2673 lines
69 KiB
C++
2673 lines
69 KiB
C++
/*
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Copyright (C) 2000-2006 Paul Davis
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This program is free software; you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation; either version 2 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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*/
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#include <cstdio>
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#include <unistd.h>
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#include <cmath>
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#include <cerrno>
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#include <cassert>
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#include <string>
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#include <climits>
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#include <fcntl.h>
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#include <cstdlib>
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#include <ctime>
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#include "pbd/gstdio_compat.h"
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#include "pbd/error.h"
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#include "pbd/xml++.h"
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#include "pbd/memento_command.h"
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#include "pbd/enumwriter.h"
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#include "pbd/stateful_diff_command.h"
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#include "ardour/analyser.h"
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#include "ardour/audio_buffer.h"
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#include "ardour/audio_diskstream.h"
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#include "ardour/audio_port.h"
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#include "ardour/audioengine.h"
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#include "ardour/audiofilesource.h"
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#include "ardour/audioplaylist.h"
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#include "ardour/audioregion.h"
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#include "ardour/butler.h"
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#include "ardour/debug.h"
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#include "ardour/io.h"
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#include "ardour/playlist_factory.h"
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#include "ardour/profile.h"
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#include "ardour/region_factory.h"
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#include "ardour/session.h"
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#include "ardour/session_playlists.h"
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#include "ardour/sndfile_helpers.h"
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#include "ardour/source_factory.h"
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#include "ardour/track.h"
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#include "ardour/types.h"
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#include "ardour/utils.h"
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#include "pbd/i18n.h"
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#include <locale.h>
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using namespace std;
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using namespace ARDOUR;
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using namespace PBD;
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Sample* AudioDiskstream::_mixdown_buffer = 0;
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gain_t* AudioDiskstream::_gain_buffer = 0;
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AudioDiskstream::AudioDiskstream (Session &sess, const string &name, Diskstream::Flag flag)
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: Diskstream(sess, name, flag)
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, channels (new ChannelList)
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{
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/* prevent any write sources from being created */
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in_set_state = true;
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use_new_playlist ();
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in_set_state = false;
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if (flag & Destructive) {
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use_destructive_playlist ();
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}
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}
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AudioDiskstream::AudioDiskstream (Session& sess, const XMLNode& node)
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: Diskstream(sess, node)
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, channels (new ChannelList)
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{
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in_set_state = true;
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init ();
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if (set_state (node, Stateful::loading_state_version)) {
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in_set_state = false;
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throw failed_constructor();
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}
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in_set_state = false;
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if (destructive()) {
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use_destructive_playlist ();
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}
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}
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void
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AudioDiskstream::init ()
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{
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/* there are no channels at this point, so these
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two calls just get speed_buffer_size and wrap_buffer
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size setup without duplicating their code.
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*/
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set_block_size (_session.get_block_size());
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allocate_temporary_buffers ();
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}
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AudioDiskstream::~AudioDiskstream ()
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{
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DEBUG_TRACE (DEBUG::Destruction, string_compose ("Audio Diskstream %1 destructor\n", _name));
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{
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RCUWriter<ChannelList> writer (channels);
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boost::shared_ptr<ChannelList> c = writer.get_copy();
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for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
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delete *chan;
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}
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c->clear();
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}
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channels.flush ();
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}
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void
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AudioDiskstream::allocate_working_buffers()
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{
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/* with varifill buffer refilling, we compute the read size in bytes (to optimize
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for disk i/o bandwidth) and then convert back into samples. These buffers
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need to reflect the maximum size we could use, which is 4MB reads, or 2M samples
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using 16 bit samples.
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*/
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_mixdown_buffer = new Sample[2*1048576];
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_gain_buffer = new gain_t[2*1048576];
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}
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void
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AudioDiskstream::free_working_buffers()
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{
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delete [] _mixdown_buffer;
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delete [] _gain_buffer;
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_mixdown_buffer = 0;
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_gain_buffer = 0;
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}
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void
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AudioDiskstream::non_realtime_input_change ()
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{
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bool need_write_sources = false;
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{
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Glib::Threads::Mutex::Lock lm (state_lock);
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if (input_change_pending.type == IOChange::NoChange) {
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return;
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}
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boost::shared_ptr<ChannelList> cr = channels.reader();
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if (!cr->empty() && !cr->front()->write_source) {
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need_write_sources = true;
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}
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if (input_change_pending.type & IOChange::ConfigurationChanged) {
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RCUWriter<ChannelList> writer (channels);
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boost::shared_ptr<ChannelList> c = writer.get_copy();
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_n_channels.set(DataType::AUDIO, c->size());
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if (_io->n_ports().n_audio() > _n_channels.n_audio()) {
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add_channel_to (c, _io->n_ports().n_audio() - _n_channels.n_audio());
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} else if (_io->n_ports().n_audio() < _n_channels.n_audio()) {
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remove_channel_from (c, _n_channels.n_audio() - _io->n_ports().n_audio());
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}
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need_write_sources = true;
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}
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if (input_change_pending.type & IOChange::ConnectionsChanged) {
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get_input_sources ();
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set_capture_offset ();
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set_align_style_from_io ();
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}
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input_change_pending = IOChange::NoChange;
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/* implicit unlock */
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}
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if (need_write_sources) {
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reset_write_sources (false);
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}
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/* now refill channel buffers */
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if (speed() != 1.0f || speed() != -1.0f) {
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seek ((framepos_t) (_session.transport_frame() * (double) speed()));
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} else {
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seek (_session.transport_frame());
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}
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}
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void
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AudioDiskstream::non_realtime_locate (framepos_t location)
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{
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/* now refill channel buffers */
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if (speed() != 1.0f || speed() != -1.0f) {
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seek ((framepos_t) (location * (double) speed()), true);
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} else {
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seek (location, true);
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}
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}
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void
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AudioDiskstream::get_input_sources ()
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{
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boost::shared_ptr<ChannelList> c = channels.reader();
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uint32_t n;
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ChannelList::iterator chan;
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uint32_t ni = _io->n_ports().n_audio();
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vector<string> connections;
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for (n = 0, chan = c->begin(); chan != c->end() && n < ni; ++chan, ++n) {
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connections.clear ();
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if ((_io->nth (n).get()) && (_io->nth (n)->get_connections (connections) == 0)) {
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if (!(*chan)->source.name.empty()) {
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// _source->disable_metering ();
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}
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(*chan)->source.name = string();
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} else {
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(*chan)->source.name = connections[0];
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}
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}
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}
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int
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AudioDiskstream::find_and_use_playlist (const string& name)
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{
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boost::shared_ptr<AudioPlaylist> playlist;
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if ((playlist = boost::dynamic_pointer_cast<AudioPlaylist> (_session.playlists->by_name (name))) == 0) {
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playlist = boost::dynamic_pointer_cast<AudioPlaylist> (PlaylistFactory::create (DataType::AUDIO, _session, name));
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}
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if (!playlist) {
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error << string_compose(_("AudioDiskstream: Playlist \"%1\" isn't an audio playlist"), name) << endmsg;
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return -1;
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}
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return use_playlist (playlist);
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}
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int
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AudioDiskstream::use_playlist (boost::shared_ptr<Playlist> playlist)
