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converted internal system to float

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This commit is contained in:
itsmattkc
2019-04-07 13:37:28 +10:00
parent a2d7025776
commit 644d096f3c
31 changed files with 365 additions and 257 deletions
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rendering/cacher.cpp
+46 -34
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@@ -46,21 +46,25 @@
// Enable verbose audio messages - good for debugging reversed audio
//#define AUDIOWARNINGS
const AVSampleFormat kDestSampleFmt = AV_SAMPLE_FMT_S16;
const AVSampleFormat kDestSampleFmt = AV_SAMPLE_FMT_FLTP;
double bytes_to_seconds(int nb_bytes, int nb_channels, int sample_rate) {
return (double(nb_bytes >> 1) / nb_channels / sample_rate);
double samples_to_seconds(int nb_samples, int nb_channels, int sample_rate) {
return (double(nb_samples) / double(nb_channels) / double(sample_rate));
}
void apply_audio_effects(Clip* clip, double timecode_start, AVFrame* frame, int nb_bytes, QVector<Clip*> nests) {
int samples_to_bytes(int nb_samples, int nb_channels) {
return nb_samples * nb_channels * sizeof(float);
}
void apply_audio_effects(Clip* clip, double timecode_start, AVFrame* frame, int nb_samples, int nb_channels, QVector<Clip*> nests) {
// perform all audio effects
double timecode_end;
timecode_end = timecode_start + bytes_to_seconds(nb_bytes, frame->channels, frame->sample_rate);
timecode_end = timecode_start + samples_to_seconds(nb_samples, frame->channels, frame->sample_rate);
for (int j=0;j<clip->effects.size();j++) {
Effect* e = clip->effects.at(j).get();
if (e->IsEnabled()) {
e->process_audio(timecode_start, timecode_end, frame->data[0], nb_bytes, 2);
e->process_audio(timecode_start, timecode_end, reinterpret_cast<float**>(frame->data), nb_samples, nb_channels, kTransitionNone);
}
}
if (clip->opening_transition != nullptr) {
@@ -71,7 +75,7 @@ void apply_audio_effects(Clip* clip, double timecode_start, AVFrame* frame, int
double adjustment = transition_end - transition_start;
double adjusted_range_start = (timecode_start - transition_start) / adjustment;
double adjusted_range_end = (timecode_end - transition_start) / adjustment;
clip->opening_transition->process_audio(adjusted_range_start, adjusted_range_end, frame->data[0], nb_bytes, kTransitionOpening);
clip->opening_transition->process_audio(adjusted_range_start, adjusted_range_end, reinterpret_cast<float**>(frame->data), nb_samples, nb_channels, kTransitionOpening);
}
}
}
@@ -84,7 +88,7 @@ void apply_audio_effects(Clip* clip, double timecode_start, AVFrame* frame, int
double adjustment = transition_end - transition_start;
double adjusted_range_start = (timecode_start - transition_start) / adjustment;
double adjusted_range_end = (timecode_end - transition_start) / adjustment;
clip->closing_transition->process_audio(adjusted_range_start, adjusted_range_end, frame->data[0], nb_bytes, kTransitionClosing);
clip->closing_transition->process_audio(adjusted_range_start, adjusted_range_end, reinterpret_cast<float**>(frame->data), nb_samples, nb_channels, kTransitionClosing);
}
}
}
@@ -95,7 +99,8 @@ void apply_audio_effects(Clip* clip, double timecode_start, AVFrame* frame, int
apply_audio_effects(next_nest,
timecode_start + (double(clip->timeline_in(true)-clip->clip_in(true))/clip->track()->sequence()->frame_rate),
frame,
nb_bytes,
nb_samples,
nb_channels,
nests);
}
}
@@ -157,16 +162,16 @@ void Cacher::CacheAudioWorker() {
while (true) {
AVFrame* frame;
int nb_bytes = INT_MAX;
int nb_samples = INT_MAX;
if (clip->media() == nullptr) {
frame = frame_;
nb_bytes = frame->nb_samples * av_get_bytes_per_sample(static_cast<AVSampleFormat>(frame->format)) * frame->channels;
while ((frame_sample_index_ == -1 || frame_sample_index_ >= nb_bytes) && nb_bytes > 0) {
nb_samples = frame->nb_samples;
while ((frame_sample_index_ == -1 || frame_sample_index_ >= nb_samples) && nb_samples > 0) {
// create "new frame"
memset(frame_->data[0], 0, nb_bytes);
