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even more improved corner pin and more consistent audio

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This commit is contained in:
itsmattkc
2018-11-12 20:49:15 +11:00
parent 18c112d83e
commit 4f3c807c6a
5 changed files with 45 additions and 31 deletions
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playback/audio.cpp
+5 -7
View File
@@ -68,9 +68,7 @@ void init_audio(Sequence* s) {
audio_thread->start(QThread::TimeCriticalPriority);
clear_audio_ibuffer();
}
dout << "h";
}
}
void stop_audio() {
@@ -88,9 +86,9 @@ void clear_audio_ibuffer() {
audio_ibuffer_read = 0;
}
int get_buffer_offset_from_frame(Sequence* s, long frame) {
int get_buffer_offset_from_frame(double framerate, long frame) {
if (frame >= audio_ibuffer_frame) {
return qFloor(((double) (frame - audio_ibuffer_frame)/s->frame_rate)*s->audio_frequency)*av_get_bytes_per_sample(AV_SAMPLE_FMT_S16)*av_get_channel_layout_nb_channels(s->audio_layout);
return qFloor(((double) (frame - audio_ibuffer_frame)/framerate)*audio_output->format().sampleRate())*av_get_bytes_per_sample(AV_SAMPLE_FMT_S16)*av_get_channel_layout_nb_channels(AV_CH_LAYOUT_STEREO);
} else {
dout << "[WARNING] Invalid values passed to get_buffer_offset_from_frame";
return 0;
@@ -160,14 +158,14 @@ int AudioSenderThread::send_audio_to_output(int offset, int max) {
int channel_count = av_get_channel_layout_nb_channels(s->audio_layout);
long sample_cache_playhead = panel_timeline->ui->audio_monitor->sample_cache_offset + (panel_timeline->ui->audio_monitor->sample_cache.size()/channel_count);
int next_buffer_offset, buffer_offset_adjusted, i;
int buffer_offset = get_buffer_offset_from_frame(s, sample_cache_playhead);
int buffer_offset = get_buffer_offset_from_frame(s->frame_rate, sample_cache_playhead);
if (samples.size() != channel_count) samples.resize(channel_count);
samples.fill(0);
// TODO: I don't like this, but i'm not sure if there's a smarter way to do it
while (buffer_offset < audio_ibuffer_limit) {
sample_cache_playhead++;
next_buffer_offset = qMin(get_buffer_offset_from_frame(s, sample_cache_playhead), audio_ibuffer_limit);
next_buffer_offset = qMin(get_buffer_offset_from_frame(s->frame_rate, sample_cache_playhead), audio_ibuffer_limit);
while (buffer_offset < next_buffer_offset) {
for (i=0;i<samples.size();i++) {
buffer_offset_adjusted = buffer_offset%audio_ibuffer_size;
playback/audio.h
+1 -1
View File
@@ -45,7 +45,7 @@ void clear_audio_ibuffer();
void init_audio(Sequence *s);
void stop_audio();
int get_buffer_offset_from_frame(Sequence* s, long frame);
int get_buffer_offset_from_frame(double framerate, long frame);
bool start_recording();
void stop_recording();
playback/cacher.cpp
+12 -12
View File
@@ -86,8 +86,8 @@ void cache_audio_worker(Clip* c, bool scrubbing, QVector<Clip*>& nests) {
long timeline_out = c->timeline_out;
long frame_skip = 0;
double last_fr = c->sequence->frame_rate;
if (!nests.isEmpty()) {
double last_fr = c->sequence->frame_rate;
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;
@@ -213,10 +213,10 @@ void cache_audio_worker(Clip* c, bool scrubbing, QVector<Clip*>& nests) {
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) * c->sequence->audio_frequency);
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:" << c->sequence->audio_frequency << ")";
dout << "cut off" << cutoff << "samples (rate:" << audio_output->format().sampleRate() << ")";
#endif
rev_frame->nb_samples -= cutoff;
}
@@ -225,7 +225,7 @@ void cache_audio_worker(Clip* c, bool scrubbing, QVector<Clip*>& nests) {
#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 * (c->sequence->audio_frequency / c->speed));
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
@@ -283,7 +283,7 @@ void cache_audio_worker(Clip* c, bool scrubbing, QVector<Clip*>& nests) {
// 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)*c->sequence->audio_frequency);
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);
@@ -297,10 +297,10 @@ void cache_audio_worker(Clip* c, bool scrubbing, QVector<Clip*>& nests) {
dout << "fsi-post-post:" << c->frame_sample_index;
#endif
if (c->audio_buffer_write == 0) {
c->audio_buffer_write = get_buffer_offset_from_frame((sequence == NULL) ? c->sequence : sequence, qMax(timeline_in, c->audio_target_frame));
c->audio_buffer_write = get_buffer_offset_from_frame(c->sequence->frame_rate, qMax(timeline_in, c->audio_target_frame));
if (frame_skip > 0) {
int target = get_buffer_offset_from_frame((sequence == NULL) ? c->sequence : sequence, qMax(timeline_in + frame_skip, c->audio_target_frame));
int target = get_buffer_offset_from_frame(c->sequence->frame_rate, qMax(timeline_in + frame_skip, c->audio_target_frame));
c->frame_sample_index += (target - c->audio_buffer_write);
c->audio_buffer_write = target;
}
@@ -328,7 +328,7 @@ void cache_audio_worker(Clip* c, bool scrubbing, QVector<Clip*>& nests) {
// 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, c->sequence->audio_frequency) + audio_ibuffer_timecode + ((double)c->clip_in/c->sequence->frame_rate) - ((double)timeline_in/((sequence == NULL) ? c->sequence->frame_rate : sequence->frame_rate)), frame, nb_bytes, nests);
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;
@@ -342,7 +342,7 @@ void cache_audio_worker(Clip* c, bool scrubbing, QVector<Clip*>& 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(sequence, qMax(timeline_in, c->audio_target_frame));
c->audio_buffer_write = get_buffer_offset_from_frame(c->sequence->frame_rate, qMax(timeline_in, c->audio_target_frame));
}
int offset = audio_ibuffer_read - c->audio_buffer_write;
if (offset > 0) {
@@ -360,7 +360,7 @@ void cache_audio_worker(Clip* c, bool scrubbing, QVector<Clip*>& nests) {
if (frame->nb_samples == 0) {
break;
} else {
long buffer_timeline_out = get_buffer_offset_from_frame((sequence == NULL) ? c->sequence : sequence, timeline_out);
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
@@ -757,7 +757,7 @@ void open_clip_worker(Clip* clip) {
AVFrame* reverse_frame = av_frame_alloc();
reverse_frame->format = sample_format;
reverse_frame->nb_samples = clip->sequence->audio_frequency*2;
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);
@@ -843,7 +843,7 @@ void open_clip_worker(Clip* clip) {
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 = clip->sequence->audio_frequency;
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)) {