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{
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assert(boost::dynamic_pointer_cast<AudioPlaylist>(playlist));
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Diskstream::use_playlist(playlist);
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return 0;
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}
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int
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AudioDiskstream::use_new_playlist ()
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{
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string newname;
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boost::shared_ptr<AudioPlaylist> playlist;
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if (!in_set_state && destructive()) {
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return 0;
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}
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if (_playlist) {
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newname = Playlist::bump_name (_playlist->name(), _session);
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} else {
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newname = Playlist::bump_name (_name, _session);
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}
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if ((playlist = boost::dynamic_pointer_cast<AudioPlaylist> (PlaylistFactory::create (DataType::AUDIO, _session, newname, hidden()))) != 0) {
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return use_playlist (playlist);
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} else {
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return -1;
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}
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}
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int
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AudioDiskstream::use_copy_playlist ()
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{
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assert(audio_playlist());
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if (destructive()) {
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return 0;
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}
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if (_playlist == 0) {
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error << string_compose(_("AudioDiskstream %1: there is no existing playlist to make a copy of!"), _name) << endmsg;
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return -1;
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}
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string newname;
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boost::shared_ptr<AudioPlaylist> playlist;
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newname = Playlist::bump_name (_playlist->name(), _session);
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if ((playlist = boost::dynamic_pointer_cast<AudioPlaylist>(PlaylistFactory::create (audio_playlist(), newname))) != 0) {
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playlist->reset_shares();
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return use_playlist (playlist);
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} else {
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return -1;
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}
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}
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void
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AudioDiskstream::setup_destructive_playlist ()
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{
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SourceList srcs;
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boost::shared_ptr<ChannelList> c = channels.reader();
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for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
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srcs.push_back ((*chan)->write_source);
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}
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/* a single full-sized region */
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assert (!srcs.empty ());
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PropertyList plist;
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plist.add (Properties::name, _name.val());
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plist.add (Properties::start, 0);
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plist.add (Properties::length, max_framepos - srcs.front()->natural_position());
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boost::shared_ptr<Region> region (RegionFactory::create (srcs, plist));
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_playlist->add_region (region, srcs.front()->natural_position());
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/* apply region properties and update write sources */
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use_destructive_playlist();
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}
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void
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AudioDiskstream::use_destructive_playlist ()
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{
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/* this is called from the XML-based constructor or ::set_destructive. when called,
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we already have a playlist and a region, but we need to
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set up our sources for write. we use the sources associated
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with the (presumed single, full-extent) region.
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*/
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boost::shared_ptr<Region> rp;
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{
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const RegionList& rl (_playlist->region_list_property().rlist());
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if (rl.size() > 0) {
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/* this can happen when dragging a region onto a tape track */
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assert((rl.size() == 1));
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rp = rl.front();
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}
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}
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if (!rp) {
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reset_write_sources (false, true);
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return;
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}
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boost::shared_ptr<AudioRegion> region = boost::dynamic_pointer_cast<AudioRegion> (rp);
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if (region == 0) {
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throw failed_constructor();
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}
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/* be sure to stretch the region out to the maximum length (non-musical)*/
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region->set_length (max_framepos - region->position(), 0);
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uint32_t n;
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ChannelList::iterator chan;
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boost::shared_ptr<ChannelList> c = channels.reader();
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for (n = 0, chan = c->begin(); chan != c->end(); ++chan, ++n) {
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(*chan)->write_source = boost::dynamic_pointer_cast<AudioFileSource>(region->source (n));
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assert((*chan)->write_source);
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(*chan)->write_source->set_allow_remove_if_empty (false);
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/* this might be false if we switched modes, so force it */
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#ifdef XXX_OLD_DESTRUCTIVE_API_XXX
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(*chan)->write_source->set_destructive (true);
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#else
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// should be set when creating the source or loading the state
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assert ((*chan)->write_source->destructive());
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#endif
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}
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/* the source list will never be reset for a destructive track */
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}
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void
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AudioDiskstream::prepare_record_status(framepos_t capture_start_frame)
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{
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if (recordable() && destructive()) {
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boost::shared_ptr<ChannelList> c = channels.reader();
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for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
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RingBufferNPT<CaptureTransition>::rw_vector transitions;
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(*chan)->capture_transition_buf->get_write_vector (&transitions);
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if (transitions.len[0] > 0) {
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transitions.buf[0]->type = CaptureStart;
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transitions.buf[0]->capture_val = capture_start_frame;
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(*chan)->capture_transition_buf->increment_write_ptr(1);
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} else {
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// bad!
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fatal << X_("programming error: capture_transition_buf is full on rec start! inconceivable!")
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<< endmsg;
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}
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}
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}
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}
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/** Do some record stuff [not described in this comment!]
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*
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* Also:
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* - Setup playback_distance with the nframes, or nframes adjusted
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* for current varispeed, if appropriate.
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* - Setup current_playback_buffer in each ChannelInfo to point to data
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* that someone can read playback_distance worth of data from.
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*/
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int
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AudioDiskstream::process (BufferSet& bufs, framepos_t transport_frame, pframes_t nframes, framecnt_t& playback_distance, bool need_disk_signal)
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{
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uint32_t n;
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boost::shared_ptr<ChannelList> c = channels.reader();
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ChannelList::iterator chan;
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framecnt_t rec_offset = 0;
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framecnt_t rec_nframes = 0;
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bool collect_playback = false;
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bool can_record = _session.actively_recording ();
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playback_distance = 0;
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if (!_io || !_io->active()) {
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return 0;
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}
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check_record_status (transport_frame, can_record);
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if (nframes == 0) {
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return 0;
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}
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Glib::Threads::Mutex::Lock sm (state_lock, Glib::Threads::TRY_LOCK);
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if (!sm.locked()) {
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return 1;
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}
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adjust_capture_position = 0;
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for (chan = c->begin(); chan != c->end(); ++chan) {
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(*chan)->current_capture_buffer = 0;
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(*chan)->current_playback_buffer = 0;
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}
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// Safeguard against situations where process() goes haywire when autopunching
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// and last_recordable_frame < first_recordable_frame
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if (last_recordable_frame < first_recordable_frame) {
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last_recordable_frame = max_framepos;
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}
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if (record_enabled()) {
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Evoral::OverlapType ot = Evoral::coverage (first_recordable_frame, last_recordable_frame, transport_frame, transport_frame + nframes);
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// XXX should this be transport_frame + nframes - 1 ? coverage() expects its parameter ranges to include their end points
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// XXX also, first_recordable_frame & last_recordable_frame may both be == max_framepos: coverage() will return OverlapNone in that case. Is thak OK?
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calculate_record_range (ot, transport_frame, nframes, rec_nframes, rec_offset);
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DEBUG_TRACE (DEBUG::CaptureAlignment, string_compose ("%1: this time record %2 of %3 frames, offset %4\n", _name, rec_nframes, nframes, rec_offset));
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if (rec_nframes && !was_recording) {
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capture_captured = 0;
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was_recording = true;
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}
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}
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if (can_record && !_last_capture_sources.empty()) {
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_last_capture_sources.clear ();
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}
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if (rec_nframes) {
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uint32_t limit = _io->n_ports ().n_audio();
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/* one or more ports could already have been removed from _io, but our
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channel setup hasn't yet been updated. prevent us from trying to
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use channels that correspond to missing ports. note that the
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process callback (from which this is called) is always atomic
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with respect to port removal/addition.
