end = false;
// seek (target_frame represents timeline timecode in frames, not clip timecode)
int64_t timestamp = seconds_to_timestamp(c, playhead_to_clip_seconds(c, target_frame));
if (c->reverse) {
c->reverse_target = timestamp;
timestamp -= av_q2d(av_inv_q(c->stream->time_base));
#ifdef AUDIOWARNINGS
dout << "seeking to" << timestamp << "(originally" << c->reverse_target << ")";
} else {
dout << "reset called; seeking to" << timestamp;
#endif
}
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->reached_end = false;
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);
dout << "[ERROR] Could not open" << filename << "-" << err;
return;
}
errCode = avformat_find_stream_info(clip->formatCtx, NULL);
if (errCode < 0) {
char err[1024];
av_strerror(errCode, err, 1024);
dout << "[ERROR] Could not open" << filename << "-" << err;
return;
}
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);
clip->max_queue_size = (ms->infinite_length) ? 1 : qCeil(ms->video_frame_rate*0.5);
if (ms->video_interlacing != VIDEO_PROGRESSIVE) clip->max_queue_size *= 2;
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) {
dout << "[ERROR] Could not open codec";
}
// allocate filtergraph
clip->filter_graph = avfilter_graph_alloc();
if (clip->filter_graph == NULL) {
dout << "[ERROR] Could not create filtergraph";
}
char filter_args[512];
if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
/* SKIP_TYPE_SEEK is used if a video is playing at a speed so fast
* that it is quicker to seek to the next frame than to just play
* up to it (e.g. 2000% speed would require playing and skipping
* 20 frames per frame and it many cases it would be quicker to
* seek to it and cache in memory instead.
*
* TODO there could probably be a better heuristic than
* (speed >= 5) for using seek mode. Experiment with the value
* but also in the future perhaps we could implement a system
* of testing how long it takes to seek vs how long it takes to
* decode a frame and compare them to choose with method.
*/
clip->skip_type = (clip->speed < 5) ? SKIP_TYPE_DISCARD : SKIP_TYPE_SEEK;
// create memory cache for video (deprecated)
// clip->cache_size = (ms->infinite_length) ? 1 : ceil(av_q2d(clip->stream->avg_frame_rate)/4); // cache is half a second in total
// if (clip->skip_type == SKIP_TYPE_SEEK) clip->cache_size *= 2;
// if (ms->video_interlacing != VIDEO_PROGRESSIVE) clip->cache_size *= 2;
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) {
dout << "[ERROR] Could not set output pixel format";
}
AVFilterContext* last_filter = clip->buffersrc_ctx;
if (ms->video_interlacing != VIDEO_PROGRESSIVE) {
AVFilterContext* yadif_filter;
char yadif_args[100];
snprintf(yadif_args, sizeof(yadif_args), "mode=3:parity=%d", ((ms->video_interlacing == VIDEO_TOP_FIELD_FIRST) ? 0 : 1)); // try mode 1
avfilter_graph_create_filter(&yadif_filter, avfilter_get_by_name("yadif"), "yadif", yadif_args, NULL, clip->filter_graph);
avfilter_link(last_filter, 0, yadif_filter, 0);
last_filter = yadif_filter;
}
/* stabilization code one day
if (false) {
AVFilterContext* stab_filter;
int stab_ret = avfilter_graph_create_filter(&stab_filter, avfilter_get_by_name("vidstabtransform"), "vidstab", "input=C\\:/Users/Matt/Desktop/samples/transforms.trf", NULL, clip->filter_graph);
if (stab_ret < 0) {
char err[100];
av_strerror(stab_ret, err, sizeof(err));
dout << "stab ret:" << stab_ret << err;
} else {
dout << "link 1:" << avfilter_link(last_filter, 0, stab_filter, 0);
last_filter = stab_filter;
}
}
*/
avfilter_link(last_filter, 0, clip->buffersink_ctx, 0);
avfilter_graph_config(clip->filter_graph, NULL);
} else if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
