removed all ui files
This commit is contained in:
itsmattkc
+55
-54
@@ -7,9 +7,10 @@
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#include "panels/panels.h"
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#include "panels/timeline.h"
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#include "panels/viewer.h"
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#include "ui_timeline.h"
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#include "ui/audiomonitor.h"
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#include "debug.h"
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#include <QApplication>
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#include <QAudioOutput>
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#include <QAudioInput>
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#include <QtMath>
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@@ -43,49 +44,49 @@ bool is_audio_device_set() {
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void init_audio() {
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stop_audio();
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QAudioFormat audio_format;
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audio_format.setSampleRate(config.audio_rate);
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audio_format.setChannelCount(2);
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audio_format.setSampleSize(16);
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audio_format.setCodec("audio/pcm");
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audio_format.setByteOrder(QAudioFormat::LittleEndian);
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audio_format.setSampleType(QAudioFormat::SignedInt);
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QAudioFormat audio_format;
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audio_format.setSampleRate(config.audio_rate);
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audio_format.setChannelCount(2);
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audio_format.setSampleSize(16);
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audio_format.setCodec("audio/pcm");
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audio_format.setByteOrder(QAudioFormat::LittleEndian);
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audio_format.setSampleType(QAudioFormat::SignedInt);
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QAudioDeviceInfo info(QAudioDeviceInfo::defaultOutputDevice());
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QList<QAudioDeviceInfo> devs = QAudioDeviceInfo::availableDevices(QAudio::AudioOutput);
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dout << "[INFO] Found the following audio devices:";
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for (int i=0;i<devs.size();i++) {
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dout << " " << devs.at(i).deviceName();
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}
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if (info.isNull() && devs.size() > 0) {
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dout << "[WARNING] Default audio returned NULL, attempting to use first device found...";
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info = devs.at(0);
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}
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dout << "[INFO] Using audio device" << info.deviceName();
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QAudioDeviceInfo info(QAudioDeviceInfo::defaultOutputDevice());
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QList<QAudioDeviceInfo> devs = QAudioDeviceInfo::availableDevices(QAudio::AudioOutput);
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dout << "[INFO] Found the following audio devices:";
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for (int i=0;i<devs.size();i++) {
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dout << " " << devs.at(i).deviceName();
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}
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if (info.isNull() && devs.size() > 0) {
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dout << "[WARNING] Default audio returned NULL, attempting to use first device found...";
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info = devs.at(0);
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}
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dout << "[INFO] Using audio device" << info.deviceName();
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if (!info.isFormatSupported(audio_format)) {
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qWarning() << "[WARNING] Audio format is not supported by backend, using nearest";
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audio_format = info.nearestFormat(audio_format);
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}
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if (!info.isFormatSupported(audio_format)) {
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qWarning() << "[WARNING] Audio format is not supported by backend, using nearest";
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audio_format = info.nearestFormat(audio_format);
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}
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audio_output = new QAudioOutput(info, audio_format);
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audio_output->moveToThread(QApplication::instance()->thread());
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audio_output->setNotifyInterval(5);
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audio_output = new QAudioOutput(info, audio_format);
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audio_output->moveToThread(QApplication::instance()->thread());
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audio_output->setNotifyInterval(5);
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// connect
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audio_io_device = audio_output->start();
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if (audio_io_device == NULL) {
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dout << "[WARNING] Received NULL audio device. No compatible audio output was found.";
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} else {
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audio_device_set = true;
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// connect
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audio_io_device = audio_output->start();
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if (audio_io_device == NULL) {
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dout << "[WARNING] Received NULL audio device. No compatible audio output was found.";
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} else {
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audio_device_set = true;
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// start sender thread
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audio_thread = new AudioSenderThread();
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QObject::connect(audio_output, SIGNAL(notify()), audio_thread, SLOT(notifyReceiver()));
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audio_thread->start(QThread::TimeCriticalPriority);
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// start sender thread
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audio_thread = new AudioSenderThread();
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QObject::connect(audio_output, SIGNAL(notify()), audio_thread, SLOT(notifyReceiver()));
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audio_thread->start(QThread::TimeCriticalPriority);
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clear_audio_ibuffer();
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}
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clear_audio_ibuffer();
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}
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}
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void stop_audio() {
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@@ -100,12 +101,12 @@ void stop_audio() {
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void clear_audio_ibuffer() {
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memset(audio_ibuffer, 0, audio_ibuffer_size);
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audio_ibuffer_read = 0;
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audio_ibuffer_read = 0;
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}
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int get_buffer_offset_from_frame(double framerate, long frame) {
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if (frame >= audio_ibuffer_frame) {
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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);
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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);
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} else {
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dout << "[WARNING] Invalid values passed to get_buffer_offset_from_frame";
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return 0;
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@@ -135,7 +136,7 @@ void AudioSenderThread::run() {
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cond.wait(&lock);
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if (close) {
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break;
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} else if (panel_sequence_viewer->playing || panel_footage_viewer->playing || audio_scrub) {
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} else if (panel_sequence_viewer->playing || panel_footage_viewer->playing || audio_scrub) {
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int written_bytes = 0;
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int adjusted_read_index = audio_ibuffer_read%audio_ibuffer_size;
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@@ -147,7 +148,7 @@ void AudioSenderThread::run() {
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written_bytes += send_audio_to_output(0, audio_ibuffer_size);
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}
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audio_scrub = false;
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audio_scrub = false;
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}
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}
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lock.unlock();
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@@ -163,26 +164,26 @@ int AudioSenderThread::send_audio_to_output(int offset, int max) {
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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 = NULL;
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/*if (panel_footage_viewer->playing) {
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s = panel_footage_viewer->seq;
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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 != NULL) {
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if (panel_timeline->ui->audio_monitor->sample_cache_offset == -1) {
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panel_timeline->ui->audio_monitor->sample_cache_offset = s->playhead;
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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 != NULL) {
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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->ui->audio_monitor->sample_cache_offset + (panel_timeline->ui->audio_monitor->sample_cache.size()/channel_count);
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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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int next_buffer_offset, buffer_offset_adjusted, i;
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int buffer_offset = get_buffer_offset_from_frame(s->frame_rate, sample_cache_playhead);
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int 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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// 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 (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), audio_ibuffer_limit);
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next_buffer_offset = qMin(get_buffer_offset_from_frame(s->frame_rate, sample_cache_playhead), 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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@@ -190,7 +191,7 @@ int AudioSenderThread::send_audio_to_output(int offset, int max) {
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buffer_offset += 2;
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}
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}
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panel_timeline->ui->audio_monitor->sample_cache.append(samples);
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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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}
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}
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