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*/
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for (n = 0, chan = c->begin(); chan != c->end() && n < limit; ++chan, ++n) {
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ChannelInfo* chaninfo (*chan);
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chaninfo->capture_buf->get_write_vector (&chaninfo->capture_vector);
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if (rec_nframes <= (framecnt_t) chaninfo->capture_vector.len[0]) {
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chaninfo->current_capture_buffer = chaninfo->capture_vector.buf[0];
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/* note: grab the entire port buffer, but only copy what we were supposed to
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for recording, and use rec_offset
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*/
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boost::shared_ptr<AudioPort> const ap = _io->audio (n);
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assert(ap);
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assert(rec_nframes <= (framecnt_t) ap->get_audio_buffer(nframes).capacity());
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Sample *buf = bufs.get_audio (n).data(rec_offset);
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memcpy (chaninfo->current_capture_buffer, buf, sizeof (Sample) * rec_nframes);
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} else {
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framecnt_t total = chaninfo->capture_vector.len[0] + chaninfo->capture_vector.len[1];
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|
|
if (rec_nframes > total) {
|
|
DEBUG_TRACE (DEBUG::Butler, string_compose ("%1 overrun in %2, rec_nframes = %3 total space = %4\n",
|
|
DEBUG_THREAD_SELF, name(), rec_nframes, total));
|
|
DiskOverrun ();
|
|
return -1;
|
|
}
|
|
|
|
boost::shared_ptr<AudioPort> const ap = _io->audio (n);
|
|
assert(ap);
|
|
|
|
Sample *buf = bufs.get_audio (n).data(rec_offset);
|
|
framecnt_t first = chaninfo->capture_vector.len[0];
|
|
|
|
memcpy (chaninfo->capture_wrap_buffer, buf, sizeof (Sample) * first);
|
|
memcpy (chaninfo->capture_vector.buf[0], buf, sizeof (Sample) * first);
|
|
memcpy (chaninfo->capture_wrap_buffer+first, buf + first, sizeof (Sample) * (rec_nframes - first));
|
|
memcpy (chaninfo->capture_vector.buf[1], buf + first, sizeof (Sample) * (rec_nframes - first));
|
|
|
|
chaninfo->current_capture_buffer = chaninfo->capture_wrap_buffer;
|
|
}
|
|
}
|
|
|
|
} else {
|
|
|
|
if (was_recording) {
|
|
finish_capture (c);
|
|
}
|
|
|
|
}
|
|
|
|
if (rec_nframes) {
|
|
|
|
/* data will be written to disk */
|
|
|
|
if (rec_nframes == nframes && rec_offset == 0) {
|
|
|
|
for (chan = c->begin(); chan != c->end(); ++chan) {
|
|
(*chan)->current_playback_buffer = (*chan)->current_capture_buffer;
|
|
}
|
|
|
|
playback_distance = nframes;
|
|
|
|
} else {
|
|
|
|
|
|
/* we can't use the capture buffer as the playback buffer, because
|
|
we recorded only a part of the current process' cycle data
|
|
for capture.
|
|
*/
|
|
|
|
collect_playback = true;
|
|
}
|
|
|
|
adjust_capture_position = rec_nframes;
|
|
|
|
} else if (can_record && record_enabled()) {
|
|
|
|
/* can't do actual capture yet - waiting for latency effects to finish before we start*/
|
|
|
|
for (chan = c->begin(); chan != c->end(); ++chan) {
|
|
(*chan)->current_playback_buffer = (*chan)->current_capture_buffer;
|
|
}
|
|
|
|
playback_distance = nframes;
|
|
|
|
} else {
|
|
|
|
collect_playback = true;
|
|
}
|
|
|
|
if ((_track->monitoring_state () & MonitoringDisk) || collect_playback) {
|
|
|
|
/* we're doing playback */
|
|
|
|
framecnt_t necessary_samples;
|
|
|
|
/* no varispeed playback if we're recording, because the output .... TBD */
|
|
|
|
if (rec_nframes == 0 && _actual_speed != 1.0) {
|
|
necessary_samples = (framecnt_t) ceil ((nframes * fabs (_actual_speed))) + 2;
|
|
} else {
|
|
necessary_samples = nframes;
|
|
}
|
|
|
|
for (chan = c->begin(); chan != c->end(); ++chan) {
|
|
(*chan)->playback_buf->get_read_vector (&(*chan)->playback_vector);
|
|
}
|
|
|
|
n = 0;
|
|
|
|
/* Setup current_playback_buffer in each ChannelInfo to point to data that someone
|
|
can read necessary_samples (== nframes at a transport speed of 1) worth of data
|
|
from right now.
|
|
*/
|
|
|
|
for (chan = c->begin(); chan != c->end(); ++chan, ++n) {
|
|
|
|
ChannelInfo* chaninfo (*chan);
|
|
|
|
if (necessary_samples <= (framecnt_t) chaninfo->playback_vector.len[0]) {
|
|
/* There are enough samples in the first part of the ringbuffer */
|
|
chaninfo->current_playback_buffer = chaninfo->playback_vector.buf[0];
|
|
|
|
} else {
|
|
framecnt_t total = chaninfo->playback_vector.len[0] + chaninfo->playback_vector.len[1];
|
|
|
|
if (necessary_samples > total) {
|
|
cerr << _name << " Need " << necessary_samples << " total = " << total << endl;
|
|
cerr << "underrun for " << _name << endl;
|
|
DEBUG_TRACE (DEBUG::Butler, string_compose ("%1 underrun in %2, rec_nframes = %3 total space = %4\n",
|
|
DEBUG_THREAD_SELF, name(), rec_nframes, total));
|
|
DiskUnderrun ();
|
|
return -1;
|
|
|
|
} else {
|
|
|
|
/* We have enough samples, but not in one lump. Coalesce the two parts
|
|
into one in playback_wrap_buffer in our ChannelInfo, and specify that
|
|
as our current_playback_buffer.