if (clip->codecCtx->channel_layout == 0) clip->codecCtx->channel_layout = av_get_default_channel_layout(clip->stream->codecpar->channels);
// set up cache
clip->queue.append(av_frame_alloc());
if (clip->reverse) {
AVFrame* reverse_frame = av_frame_alloc();
reverse_frame->format = sample_format;
reverse_frame->nb_samples = audio_output->format().sampleRate()*2;
reverse_frame->channel_layout = clip->sequence->audio_layout;
reverse_frame->channels = av_get_channel_layout_nb_channels(clip->sequence->audio_layout);
av_frame_get_buffer(reverse_frame, 0);
clip->queue.append(reverse_frame);
}
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) {
dout << "[ERROR] Could not set output sample format";
}
int64_t channel_layouts[] = { AV_CH_LAYOUT_STEREO, static_cast<AVSampleFormat>(-1) };
if (av_opt_set_int_list(clip->buffersink_ctx, "channel_layouts", channel_layouts, -1, AV_OPT_SEARCH_CHILDREN) < 0) {
dout << "[ERROR] Could not set output sample format";
}
int target_sample_rate = audio_output->format().sampleRate();
if (qFuzzyCompare(clip->speed, 1.0)) {
avfilter_link(clip->buffersrc_ctx, 0, clip->buffersink_ctx, 0);
} else if (clip->maintain_audio_pitch) {
AVFilterContext* previous_filter = clip->buffersrc_ctx;
AVFilterContext* last_filter = clip->buffersrc_ctx;
char speed_param[10];
if (clip->speed != 1.0) {
double base = (clip->speed > 1.0) ? 2.0 : 0.5;
double speedlog = log(clip->speed) / log(base);
int whole2 = qFloor(speedlog);
speedlog -= whole2;
if (whole2 > 0) {
snprintf(speed_param, sizeof(speed_param), "%f", base);
for (int i=0;i<whole2;i++) {
AVFilterContext* tempo_filter = NULL;
avfilter_graph_create_filter(&tempo_filter, avfilter_get_by_name("atempo"), "atempo", speed_param, NULL, clip->filter_graph);
avfilter_link(previous_filter, 0, tempo_filter, 0);
previous_filter = tempo_filter;
}
}
snprintf(speed_param, sizeof(speed_param), "%f", qPow(base, speedlog));
last_filter = NULL;
avfilter_graph_create_filter(&last_filter, avfilter_get_by_name("atempo"), "atempo", speed_param, NULL, clip->filter_graph);
avfilter_link(previous_filter, 0, last_filter, 0);
}
avfilter_link(last_filter, 0, clip->buffersink_ctx, 0);
} else {
target_sample_rate = qRound64(target_sample_rate / 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) {
dout << "[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 = clip->sequence->audio_layout;
clip->frame->channels = av_get_channel_layout_nb_channels(clip->frame->channel_layout);
clip->frame->sample_rate = audio_output->format().sampleRate();
clip->frame->nb_samples = 2048;
av_frame_make_writable(clip->frame);
if (av_frame_get_buffer(clip->frame, 0)) {
dout << "[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;
dout << "[INFO] Clip opened on track" << clip->track;
}
void cache_clip_worker(Clip* clip, long playhead, bool reset, bool scrubbing, QVector<Clip*> nests) {
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) {
cache_video_worker(clip, playhead);
} else if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
cache_audio_worker(clip, scrubbing, nests);
}
break;
case MEDIA_TYPE_TONE:
cache_audio_worker(clip, scrubbing, nests);
break;
}
}
void close_clip_worker(Clip* clip) {
clip->finished_opening = false;
if (clip->media_type == MEDIA_TYPE_FOOTAGE) {
clip->queue_clear();
avfilter_graph_free(&clip->filter_graph);
avcodec_close(clip->codecCtx);
avcodec_free_context(&clip->codecCtx);
avformat_close_input(&clip->formatCtx);
}
av_frame_free(&clip->frame);
clip->reset();
dout << "[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, reset, scrubbing, nests);
}
}
close_clip_worker(clip);
clip->lock.unlock();
clip->open_lock.unlock();
}
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