fixed audio monitor regression
This commit is contained in:
+28
-34
@@ -164,47 +164,36 @@ void AudioSenderThread::run() {
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lock.unlock();
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}
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int AudioSenderThread::send_audio_to_output(int offset, int max) {
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int AudioSenderThread::send_audio_to_output(unsigned long offset, int max) {
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// send audio to device
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unsigned long actual_write = audio_io_device->write((const char*) audio_ibuffer+offset, max);
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unsigned long audio_ibuffer_limit = audio_ibuffer_read + actual_write;
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// send samples to audio monitor cache
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// TODO make this work for the footage viewer - currently, enabling it causes crash due to an ASSERT
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Sequence* s = nullptr;
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/*if (panel_footage_viewer->playing) {
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s = panel_footage_viewer->seq;
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}*/
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if (panel_sequence_viewer->playing) {
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s = panel_sequence_viewer->seq;
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}
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if (s != nullptr) {
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if (panel_timeline->audio_monitor->sample_cache_offset == -1) {
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panel_timeline->audio_monitor->sample_cache_offset = s->playhead;
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}
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int channel_count = av_get_channel_layout_nb_channels(s->audio_layout);
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long sample_cache_playhead = panel_timeline->audio_monitor->sample_cache_offset + (panel_timeline->audio_monitor->sample_cache.size()/channel_count);
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unsigned long next_buffer_offset, buffer_offset_adjusted;
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int i;
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unsigned long buffer_offset = get_buffer_offset_from_frame(s->frame_rate, sample_cache_playhead);
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if (samples.size() != channel_count) samples.resize(channel_count);
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samples.fill(0);
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if (actual_write > 0) {
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// average values and send to audio monitor
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int channels = audio_output->format().channelCount();
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unsigned long lim = offset + actual_write;
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QVector<double> averages;
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averages.resize(channels);
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averages.fill(0);
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// TODO: I don't like this, but i'm not sure if there's a smarter way to do it
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while (static_cast<unsigned long>(buffer_offset) < audio_ibuffer_limit) {
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sample_cache_playhead++;
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next_buffer_offset = qMin(get_buffer_offset_from_frame(s->frame_rate, sample_cache_playhead), static_cast<unsigned long>(audio_ibuffer_limit));
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while (buffer_offset < next_buffer_offset) {
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for (i=0;i<samples.size();i++) {
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buffer_offset_adjusted = buffer_offset%audio_ibuffer_size;
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samples[i] = qMax(qAbs(qint16(((audio_ibuffer[buffer_offset_adjusted+1] & 0xFF) << 8) | (audio_ibuffer[buffer_offset_adjusted] & 0xFF))), samples[i]);
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buffer_offset += 2;
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}
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}
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panel_timeline->audio_monitor->sample_cache.append(samples);
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buffer_offset = next_buffer_offset;
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for (int i=0;i<channels;i++) {
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averages[i] = 0;
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}
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int counter = 0;
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qint16 sample;
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for (unsigned long i=offset;i<lim;i+=2) {
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sample = qint16(((audio_ibuffer[i+1] & 0xFF) << 8) | (audio_ibuffer[i] & 0xFF));
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averages[counter%channels] += qAbs(double(sample)/32768.0);
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counter++;
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}
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double divider = counter / channels;
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for (int i=0;i<channels;i++) {
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averages[i] = log_volume(1.0-(averages[i]/divider));
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}
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panel_timeline->audio_monitor->set_value(averages);
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}
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memset(audio_ibuffer+offset, 0, actual_write);
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@@ -214,6 +203,11 @@ int AudioSenderThread::send_audio_to_output(int offset, int max) {
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return actual_write;
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}
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double log_volume(double linear) {
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// expects a value between 0 and 1 (or more if amplifying)
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return (qExp(linear)-1)/(M_E-1);
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}
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void int32_to_char_array(qint32 i, char* array) {
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memcpy(array, &i, 4);
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}
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+3
-1
@@ -27,9 +27,11 @@ public slots:
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void notifyReceiver();
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private:
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QVector<qint16> samples;
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int send_audio_to_output(int offset, int max);
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int send_audio_to_output(unsigned long offset, int max);
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};
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double log_volume(double linear);
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extern QAudioOutput* audio_output;
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extern QIODevice* audio_io_device;
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extern AudioSenderThread* audio_thread;
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+1
-6
@@ -693,7 +693,7 @@ void Effect::startEffect() {
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void Effect::endEffect() {
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if (bound) glslProgram->release();
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bound = false;
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bound = false;
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}
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void Effect::process_image(double, uint8_t *, uint8_t *, int){}
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@@ -912,8 +912,3 @@ qint16 mix_audio_sample(qint16 a, qint16 b) {
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mixed_sample = qMax(qMin(mixed_sample, static_cast<qint32>(INT16_MAX)), static_cast<qint32>(INT16_MIN));
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return static_cast<qint16>(mixed_sample);
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}
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double log_volume(double linear) {
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// expects a value between 0 and 1 (or more if amplifying)
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return (qExp(linear)-1)/(M_E-1);
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}
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+38
-63
@@ -9,6 +9,8 @@
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#include <QLinearGradient>
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#include <QtMath>
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#include <QDebug>
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#define AUDIO_MONITOR_PEAK_HEIGHT 15
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#define AUDIO_MONITOR_GAP 3
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@@ -16,79 +18,52 @@ extern "C" {
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#include "libavformat/avformat.h"
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}
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AudioMonitor::AudioMonitor(QWidget *parent) : QWidget(parent) {
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reset();
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AudioMonitor::AudioMonitor(QWidget *parent) :