|
|
*/
|
|
|
|
assert(wrap_buffer_size >= necessary_samples);
|
|
|
|
/* Copy buf[0] from playback_buf */
|
|
memcpy ((char *) chaninfo->playback_wrap_buffer,
|
|
chaninfo->playback_vector.buf[0],
|
|
chaninfo->playback_vector.len[0] * sizeof (Sample));
|
|
|
|
/* Copy buf[1] from playback_buf */
|
|
memcpy (chaninfo->playback_wrap_buffer + chaninfo->playback_vector.len[0],
|
|
chaninfo->playback_vector.buf[1],
|
|
(necessary_samples - chaninfo->playback_vector.len[0])
|
|
* sizeof (Sample));
|
|
|
|
chaninfo->current_playback_buffer = chaninfo->playback_wrap_buffer;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (rec_nframes == 0 && _actual_speed != 1.0f && _actual_speed != -1.0f) {
|
|
|
|
interpolation.set_speed (_target_speed);
|
|
|
|
int channel = 0;
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan, ++channel) {
|
|
ChannelInfo* chaninfo (*chan);
|
|
|
|
playback_distance = interpolation.interpolate (
|
|
channel, nframes, chaninfo->current_playback_buffer, chaninfo->speed_buffer);
|
|
|
|
chaninfo->current_playback_buffer = chaninfo->speed_buffer;
|
|
}
|
|
|
|
} else {
|
|
playback_distance = nframes;
|
|
}
|
|
|
|
_speed = _target_speed;
|
|
}
|
|
|
|
if (need_disk_signal) {
|
|
|
|
/* copy data over to buffer set */
|
|
|
|
size_t n_buffers = bufs.count().n_audio();
|
|
size_t n_chans = c->size();
|
|
gain_t scaling = 1.0f;
|
|
|
|
if (n_chans > n_buffers) {
|
|
scaling = ((float) n_buffers)/n_chans;
|
|
}
|
|
|
|
for (n = 0, chan = c->begin(); chan != c->end(); ++chan, ++n) {
|
|
|
|
AudioBuffer& buf (bufs.get_audio (n%n_buffers));
|
|
ChannelInfo* chaninfo (*chan);
|
|
|
|
if (n < n_chans) {
|
|
if (scaling != 1.0f) {
|
|
buf.read_from_with_gain (chaninfo->current_playback_buffer, nframes, scaling);
|
|
} else {
|
|
buf.read_from (chaninfo->current_playback_buffer, nframes);
|
|
}
|
|
} else {
|
|
if (scaling != 1.0f) {
|
|
buf.accumulate_with_gain_from (chaninfo->current_playback_buffer, nframes, scaling);
|
|
} else {
|
|
buf.accumulate_from (chaninfo->current_playback_buffer, nframes);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* leave the MIDI count alone */
|
|
ChanCount cnt (DataType::AUDIO, n_chans);
|
|
cnt.set (DataType::MIDI, bufs.count().n_midi());
|
|
bufs.set_count (cnt);
|
|
|
|
/* extra buffers will already be silent, so leave them alone */
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
frameoffset_t
|
|
AudioDiskstream::calculate_playback_distance (pframes_t nframes)
|
|
{
|
|
frameoffset_t playback_distance = nframes;
|
|
|
|
if (record_enabled()) {
|
|
playback_distance = nframes;
|
|
} else if (_actual_speed != 1.0f && _actual_speed != -1.0f) {
|
|
interpolation.set_speed (_target_speed);
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
int channel = 0;
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan, ++channel) {
|
|
playback_distance = interpolation.interpolate (channel, nframes, NULL, NULL);
|
|
}
|
|
} else {
|
|
playback_distance = nframes;
|
|
}
|
|
|
|
if (_actual_speed < 0.0) {
|
|
return -playback_distance;
|
|
} else {
|
|
return playback_distance;
|
|
}
|
|
}
|
|
|
|
/** Update various things including playback_sample, read pointer on each channel's playback_buf
|
|
* and write pointer on each channel's capture_buf. Also wout whether the butler is needed.
|
|
* @return true if the butler is required.
|
|
*/
|
|
bool
|
|
AudioDiskstream::commit (framecnt_t playback_distance)
|
|
{
|
|
bool need_butler = false;
|
|
|
|
if (!_io || !_io->active()) {
|
|
return false;
|
|
}
|
|
|
|
if (_actual_speed < 0.0) {
|
|
playback_sample -= playback_distance;
|
|
} else {
|
|
playback_sample += playback_distance;
|
|
}
|
|
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
|
|
(*chan)->playback_buf->increment_read_ptr (playback_distance);
|
|
|
|
if (adjust_capture_position) {
|
|
(*chan)->capture_buf->increment_write_ptr (adjust_capture_position);
|
|
}
|
|
}
|
|
|
|
if (adjust_capture_position != 0) {
|
|
capture_captured += adjust_capture_position;
|
|
DEBUG_TRACE (DEBUG::CaptureAlignment, string_compose ("%1 now captured %2 (by %3)\n", name(), capture_captured, adjust_capture_position));
|
|
adjust_capture_position = 0;
|
|
}
|
|
|
|
if (c->empty()) {
|
|
return false;
|
|
}
|
|
|
|
if (_slaved) {
|
|
if (_io && _io->active()) {
|
|
need_butler = c->front()->playback_buf->write_space() >= c->front()->playback_buf->bufsize() / 2;
|
|
} else {
|
|
need_butler = false;
|
|
}
|
|
} else {
|
|
if (_io && _io->active()) {
|
|
need_butler = ((framecnt_t) c->front()->playback_buf->write_space() >= disk_read_chunk_frames)
|
|
|| ((framecnt_t) c->front()->capture_buf->read_space() >= disk_write_chunk_frames);
|
|
} else {
|
|
need_butler = ((framecnt_t) c->front()->capture_buf->read_space() >= disk_write_chunk_frames);
|
|
}
|
|
}
|
|
|
|
return need_butler;
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::set_pending_overwrite (bool yn)
|
|
{
|
|
/* called from audio thread, so we can use the read ptr and playback sample as we wish */
|
|
|
|
_pending_overwrite = yn;
|
|
|
|
overwrite_frame = playback_sample;
|
|
|
|
boost::shared_ptr<ChannelList> c = channels.reader ();
|
|
if (!c->empty ()) {
|
|
overwrite_offset = c->front()->playback_buf->get_read_ptr();
|
|
}
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::overwrite_existing_buffers ()
|
|
{
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
if (c->empty ()) {
|
|
_pending_overwrite = false;
|
|
return 0;
|
|
}
|
|
|
|
Sample* mixdown_buffer;
|
|
float* gain_buffer;
|
|
int ret = -1;
|
|
bool reversed = (_visible_speed * _session.transport_speed()) < 0.0f;
|
|
|
|
overwrite_queued = false;
|
|
|
|
/* assume all are the same size */
|
|
framecnt_t size = c->front()->playback_buf->bufsize();
|
|
|
|
mixdown_buffer = new Sample[size];
|
|
gain_buffer = new float[size];
|
|
|
|
/* reduce size so that we can fill the buffer correctly (ringbuffers
|
|
can only handle size-1, otherwise they appear to be empty)
|
|
*/
|
|
size--;
|
|
|
|
uint32_t n=0;
|
|
framepos_t start;
|
|
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan, ++n) {
|
|
|
|
start = overwrite_frame;
|
|
framecnt_t cnt = size;
|
|
|
|
/* to fill the buffer without resetting the playback sample, we need to
|
|
do it one or two chunks (normally two).
|
|
|
|
|----------------------------------------------------------------------|
|
|
|
|
^
|
|
overwrite_offset
|
|
|<- second chunk->||<----------------- first chunk ------------------>|
|
|
|
|
*/
|
|
|
|
framecnt_t to_read = size - overwrite_offset;
|
|
|
|
if (read ((*chan)->playback_buf->buffer() + overwrite_offset, mixdown_buffer, gain_buffer, start, to_read, n, reversed)) {
|
|
error << string_compose(_("AudioDiskstream %1: when refilling, cannot read %2 from playlist at frame %3"),
|
|
id(), size, playback_sample) << endmsg;
|
|
goto out;
|
|
}
|
|
|
|
if (cnt > to_read) {
|
|
|
|
cnt -= to_read;
|
|
|
|
if (read ((*chan)->playback_buf->buffer(), mixdown_buffer, gain_buffer, start, cnt, n, reversed)) {
|
|
error << string_compose(_("AudioDiskstream %1: when refilling, cannot read %2 from playlist at frame %3"),
|
|
id(), size, playback_sample) << endmsg;
|
|
goto out;
|
|
}
|
|
}
|
|
}
|
|
|
|
ret = 0;
|
|
|
|
out:
|
|
_pending_overwrite = false;
|
|
delete [] gain_buffer;
|
|
delete [] mixdown_buffer;
|
|
return ret;
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::seek (framepos_t frame, bool complete_refill)
|
|
{
|
|
uint32_t n;
|
|
int ret = -1;
|
|
ChannelList::iterator chan;
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
Glib::Threads::Mutex::Lock lm (state_lock);
|
|
|
|
for (n = 0, chan = c->begin(); chan != c->end(); ++chan, ++n) {
|
|
(*chan)->playback_buf->reset ();
|
|
(*chan)->capture_buf->reset ();
|
|
}
|
|
|
|
/* can't rec-enable in destructive mode if transport is before start */
|
|
|
|
if (destructive() && record_enabled() && frame < _session.current_start_frame()) {
|
|
disengage_record_enable ();
|
|
}
|
|
|
|
playback_sample = frame;
|
|
file_frame = frame;
|
|
|
|
if (complete_refill) {
|
|
/* call _do_refill() to refill the entire buffer, using
|
|
the largest reads possible.
