464 lines
17 KiB
C++
464 lines
17 KiB
C++
#include "cacher.h"
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#include "project/clip.h"
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#include "project/sequence.h"
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#include "io/media.h"
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#include "playback/audio.h"
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#include "playback/playback.h"
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#include "effects/effect.h"
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#include "panels/timeline.h"
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#include "panels/project.h"
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#include "effects/transition.h"
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extern "C" {
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#include <libavformat/avformat.h>
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#include <libavcodec/avcodec.h>
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#include <libswscale/swscale.h>
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#include <libswresample/swresample.h>
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}
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#include <QDebug>
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#include <QtMath>
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#include <math.h>
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int dest_format = AV_PIX_FMT_RGBA;
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void apply_audio_effects(Clip* c, AVFrame* frame, int nb_bytes) {
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// perform all audio effects
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for (int j=0;j<c->effects.size();j++) {
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Effect* e = c->effects.at(j);
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if (e->is_enabled()) e->process_audio(frame->data[0], nb_bytes);
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}
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if (c->opening_transition != NULL) {
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if (c->media_type == MEDIA_TYPE_FOOTAGE) {
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double transition_start = (c->clip_in / c->sequence->frame_rate);
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double transition_end = transition_start + (c->opening_transition->length / c->sequence->frame_rate);
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double range_start = (frame->pts * av_q2d(c->stream->time_base));
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double range_end = range_start + ((double) nb_bytes * 0.5 / frame->channels / frame->sample_rate);
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if (range_end < transition_end) {
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double adjustment = transition_end - transition_start;
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double adjusted_range_start = (range_start - transition_start) / adjustment;
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double adjusted_range_end = (range_end - transition_start) / adjustment;
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c->opening_transition->process_audio(adjusted_range_start, adjusted_range_end, frame->data[0], nb_bytes, false);
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}
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}
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}
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if (c->closing_transition != NULL) {
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if (c->media_type == MEDIA_TYPE_FOOTAGE) {
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double transition_end = (c->clip_in + c->getLength()) / c->sequence->frame_rate;
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double transition_start = transition_end - (c->closing_transition->length / c->sequence->frame_rate);
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double range_start = (frame->pts * av_q2d(c->stream->time_base));
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double range_end = range_start + ((double) nb_bytes * 0.5 / frame->channels / frame->sample_rate);
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if (range_start > transition_start) {
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double adjustment = transition_end - transition_start;
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double adjusted_range_start = (range_start - transition_start) / adjustment;
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double adjusted_range_end = (range_end - transition_start) / adjustment;
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c->closing_transition->process_audio(adjusted_range_start, adjusted_range_end, frame->data[0], nb_bytes, true);
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}
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}
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}
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}
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void cache_audio_worker(Clip* c, Clip* nest) {
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int written = 0;
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int max_write = 16384;
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long timeline_in = c->timeline_in;
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long timeline_out = c->timeline_out;
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if (nest != NULL) {
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timeline_in = refactor_frame_number(timeline_in, c->sequence->frame_rate, sequence->frame_rate) + nest->timeline_in;
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timeline_out = refactor_frame_number(timeline_out, c->sequence->frame_rate, sequence->frame_rate) + nest->timeline_in;
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}
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//
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// (c->clip_in / c->sequence->frame_rate) is the starting point in seconds
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// we need to convert it to bytes
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//
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//
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while (written < max_write) {
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// gets one frame worth of audio and sends it to the audio buffer
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AVFrame* frame = c->cache_A.frames[0];
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if (c->need_new_audio_frame) {
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// no more audio left in frame, get a new one
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if (!c->reached_end) {
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retrieve_next_frame_raw_data(c, frame);
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int byte_count = av_samples_get_buffer_size(NULL, frame->channels, frame->nb_samples, static_cast<AVSampleFormat>(frame->format), 1);
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apply_audio_effects(c, frame, byte_count);
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if (nest != NULL) apply_audio_effects(nest, frame, byte_count);
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} else {
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// set by retrieve_next_frame_raw_data indicating no more frames in file,
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// but there still may be samples in swresample
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swr_convert_frame(c->swr_ctx, frame, NULL);
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}
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c->need_new_audio_frame = false;
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if (c->audio_just_reset) {
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// get precise sample offset for the elected clip_in from this audio frame
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double target_sts = 0;
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if (c->audio_target_frame < timeline_in) {
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target_sts = clip_frame_to_seconds(c, c->clip_in);