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QWidget(parent)
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{
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values.resize(2);
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values.fill(1.0);
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}
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void AudioMonitor::reset() {
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sample_cache_offset = -1;
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sample_cache.clear();
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update();
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void AudioMonitor::set_value(const QVector<double> &ivalues) {
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values = ivalues;
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update();
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}
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void AudioMonitor::resizeEvent(QResizeEvent *e) {
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gradient = QLinearGradient(QPoint(0, rect().top()), QPoint(0, rect().bottom()));
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gradient.setColorAt(0, Qt::red);
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gradient.setColorAt(0.25, Qt::yellow);
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gradient.setColorAt(1, Qt::green);
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QWidget::resizeEvent(e);
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gradient = QLinearGradient(QPoint(0, rect().top()), QPoint(0, rect().bottom()));
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gradient.setColorAt(0, Qt::red);
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gradient.setColorAt(0.25, Qt::yellow);
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gradient.setColorAt(1, Qt::green);
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QWidget::resizeEvent(e);
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}
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void AudioMonitor::paintEvent(QPaintEvent *) {
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if (sequence != nullptr) {
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QPainter p(this);
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int channel_x = AUDIO_MONITOR_GAP;
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int channel_count = av_get_channel_layout_nb_channels(sequence->audio_layout);
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/*if (peaks.size() != channel_count) {
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peaks.resize(channel_count);
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peaks.fill(false);
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}*/
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int channel_width = (width()/channel_count) - AUDIO_MONITOR_GAP;
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long playhead_offset = -1;
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int i;
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for (i=0;i<channel_count;i++) {
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QRect r(channel_x, AUDIO_MONITOR_PEAK_HEIGHT + AUDIO_MONITOR_GAP, channel_width, height());
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p.fillRect(r, gradient);
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if (sequence != nullptr) {
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QPainter p(this);
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int channel_x = AUDIO_MONITOR_GAP;
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// int channel_count = av_get_channel_layout_nb_channels(sequence->audio_layout);
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int channel_count = values.size();
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int channel_width = (width()/channel_count) - AUDIO_MONITOR_GAP;
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int i;
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for (i=0;i<channel_count;i++) {
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QRect r(channel_x, AUDIO_MONITOR_PEAK_HEIGHT + AUDIO_MONITOR_GAP, channel_width, height());
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p.fillRect(r, gradient);
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if (sample_cache_offset != -1 && sample_cache.size() > 0) {
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playhead_offset = (sequence->playhead - sample_cache_offset) * channel_count;
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if (playhead_offset >= 0 && playhead_offset < sample_cache.size()) {
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double multiplier = 1 - qAbs((double) sample_cache.at(playhead_offset+i) / 32768.0); // 16-bit int divided to float
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/*if (multiplier == (double) 0) {
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peaks[i] = true;
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}*/
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r.setHeight(r.height()*multiplier);
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} else {
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reset();
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}
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}
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bool peak = false;
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/*QRect peak_rect(channel_x, 0, channel_width, AUDIO_MONITOR_PEAK_HEIGHT);
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if (peaks.at(i)) {
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p.fillRect(peak_rect, QColor(255, 0, 0));
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} else {
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p.fillRect(peak_rect, QColor(64, 0, 0));
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}*/
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r.setHeight(qRound(r.height()*(values.at(i))));
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p.fillRect(r, QColor(0, 0, 0, 160));
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QRect peak_rect(channel_x, 0, channel_width, AUDIO_MONITOR_PEAK_HEIGHT);
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if (peak) {
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p.fillRect(peak_rect, QColor(255, 0, 0));
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} else {
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p.fillRect(peak_rect, QColor(64, 0, 0));
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}
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channel_x += channel_width + AUDIO_MONITOR_GAP;
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}
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if (playhead_offset > -1) {
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// clean up used samples
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bool error = false;
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while (sample_cache_offset < sequence->playhead && !error) {
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sample_cache_offset++;
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for (i=0;i<channel_count;i++) {
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if (sample_cache.isEmpty()) {
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reset();
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error = true;
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break;
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} else {
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sample_cache.removeFirst();
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}
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}
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}
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}
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}
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p.fillRect(r, QColor(0, 0, 0, 160));
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channel_x += channel_width + AUDIO_MONITOR_GAP;
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}
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}
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}
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+7
-9
@@ -5,24 +5,22 @@
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class AudioMonitor : public QWidget
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{
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Q_OBJECT
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Q_OBJECT
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public:
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explicit AudioMonitor(QWidget *parent = 0);
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QVector<qint16> sample_cache;
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long sample_cache_offset;
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void reset();
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explicit AudioMonitor(QWidget *parent = 0);
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void set_value(const QVector<double>& values);
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protected:
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void paintEvent(QPaintEvent *);
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void resizeEvent(QResizeEvent *);
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void paintEvent(QPaintEvent *);
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void resizeEvent(QResizeEvent *);
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signals:
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public slots:
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private:
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QLinearGradient gradient;
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// QVector<bool> peaks;
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QLinearGradient gradient;
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QVector<double> values;
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};
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#endif // AUDIOMONITOR_H
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