|
|
*/
|
|
while ((ret = do_refill_with_alloc (false)) > 0) ;
|
|
} else {
|
|
/* call _do_refill() to refill just one chunk, and then
|
|
return.
|
|
*/
|
|
ret = do_refill_with_alloc (true);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::can_internal_playback_seek (framecnt_t distance)
|
|
{
|
|
ChannelList::iterator chan;
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
for (chan = c->begin(); chan != c->end(); ++chan) {
|
|
if ((*chan)->playback_buf->read_space() < (size_t) distance) {
|
|
return false;
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::internal_playback_seek (framecnt_t distance)
|
|
{
|
|
ChannelList::iterator chan;
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
for (chan = c->begin(); chan != c->end(); ++chan) {
|
|
(*chan)->playback_buf->increment_read_ptr (::llabs(distance));
|
|
}
|
|
|
|
if (first_recordable_frame < max_framepos) {
|
|
first_recordable_frame += distance;
|
|
}
|
|
playback_sample += distance;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/** Read some data for 1 channel from our playlist into a buffer.
|
|
* @param buf Buffer to write to.
|
|
* @param start Session frame to start reading from; updated to where we end up
|
|
* after the read.
|
|
* @param cnt Count of samples to read.
|
|
* @param reversed true if we are running backwards, otherwise false.
|
|
*/
|
|
int
|
|
AudioDiskstream::read (Sample* buf, Sample* mixdown_buffer, float* gain_buffer,
|
|
framepos_t& start, framecnt_t cnt,
|
|
int channel, bool reversed)
|
|
{
|
|
framecnt_t this_read = 0;
|
|
bool reloop = false;
|
|
framepos_t loop_end = 0;
|
|
framepos_t loop_start = 0;
|
|
framecnt_t offset = 0;
|
|
Location *loc = 0;
|
|
|
|
/* XXX we don't currently play loops in reverse. not sure why */
|
|
|
|
if (!reversed) {
|
|
|
|
framecnt_t loop_length = 0;
|
|
|
|
/* Make the use of a Location atomic for this read operation.
|
|
|
|
Note: Locations don't get deleted, so all we care about
|
|
when I say "atomic" is that we are always pointing to
|
|
the same one and using a start/length values obtained
|
|
just once.
|
|
*/
|
|
|
|
if ((loc = loop_location) != 0) {
|
|
loop_start = loc->start();
|
|
loop_end = loc->end();
|
|
loop_length = loop_end - loop_start;
|
|
}
|
|
|
|
/* if we are looping, ensure that the first frame we read is at the correct
|
|
position within the loop.
|
|
*/
|
|
|
|
if (loc && start >= loop_end) {
|
|
start = loop_start + ((start - loop_start) % loop_length);
|
|
}
|
|
|
|
}
|
|
|
|
if (reversed) {
|
|
start -= cnt;
|
|
}
|
|
|
|
/* We need this while loop in case we hit a loop boundary, in which case our read from
|
|
the playlist must be split into more than one section.
|
|
*/
|
|
|
|
while (cnt) {
|
|
|
|
/* take any loop into account. we can't read past the end of the loop. */
|
|
|
|
if (loc && (loop_end - start < cnt)) {
|
|
this_read = loop_end - start;
|
|
reloop = true;
|
|
} else {
|
|
reloop = false;
|
|
this_read = cnt;
|
|
}
|
|
|
|
if (this_read == 0) {
|
|
break;
|
|
}
|
|
|
|
this_read = min(cnt,this_read);
|
|
|
|
if (audio_playlist()->read (buf+offset, mixdown_buffer, gain_buffer, start, this_read, channel) != this_read) {
|
|
error << string_compose(_("AudioDiskstream %1: cannot read %2 from playlist at frame %3"), id(), this_read,
|
|
start) << endmsg;
|
|
return -1;
|
|
}
|
|
|
|
if (reversed) {
|
|
|
|
swap_by_ptr (buf, buf + this_read - 1);
|
|
|
|
} else {
|
|
|
|
/* if we read to the end of the loop, go back to the beginning */
|
|
|
|
if (reloop) {
|
|
start = loop_start;
|
|
} else {
|
|
start += this_read;
|
|
}
|
|
}
|
|
|
|
cnt -= this_read;
|
|
offset += this_read;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::_do_refill_with_alloc (bool partial_fill)
|
|
{
|
|
/* We limit disk reads to at most 4MB chunks, which with floating point
|
|
samples would be 1M samples. But we might use 16 or 14 bit samples,
|
|
in which case 4MB is more samples than that. Therefore size this for
|
|
the smallest sample value .. 4MB = 2M samples (16 bit).
|
|
*/
|
|
|
|
Sample* mix_buf = new Sample[2*1048576];
|
|
float* gain_buf = new float[2*1048576];
|
|
|
|
int ret = _do_refill (mix_buf, gain_buf, (partial_fill ? disk_read_chunk_frames : 0));
|
|
|
|
delete [] mix_buf;
|
|
delete [] gain_buf;
|
|
|
|
return ret;
|
|
}
|
|
|
|
/** Get some more data from disk and put it in our channels' playback_bufs,
|
|
* if there is suitable space in them.
|
|
*
|
|
* If fill_level is non-zero, then we will refill the buffer so that there is
|
|
* still at least fill_level samples of space left to be filled. This is used
|
|
* after locates so that we do not need to wait to fill the entire buffer.
|
|
*
|
|
*/
|
|
|
|
int
|
|
AudioDiskstream::_do_refill (Sample* mixdown_buffer, float* gain_buffer, framecnt_t fill_level)
|
|
{
|
|
int32_t ret = 0;
|
|
framecnt_t to_read;
|
|
RingBufferNPT<Sample>::rw_vector vector;
|
|
bool const reversed = (_visible_speed * _session.transport_speed()) < 0.0f;
|
|
framecnt_t total_space;
|
|
framecnt_t zero_fill;
|
|
uint32_t chan_n;
|
|
ChannelList::iterator i;
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
framecnt_t ts;
|
|
|
|
/* do not read from disk while session is marked as Loading, to avoid
|
|
useless redundant I/O.
|
|
*/
|
|
|
|
if (_session.state_of_the_state() & Session::Loading) {
|
|
return 0;
|
|
}
|
|
|
|
if (c->empty()) {
|
|
return 0;
|
|
}
|
|
|
|
assert(mixdown_buffer);
|
|
assert(gain_buffer);
|
|
|
|
vector.buf[0] = 0;
|
|
vector.len[0] = 0;
|
|
vector.buf[1] = 0;
|
|
vector.len[1] = 0;
|
|
|
|
c->front()->playback_buf->get_write_vector (&vector);
|
|
|
|
if ((total_space = vector.len[0] + vector.len[1]) == 0) {
|
|
/* nowhere to write to */
|
|
return 0;
|
|
}
|
|
|
|
if (fill_level) {
|
|
if (fill_level < total_space) {
|
|
total_space -= fill_level;
|
|
} else {
|
|
/* we can't do anything with it */
|
|
fill_level = 0;
|
|
}
|
|
}
|
|
|
|
/* if we're running close to normal speed and there isn't enough
|
|
space to do disk_read_chunk_frames of I/O, then don't bother.