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} else {
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target_sts = playhead_to_seconds(c, c->audio_target_frame);
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}
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double frame_sts = (frame->pts * av_q2d(c->stream->time_base));
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int nb_samples = qRound((target_sts - frame_sts)*c->sequence->audio_frequency);
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if (nb_samples == 0) {
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c->frame_sample_index = 0;
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} else {
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c->frame_sample_index = av_samples_get_buffer_size(NULL, av_get_channel_layout_nb_channels(c->sequence->audio_layout), nb_samples, AV_SAMPLE_FMT_S16, 1);
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}
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c->audio_just_reset = false;
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} else {
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c->frame_sample_index = 0;
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}
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}
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if (frame->nb_samples == 0) {
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written = max_write;
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} else {
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int nb_bytes = av_samples_get_buffer_size(NULL, frame->channels, frame->nb_samples, static_cast<AVSampleFormat>(frame->format), 1);
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int half_buffer = (audio_ibuffer_size/2);
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if (c->audio_buffer_write == 0) {
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c->audio_buffer_write = get_buffer_offset_from_frame(qMax(timeline_in, c->audio_target_frame));
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int offset = audio_ibuffer_read - c->audio_buffer_write;
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if (offset > 0) {
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c->audio_buffer_write += offset;
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c->frame_sample_index += offset;
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}
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}
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bool apply_effects = false;
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while (c->frame_sample_index > nb_bytes) {
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// get new frame
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retrieve_next_frame_raw_data(c, frame);
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apply_effects = true;
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nb_bytes = av_samples_get_buffer_size(NULL, frame->channels, frame->nb_samples, static_cast<AVSampleFormat>(frame->format), 1);
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c->frame_sample_index -= nb_bytes;
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// code DISGUSTINGLY copy/pasted from above
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int offset = audio_ibuffer_read - c->audio_buffer_write;
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if (offset > 0) {
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c->audio_buffer_write += offset;
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c->frame_sample_index += offset;
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}
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}
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if (apply_effects) {
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apply_audio_effects(c, frame, nb_bytes);
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}
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while (c->frame_sample_index < nb_bytes) {
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if (c->audio_buffer_write >= audio_ibuffer_read+half_buffer || c->audio_buffer_write >= get_buffer_offset_from_frame(timeline_out)) {
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written = max_write;
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break;
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} else {
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int upper_byte_index = (c->audio_buffer_write+1)%audio_ibuffer_size;
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int lower_byte_index = (c->audio_buffer_write)%audio_ibuffer_size;
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qint16 old_sample = (qint16) ((audio_ibuffer[upper_byte_index] & 0xFF) << 8 | (audio_ibuffer[lower_byte_index] & 0xFF));
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qint16 new_sample = (qint16) ((frame->data[0][c->frame_sample_index+1] & 0xFF) << 8 | (frame->data[0][c->frame_sample_index] & 0xFF));
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qint32 mixed_sample = old_sample + new_sample;
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mixed_sample = qMax(qMin(mixed_sample, (qint32)INT16_MAX), (qint32)INT16_MIN);
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audio_ibuffer[upper_byte_index] = (quint8) (mixed_sample >> 8);
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audio_ibuffer[lower_byte_index] = (quint8) mixed_sample;
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c->audio_buffer_write+=2;
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c->frame_sample_index+=2;
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written+=2;
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}
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}
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if (c->frame_sample_index == nb_bytes) {
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c->need_new_audio_frame = true;
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}
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}
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}
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}
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void cache_video_worker(Clip* c, long playhead, ClipCache* cache) {
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cache->mutex.lock();
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cache->offset = playhead;
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bool error = false;
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int i = 0;
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if (!c->reached_end) {
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while (i < c->cache_size) {
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retrieve_next_frame_raw_data(c, cache->frames[i]);
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if (c->reached_end) break;
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i++;
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}
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}
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cache->write_count = i;
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if (!error) {
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// setting the cache to written even if it hasn't reached_end prevents playback from
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// signaling a seek/reset because it wasn't able to find the frame
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cache->written = true;
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cache->unread = true;
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}
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cache->mutex.unlock();
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}
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void reset_cache(Clip* c, long target_frame) {
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// if we seek to a whole other place in the timeline, we'll need to reset the cache with new values
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MediaStream* ms = static_cast<Media*>(c->media)->get_stream_from_file_index(c->media_stream);
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if (!ms->infinite_length) {
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// flush ffmpeg codecs
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avcodec_flush_buffers(c->codecCtx);