apply_audio_effects(clip, bytes_to_seconds(frame->pts, frame->channels, frame->sample_rate), frame, nb_bytes, nests_);
frame_->pts += nb_bytes;
memset(frame_->data[0], 0, nb_samples);
apply_audio_effects(clip, samples_to_seconds(frame->pts, frame->channels, frame->sample_rate), frame, nb_samples, frame->channels, nests_);
frame_->pts += nb_samples;
frame_sample_index_ = 0;
if (audio_buffer_write == 0) {
audio_buffer_write = get_buffer_offset_from_frame(last_fr, qMax(timeline_in, target_frame));
@@ -184,7 +189,7 @@ void Cacher::CacheAudioWorker() {
// retrieve frame
bool new_frame = false;
while ((frame_sample_index_ == -1 || frame_sample_index_ >= nb_bytes) && nb_bytes > 0) {
while ((frame_sample_index_ == -1 || frame_sample_index_ >= nb_samples) && nb_samples > 0) {
// no more audio left in frame, get a new one
if (!reached_end) {
@@ -338,10 +343,10 @@ void Cacher::CacheAudioWorker() {
if (frame_sample_index_ < 0) {
frame_sample_index_ = 0;
} else {
frame_sample_index_ -= nb_bytes;
frame_sample_index_ -= nb_samples;
}
nb_bytes = frame->nb_samples * av_get_bytes_per_sample(static_cast<AVSampleFormat>(frame->format)) * frame->channels;
nb_samples = frame->nb_samples;
if (audio_just_reset) {
// get precise sample offset for the elected clip_in from this audio frame
@@ -356,7 +361,7 @@ void Cacher::CacheAudioWorker() {
dout << "fsts:" << frame_sts << "tsts:" << target_sts << "nbs:" << nb_samples << "nbb:" << nb_bytes << "rev_targetToSec:" << (reverse_target * timebase);
dout << "fsi-calc:" << frame_sample_index;
#endif
if (reverse_audio) frame_sample_index_ = nb_bytes - frame_sample_index_;
if (reverse_audio) frame_sample_index_ = nb_samples - frame_sample_index_;
audio_just_reset = false;
}
@@ -393,9 +398,19 @@ void Cacher::CacheAudioWorker() {
#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 (nb_samples == INT_MAX) {
nb_samples = frame->nb_samples;
}
if (new_frame) {
apply_audio_effects(clip, bytes_to_seconds(audio_buffer_write, 2, current_audio_freq()) + audio_ibuffer_timecode + ((double)clip->clip_in(true)/clip->track()->sequence()->frame_rate) - ((double)timeline_in/last_fr), frame, nb_bytes, nests_);
apply_audio_effects(clip,
samples_to_seconds(audio_buffer_write, 2, current_audio_freq())
+ audio_ibuffer_timecode
+ (double(clip->clip_in(true))/clip->track()->sequence()->frame_rate)
- (double(timeline_in)/last_fr),
frame,
nb_samples,
frame->channels,
nests_);
}
}
@@ -408,23 +423,20 @@ void Cacher::CacheAudioWorker() {
audio_write_lock.lock();
int sample_skip = 4*qMax(0, qAbs(playback_speed_)-1);
int sample_byte_size = av_get_bytes_per_sample(static_cast<AVSampleFormat>(frame->format));
while (frame_sample_index_ < nb_bytes
while (frame_sample_index_ < nb_samples
&& audio_buffer_write < audio_ibuffer_read+(audio_ibuffer_size>>1)
&& audio_buffer_write < buffer_timeline_out) {
for (int i=0;i<frame->channels;i++) {
int upper_byte_index = (audio_buffer_write+1)%audio_ibuffer_size;
int lower_byte_index = (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][frame_sample_index_+1] & 0xFF) << 8 | (frame->data[0][frame_sample_index_] & 0xFF));
qint16 mixed_sample = mix_audio_sample(old_sample, new_sample);
int buffer_index = audio_buffer_write%audio_ibuffer_size;
int frame_index = frame_sample_index_ * sizeof(float);
audio_ibuffer[upper_byte_index] = quint8((mixed_sample >> 8) & 0xFF);
audio_ibuffer[lower_byte_index] = quint8(mixed_sample & 0xFF);
audio_buffer_write+=sample_byte_size;
frame_sample_index_+=sample_byte_size;
audio_ibuffer[buffer_index] += static_cast<float>((frame->data[0][frame_index+3] & 0xFF) << 24
| (frame->data[0][frame_index+2] & 0xFF) << 16
| (frame->data[0][frame_index+1] & 0xFF) << 8
| (frame->data[0][frame_index] & 0xFF));
audio_buffer_write++;
frame_sample_index_++;
}
frame_sample_index_ += sample_skip;
@@ -444,7 +456,7 @@ void Cacher::CacheAudioWorker() {
if (audio_thread != nullptr) audio_thread->notifyReceiver();
}
if (frame_sample_index_ >= nb_bytes) {
if (frame_sample_index_ >= nb_samples) {
frame_sample_index_ = -1;
} else {
// assume we have no more data to send