|
|
|
|
at higher speeds, just do it because the sync between butler
|
|
and audio thread may not be good enough.
|
|
|
|
Note: it is a design assumption that disk_read_chunk_frames is smaller
|
|
than the playback buffer size, so this check should never trip when
|
|
the playback buffer is empty.
|
|
*/
|
|
|
|
if ((total_space < disk_read_chunk_frames) && fabs (_actual_speed) < 2.0f) {
|
|
return 0;
|
|
}
|
|
|
|
/* when slaved, don't try to get too close to the read pointer. this
|
|
leaves space for the buffer reversal to have something useful to
|
|
work with.
|
|
*/
|
|
|
|
if (_slaved && total_space < (framecnt_t) (c->front()->playback_buf->bufsize() / 2)) {
|
|
return 0;
|
|
}
|
|
|
|
if (reversed) {
|
|
|
|
if (file_frame == 0) {
|
|
|
|
/* at start: nothing to do but fill with silence */
|
|
|
|
for (chan_n = 0, i = c->begin(); i != c->end(); ++i, ++chan_n) {
|
|
|
|
ChannelInfo* chan (*i);
|
|
chan->playback_buf->get_write_vector (&vector);
|
|
memset (vector.buf[0], 0, sizeof(Sample) * vector.len[0]);
|
|
if (vector.len[1]) {
|
|
memset (vector.buf[1], 0, sizeof(Sample) * vector.len[1]);
|
|
}
|
|
chan->playback_buf->increment_write_ptr (vector.len[0] + vector.len[1]);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
if (file_frame < total_space) {
|
|
|
|
/* too close to the start: read what we can,
|
|
and then zero fill the rest
|
|
*/
|
|
|
|
zero_fill = total_space - file_frame;
|
|
total_space = file_frame;
|
|
|
|
} else {
|
|
|
|
zero_fill = 0;
|
|
}
|
|
|
|
} else {
|
|
|
|
if (file_frame == max_framepos) {
|
|
|
|
/* at end: nothing to do but fill with silence */
|
|
|
|
for (chan_n = 0, i = c->begin(); i != c->end(); ++i, ++chan_n) {
|
|
|
|
ChannelInfo* chan (*i);
|
|
chan->playback_buf->get_write_vector (&vector);
|
|
memset (vector.buf[0], 0, sizeof(Sample) * vector.len[0]);
|
|
if (vector.len[1]) {
|
|
memset (vector.buf[1], 0, sizeof(Sample) * vector.len[1]);
|
|
}
|
|
chan->playback_buf->increment_write_ptr (vector.len[0] + vector.len[1]);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
if (file_frame > max_framepos - total_space) {
|
|
|
|
/* to close to the end: read what we can, and zero fill the rest */
|
|
|
|
zero_fill = total_space - (max_framepos - file_frame);
|
|
total_space = max_framepos - file_frame;
|
|
|
|
} else {
|
|
zero_fill = 0;
|
|
}
|
|
}
|
|
|
|
framepos_t file_frame_tmp = 0;
|
|
|
|
/* total_space is in samples. We want to optimize read sizes in various sizes using bytes */
|
|
|
|
const size_t bits_per_sample = format_data_width (_session.config.get_native_file_data_format());
|
|
size_t total_bytes = total_space * bits_per_sample / 8;
|
|
|
|
/* chunk size range is 256kB to 4MB. Bigger is faster in terms of MB/sec, but bigger chunk size always takes longer
|
|
*/
|
|
size_t byte_size_for_read = max ((size_t) (256 * 1024), min ((size_t) (4 * 1048576), total_bytes));
|
|
|
|
/* find nearest (lower) multiple of 16384 */
|
|
|
|
byte_size_for_read = (byte_size_for_read / 16384) * 16384;
|
|
|
|
/* now back to samples */
|
|
|
|
framecnt_t samples_to_read = byte_size_for_read / (bits_per_sample / 8);
|
|
|
|
//cerr << name() << " will read " << byte_size_for_read << " out of total bytes " << total_bytes << " in buffer of "
|
|
// << c->front()->playback_buf->bufsize() * bits_per_sample / 8 << " bps = " << bits_per_sample << endl;
|
|
// cerr << name () << " read samples = " << samples_to_read << " out of total space " << total_space << " in buffer of " << c->front()->playback_buf->bufsize() << " samples\n";
|
|
|
|
// uint64_t before = g_get_monotonic_time ();
|
|
// uint64_t elapsed;
|
|
|
|
for (chan_n = 0, i = c->begin(); i != c->end(); ++i, ++chan_n) {
|
|
|
|
ChannelInfo* chan (*i);
|
|
Sample* buf1;
|
|
Sample* buf2;
|
|
framecnt_t len1, len2;
|
|
|
|
chan->playback_buf->get_write_vector (&vector);
|
|
|
|
if ((framecnt_t) vector.len[0] > samples_to_read) {
|
|
|
|
/* we're not going to fill the first chunk, so certainly do not bother with the
|
|
other part. it won't be connected with the part we do fill, as in:
|
|
|
|
.... => writable space
|
|
++++ => readable space
|
|
^^^^ => 1 x disk_read_chunk_frames that would be filled
|
|
|
|
|......|+++++++++++++|...............................|
|
|
buf1 buf0
|
|
^^^^^^^^^^^^^^^
|
|
|
|
|
|
So, just pretend that the buf1 part isn't there.
|
|
|
|
*/
|
|
|
|
vector.buf[1] = 0;
|
|
vector.len[1] = 0;
|
|
|
|
}
|
|
|
|
ts = total_space;
|
|
file_frame_tmp = file_frame;
|
|
|
|
buf1 = vector.buf[0];
|
|
len1 = vector.len[0];
|
|
buf2 = vector.buf[1];
|
|
len2 = vector.len[1];
|
|
|
|
to_read = min (ts, len1);
|
|
to_read = min (to_read, (framecnt_t) samples_to_read);
|
|
|
|
assert (to_read >= 0);
|
|
|
|
if (to_read) {
|
|
|
|
if (read (buf1, mixdown_buffer, gain_buffer, file_frame_tmp, to_read, chan_n, reversed)) {
|
|
ret = -1;
|
|
goto out;
|
|
}
|
|
|
|
chan->playback_buf->increment_write_ptr (to_read);
|
|
ts -= to_read;
|
|
}
|
|
|
|
to_read = min (ts, len2);
|
|
|
|
if (to_read) {
|
|
|
|
/* we read all of vector.len[0], but it wasn't the
|
|
entire samples_to_read of data, so read some or
|
|
all of vector.len[1] as well.