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double timebase = av_q2d(c->stream->time_base);
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if (c->stream->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
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// seeks to nearest keyframe (target_frame represents internal clip frame)
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av_seek_frame(c->formatCtx, ms->file_index, (int64_t) qFloor(clip_frame_to_seconds(c, target_frame) / timebase), AVSEEK_FLAG_BACKWARD);
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// play up to the frame we actually want
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long retrieved_frame = 0;
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AVFrame* temp = av_frame_alloc();
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do {
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retrieve_next_frame(c, temp);
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if (retrieved_frame == 0) {
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if (target_frame != 0) retrieved_frame = floor(temp->pts * timebase * av_q2d(av_guess_frame_rate(c->formatCtx, c->stream, temp)));
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} else {
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retrieved_frame++;
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}
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} while (retrieved_frame < target_frame);
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av_frame_free(&temp);
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} else if (c->stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
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// seek (target_frame represents timeline timecode in frames, not clip timecode)
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swr_drop_output(c->swr_ctx, swr_get_out_samples(c->swr_ctx, 0));
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av_seek_frame(c->formatCtx, ms->file_index, playhead_to_seconds(c, target_frame) / timebase, AVSEEK_FLAG_BACKWARD);
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c->audio_target_frame = target_frame;
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c->need_new_audio_frame = true;
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c->audio_just_reset = true;
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}
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}
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}
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Cacher::Cacher(Clip* c) : clip(c) {}
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void open_clip_worker(Clip* clip) {
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// opens file resource for FFmpeg and prepares Clip struct for playback
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Media* m = static_cast<Media*>(clip->media);
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QByteArray ba = m->url.toUtf8();
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const char* filename = ba.constData();
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MediaStream* ms = m->get_stream_from_file_index(clip->media_stream);
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int errCode = avformat_open_input(
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&clip->formatCtx,
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filename,
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NULL,
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NULL
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);
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if (errCode != 0) {
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char err[1024];
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av_strerror(errCode, err, 1024);
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qDebug() << "[ERROR] Could not open" << filename << "-" << err;
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}
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errCode = avformat_find_stream_info(clip->formatCtx, NULL);
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if (errCode < 0) {
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char err[1024];
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av_strerror(errCode, err, 1024);
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qDebug() << "[ERROR] Could not open" << filename << "-" << err;
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}
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av_dump_format(clip->formatCtx, 0, filename, 0);
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clip->stream = clip->formatCtx->streams[ms->file_index];
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clip->codec = avcodec_find_decoder(clip->stream->codecpar->codec_id);
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clip->codecCtx = avcodec_alloc_context3(clip->codec);
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avcodec_parameters_to_context(clip->codecCtx, clip->stream->codecpar);
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AVDictionary* opts = NULL;
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// decoding optimization configuration
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if (clip->stream->codecpar->codec_id != AV_CODEC_ID_PNG &&
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clip->stream->codecpar->codec_id != AV_CODEC_ID_APNG &&
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clip->stream->codecpar->codec_id != AV_CODEC_ID_TIFF &&
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clip->stream->codecpar->codec_id != AV_CODEC_ID_PSD) {
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av_dict_set(&opts, "threads", "auto", 0);
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}
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if (clip->stream->codecpar->codec_id == AV_CODEC_ID_H264) {
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av_dict_set(&opts, "tune", "fastdecode", 0);
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av_dict_set(&opts, "tune", "zerolatency", 0);
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}
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// Open codec
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if (avcodec_open2(clip->codecCtx, clip->codec, &opts) < 0) {
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qDebug() << "[ERROR] Could not open codec";
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}
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if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
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// set up swscale context - primarily used for colorspace conversion
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// as "scaling" is actually done by OpenGL
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int dstW = ceil(clip->stream->codecpar->width/2)*2;
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int dstH = ceil(clip->stream->codecpar->height/2)*2;
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clip->sws_ctx = sws_getContext(
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clip->stream->codecpar->width,
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clip->stream->codecpar->height,
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static_cast<AVPixelFormat>(clip->stream->codecpar->format),
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dstW,
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dstH,
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static_cast<AVPixelFormat>(dest_format),
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SWS_FAST_BILINEAR,
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NULL,
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NULL,
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NULL
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);
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// create memory cache for video
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clip->cache_size = (ms->infinite_length) ? 1 : ceil(av_q2d(av_guess_frame_rate(clip->formatCtx, clip->stream, NULL))/4); // cache is half a second in total
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// infinite length doesn't need cache B
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clip->cache_A.frames = new AVFrame* [clip->cache_size];
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clip->cache_B.frames = new AVFrame* [clip->cache_size];