|
|
*/
|
|
|
|
if (read (buf2, mixdown_buffer, gain_buffer, file_frame_tmp, to_read, chan_n, reversed)) {
|
|
ret = -1;
|
|
goto out;
|
|
}
|
|
|
|
chan->playback_buf->increment_write_ptr (to_read);
|
|
}
|
|
|
|
if (zero_fill) {
|
|
/* XXX: do something */
|
|
}
|
|
|
|
}
|
|
|
|
// elapsed = g_get_monotonic_time () - before;
|
|
// cerr << "\tbandwidth = " << (byte_size_for_read / 1048576.0) / (elapsed/1000000.0) << "MB/sec\n";
|
|
|
|
file_frame = file_frame_tmp;
|
|
assert (file_frame >= 0);
|
|
|
|
ret = ((total_space - samples_to_read) > disk_read_chunk_frames);
|
|
|
|
c->front()->playback_buf->get_write_vector (&vector);
|
|
|
|
out:
|
|
return ret;
|
|
}
|
|
|
|
/** Flush pending data to disk.
|
|
*
|
|
* Important note: this function will write *AT MOST* disk_write_chunk_frames
|
|
* of data to disk. it will never write more than that. If it writes that
|
|
* much and there is more than that waiting to be written, it will return 1,
|
|
* otherwise 0 on success or -1 on failure.
|
|
*
|
|
* If there is less than disk_write_chunk_frames to be written, no data will be
|
|
* written at all unless @a force_flush is true.
|
|
*/
|
|
int
|
|
AudioDiskstream::do_flush (RunContext /*context*/, bool force_flush)
|
|
{
|
|
uint32_t to_write;
|
|
int32_t ret = 0;
|
|
RingBufferNPT<Sample>::rw_vector vector;
|
|
RingBufferNPT<CaptureTransition>::rw_vector transvec;
|
|
framecnt_t total;
|
|
|
|
transvec.buf[0] = 0;
|
|
transvec.buf[1] = 0;
|
|
vector.buf[0] = 0;
|
|
vector.buf[1] = 0;
|
|
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
|
|
(*chan)->capture_buf->get_read_vector (&vector);
|
|
|
|
total = vector.len[0] + vector.len[1];
|
|
|
|
if (total == 0 || (total < disk_write_chunk_frames && !force_flush && was_recording)) {
|
|
goto out;
|
|
}
|
|
|
|
/* if there are 2+ chunks of disk i/o possible for
|
|
this track, let the caller know so that it can arrange
|
|
for us to be called again, ASAP.
|
|
|
|
if we are forcing a flush, then if there is* any* extra
|
|
work, let the caller know.
|
|
|
|
if we are no longer recording and there is any extra work,
|
|
let the caller know too.
|
|
*/
|
|
|
|
if (total >= 2 * disk_write_chunk_frames || ((force_flush || !was_recording) && total > disk_write_chunk_frames)) {
|
|
ret = 1;
|
|
}
|
|
|
|
to_write = min (disk_write_chunk_frames, (framecnt_t) vector.len[0]);
|
|
|
|
// check the transition buffer when recording destructive
|
|
// important that we get this after the capture buf
|
|
|
|
if (destructive()) {
|
|
(*chan)->capture_transition_buf->get_read_vector(&transvec);
|
|
size_t transcount = transvec.len[0] + transvec.len[1];
|
|
size_t ti;
|
|
|
|
for (ti=0; ti < transcount; ++ti) {
|
|
CaptureTransition & captrans = (ti < transvec.len[0]) ? transvec.buf[0][ti] : transvec.buf[1][ti-transvec.len[0]];
|
|
|
|
if (captrans.type == CaptureStart) {
|
|
// by definition, the first data we got above represents the given capture pos
|
|
|
|
(*chan)->write_source->mark_capture_start (captrans.capture_val);
|
|
(*chan)->curr_capture_cnt = 0;
|
|
|
|
} else if (captrans.type == CaptureEnd) {
|
|
|
|
// capture end, the capture_val represents total frames in capture
|
|
|
|
if (captrans.capture_val <= (*chan)->curr_capture_cnt + to_write) {
|
|
|
|
// shorten to make the write a perfect fit
|
|
uint32_t nto_write = (captrans.capture_val - (*chan)->curr_capture_cnt);
|
|
|
|
if (nto_write < to_write) {
|
|
ret = 1; // should we?
|
|
}
|
|
to_write = nto_write;
|
|
|
|
(*chan)->write_source->mark_capture_end ();
|
|
|
|
// increment past this transition, but go no further
|
|
++ti;
|
|
break;
|
|
}
|
|
else {
|
|
// actually ends just beyond this chunk, so force more work
|
|
ret = 1;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (ti > 0) {
|
|
(*chan)->capture_transition_buf->increment_read_ptr(ti);
|
|
}
|
|
}
|
|
|
|
if ((!(*chan)->write_source) || (*chan)->write_source->write (vector.buf[0], to_write) != to_write) {
|
|
error << string_compose(_("AudioDiskstream %1: cannot write to disk"), id()) << endmsg;
|
|
return -1;
|
|
}
|
|
|
|
(*chan)->capture_buf->increment_read_ptr (to_write);
|
|
(*chan)->curr_capture_cnt += to_write;
|
|
|
|
if ((to_write == vector.len[0]) && (total > to_write) && (to_write < disk_write_chunk_frames) && !destructive()) {
|
|
|
|
/* we wrote all of vector.len[0] but it wasn't an entire
|
|
disk_write_chunk_frames of data, so arrange for some part
|
|
of vector.len[1] to be flushed to disk as well.
|
|
*/
|
|
|
|
to_write = min ((framecnt_t)(disk_write_chunk_frames - to_write), (framecnt_t) vector.len[1]);
|
|
|
|
DEBUG_TRACE (DEBUG::Butler, string_compose ("%1 additional write of %2\n", name(), to_write));
|
|
|
|
if ((*chan)->write_source->write (vector.buf[1], to_write) != to_write) {
|
|
error << string_compose(_("AudioDiskstream %1: cannot write to disk"), id()) << endmsg;
|
|
return -1;
|
|
}
|
|
|
|
(*chan)->capture_buf->increment_read_ptr (to_write);
|
|
(*chan)->curr_capture_cnt += to_write;
|
|
}
|
|
}
|
|
|
|
out:
|
|
return ret;
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::transport_stopped_wallclock (struct tm& when, time_t twhen, bool abort_capture)
|
|
{
|
|
uint32_t buffer_position;
|
|
bool more_work = true;
|
|
int err = 0;
|
|
boost::shared_ptr<AudioRegion> region;
|
|
framecnt_t total_capture;
|
|
SourceList srcs;
|
|
SourceList::iterator src;
|
|
ChannelList::iterator chan;
|
|
vector<CaptureInfo*>::iterator ci;
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
uint32_t n = 0;
|
|
bool mark_write_completed = false;
|
|
|
|
finish_capture (c);
|
|
|
|
/* butler is already stopped, but there may be work to do
|
|
to flush remaining data to disk.