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for (int i=0;i<clip->cache_size;i++) {
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clip->cache_A.frames[i] = av_frame_alloc();
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av_frame_make_writable(clip->cache_A.frames[i]);
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clip->cache_A.frames[i]->width = dstW;
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clip->cache_A.frames[i]->height = dstH;
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clip->cache_A.frames[i]->format = dest_format;
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if (av_frame_get_buffer(clip->cache_A.frames[i], 0)) {
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qDebug() << "[ERROR] Could not allocate buffer for sws_frame";
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}
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clip->cache_A.frames[i]->linesize[0] = dstW*4;
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clip->cache_B.frames[i] = av_frame_alloc();
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av_frame_make_writable(clip->cache_B.frames[i]);
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clip->cache_B.frames[i]->width = dstW;
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clip->cache_B.frames[i]->height = dstH;
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clip->cache_B.frames[i]->format = dest_format;
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if (av_frame_get_buffer(clip->cache_B.frames[i], 0)) {
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qDebug() << "[ERROR] Could not allocate buffer for sws_frame";
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}
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clip->cache_B.frames[i]->linesize[0] = dstW*4;
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}
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clip->comp_frame_size = clip->stream->codecpar->width * clip->stream->codecpar->height * 4;
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clip->comp_frame = new uchar[clip->comp_frame_size];
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} else if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
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// if FFmpeg can't pick up the channel layout (usually WAV), assume
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// based on channel count (doesn't support surround sound sources yet)
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if (clip->codecCtx->channel_layout == 0) {
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clip->codecCtx->channel_layout = guess_layout_from_channels(clip->stream->codecpar->channels);
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}
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int sample_format = AV_SAMPLE_FMT_S16;
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// init resampling context
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clip->swr_ctx = swr_alloc_set_opts(
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NULL,
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clip->sequence->audio_layout,
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static_cast<AVSampleFormat>(sample_format),
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clip->sequence->audio_frequency,
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clip->codecCtx->channel_layout,
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static_cast<AVSampleFormat>(clip->stream->codecpar->format),
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clip->stream->codecpar->sample_rate,
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0,
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NULL
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);
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swr_init(clip->swr_ctx);
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// set up cache
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clip->cache_A.frames = new AVFrame* [1];
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clip->cache_A.frames[0] = av_frame_alloc();
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clip->cache_A.frames[0]->format = sample_format;
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clip->cache_A.frames[0]->channel_layout = clip->sequence->audio_layout;
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clip->cache_A.frames[0]->channels = av_get_channel_layout_nb_channels(clip->cache_A.frames[0]->channel_layout);
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clip->cache_A.frames[0]->sample_rate = clip->sequence->audio_frequency;
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av_frame_make_writable(clip->cache_A.frames[0]);
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clip->audio_reset = true;
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}
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clip->frame = av_frame_alloc();
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clip->finished_opening = true;
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qDebug() << "[INFO] Clip opened on track" << clip->track;
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}
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void cache_clip_worker(Clip* clip, long playhead, bool write_A, bool write_B, bool reset, Clip* nest) {
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if (reset) {
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// note: for video, playhead is in "internal clip" frames - for audio, it's the timeline playhead
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reset_cache(clip, playhead);
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clip->audio_reset = false;
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}
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if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
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if (write_A) {
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cache_video_worker(clip, playhead, &clip->cache_A);
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playhead += clip->cache_size;
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}
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if (write_B) {
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cache_video_worker(clip, playhead, &clip->cache_B);
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}
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} else if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
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cache_audio_worker(clip, nest);
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}
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}
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void close_clip_worker(Clip* clip) {
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// closes ffmpeg file handle and frees any memory used for caching
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MediaStream* ms = static_cast<Media*>(clip->media)->get_stream_from_file_index(clip->media_stream);
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if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
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// sws_freeContext(clip->sws_ctx);
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// TODO will eventually be in audio too
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delete [] clip->comp_frame;
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} else if (clip->stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
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swr_free(&clip->swr_ctx);
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}
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avcodec_close(clip->codecCtx);
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avcodec_free_context(&clip->codecCtx);
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avformat_close_input(&clip->formatCtx);
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for (int i=0;i<clip->cache_size;i++) {
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av_frame_free(&clip->cache_A.frames[i]);
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if (!ms->infinite_length) av_frame_free(&clip->cache_B.frames[i]);
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}
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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();
|
|
}
|