|
|
*/
|
|
|
|
while (more_work && !err) {
|
|
switch (do_flush (TransportContext, true)) {
|
|
case 0:
|
|
more_work = false;
|
|
break;
|
|
case 1:
|
|
break;
|
|
case -1:
|
|
error << string_compose(_("AudioDiskstream \"%1\": cannot flush captured data to disk!"), _name) << endmsg;
|
|
err++;
|
|
}
|
|
}
|
|
|
|
/* XXX is there anything we can do if err != 0 ? */
|
|
Glib::Threads::Mutex::Lock lm (capture_info_lock);
|
|
|
|
if (capture_info.empty()) {
|
|
return;
|
|
}
|
|
|
|
if (abort_capture) {
|
|
|
|
if (destructive()) {
|
|
goto outout;
|
|
}
|
|
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
|
|
if ((*chan)->write_source) {
|
|
|
|
(*chan)->write_source->mark_for_remove ();
|
|
(*chan)->write_source->drop_references ();
|
|
(*chan)->write_source.reset ();
|
|
}
|
|
|
|
/* new source set up in "out" below */
|
|
}
|
|
|
|
goto out;
|
|
}
|
|
|
|
for (total_capture = 0, ci = capture_info.begin(); ci != capture_info.end(); ++ci) {
|
|
total_capture += (*ci)->frames;
|
|
}
|
|
|
|
/* figure out the name for this take */
|
|
|
|
for (n = 0, chan = c->begin(); chan != c->end(); ++chan, ++n) {
|
|
|
|
boost::shared_ptr<AudioFileSource> s = (*chan)->write_source;
|
|
|
|
if (s) {
|
|
srcs.push_back (s);
|
|
s->update_header (capture_info.front()->start, when, twhen);
|
|
s->set_captured_for (_name.val());
|
|
s->mark_immutable ();
|
|
|
|
if (Config->get_auto_analyse_audio()) {
|
|
Analyser::queue_source_for_analysis (s, true);
|
|
}
|
|
|
|
DEBUG_TRACE (DEBUG::CaptureAlignment, string_compose ("newly captured source %1 length %2\n", s->path(), s->length (0)));
|
|
}
|
|
}
|
|
|
|
/* destructive tracks have a single, never changing region */
|
|
|
|
if (destructive()) {
|
|
|
|
/* send a signal that any UI can pick up to do the right thing. there is
|
|
a small problem here in that a UI may need the peak data to be ready
|
|
for the data that was recorded and this isn't interlocked with that
|
|
process. this problem is deferred to the UI.
|
|
*/
|
|
|
|
_playlist->LayeringChanged(); // XXX this may not get the UI to do the right thing
|
|
|
|
} else {
|
|
|
|
string whole_file_region_name;
|
|
whole_file_region_name = region_name_from_path (c->front()->write_source->name(), true);
|
|
|
|
/* Register a new region with the Session that
|
|
describes the entire source. Do this first
|
|
so that any sub-regions will obviously be
|
|
children of this one (later!)
|
|
*/
|
|
|
|
try {
|
|
PropertyList plist;
|
|
|
|
plist.add (Properties::start, c->front()->write_source->last_capture_start_frame());
|
|
plist.add (Properties::length, total_capture);
|
|
plist.add (Properties::name, whole_file_region_name);
|
|
boost::shared_ptr<Region> rx (RegionFactory::create (srcs, plist));
|
|
rx->set_automatic (true);
|
|
rx->set_whole_file (true);
|
|
|
|
region = boost::dynamic_pointer_cast<AudioRegion> (rx);
|
|
region->special_set_position (capture_info.front()->start);
|
|
}
|
|
|
|
|
|
catch (failed_constructor& err) {
|
|
error << string_compose(_("%1: could not create region for complete audio file"), _name) << endmsg;
|
|
/* XXX what now? */
|
|
}
|
|
|
|
_last_capture_sources.insert (_last_capture_sources.end(), srcs.begin(), srcs.end());
|
|
|
|
_playlist->clear_changes ();
|
|
_playlist->set_capture_insertion_in_progress (true);
|
|
_playlist->freeze ();
|
|
|
|
const framepos_t preroll_off = _session.preroll_record_trim_len ();
|
|
for (buffer_position = c->front()->write_source->last_capture_start_frame(), ci = capture_info.begin(); ci != capture_info.end(); ++ci) {
|
|
|
|
string region_name;
|
|
|
|
RegionFactory::region_name (region_name, whole_file_region_name, false);
|
|
|
|
DEBUG_TRACE (DEBUG::CaptureAlignment, string_compose ("%1 capture bufpos %5 start @ %2 length %3 add new region %4\n",
|
|
_name, (*ci)->start, (*ci)->frames, region_name, buffer_position));
|
|
|
|
try {
|
|
|
|
PropertyList plist;
|
|
|
|
plist.add (Properties::start, buffer_position);
|
|
plist.add (Properties::length, (*ci)->frames);
|
|
plist.add (Properties::name, region_name);
|
|
|
|
boost::shared_ptr<Region> rx (RegionFactory::create (srcs, plist));
|
|
region = boost::dynamic_pointer_cast<AudioRegion> (rx);
|
|
if (preroll_off > 0) {
|
|
region->trim_front (buffer_position + preroll_off);
|
|
}
|
|
}
|
|
|
|
catch (failed_constructor& err) {
|
|
error << _("AudioDiskstream: could not create region for captured audio!") << endmsg;
|
|
continue; /* XXX is this OK? */
|
|
}
|
|
|
|
i_am_the_modifier++;
|
|
|
|
_playlist->add_region (region, (*ci)->start + preroll_off, 1, non_layered());
|
|
_playlist->set_layer (region, DBL_MAX);
|
|
i_am_the_modifier--;
|
|
|
|
buffer_position += (*ci)->frames;
|
|
}
|
|
|
|
_playlist->thaw ();
|
|
_playlist->set_capture_insertion_in_progress (false);
|
|
_session.add_command (new StatefulDiffCommand (_playlist));
|
|
}
|
|
|
|
mark_write_completed = true;
|
|
|
|
out:
|
|
reset_write_sources (mark_write_completed);
|
|
|
|
outout:
|
|
|
|
for (ci = capture_info.begin(); ci != capture_info.end(); ++ci) {
|
|
delete *ci;
|
|
}
|
|
|
|
capture_info.clear ();
|
|
capture_start_frame = 0;
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::transport_looped (framepos_t transport_frame)
|
|
{
|
|
if (was_recording) {
|
|
// all we need to do is finish this capture, with modified capture length
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
finish_capture (c);
|
|
|
|
// the next region will start recording via the normal mechanism
|
|
// we'll set the start position to the current transport pos
|
|
// no latency adjustment or capture offset needs to be made, as that already happened the first time
|
|
capture_start_frame = transport_frame;
|
|
first_recordable_frame = transport_frame; // mild lie
|
|
last_recordable_frame = max_framepos;
|
|
was_recording = true;
|
|
|
|
if (recordable() && destructive()) {
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
|
|
RingBufferNPT<CaptureTransition>::rw_vector transvec;
|
|
(*chan)->capture_transition_buf->get_write_vector(&transvec);
|
|
|
|
if (transvec.len[0] > 0) {
|
|
transvec.buf[0]->type = CaptureStart;
|
|
transvec.buf[0]->capture_val = capture_start_frame;
|
|
(*chan)->capture_transition_buf->increment_write_ptr(1);
|
|
}
|
|
else {
|
|
// bad!
|
|
fatal << X_("programming error: capture_transition_buf is full on rec loop! inconceivable!")
|
|
<< endmsg;
|
|
}
|
|
}
|
|
}
|
|
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::finish_capture (boost::shared_ptr<ChannelList> c)
|
|
{
|
|
was_recording = false;
|
|
first_recordable_frame = max_framepos;
|
|
last_recordable_frame = max_framepos;
|
|
|
|
if (capture_captured == 0) {
|
|
return;
|
|
}
|
|
|
|
if (recordable() && destructive()) {
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
|
|
RingBufferNPT<CaptureTransition>::rw_vector transvec;
|
|
(*chan)->capture_transition_buf->get_write_vector(&transvec);
|
|
|
|
if (transvec.len[0] > 0) {
|
|
transvec.buf[0]->type = CaptureEnd;
|
|
transvec.buf[0]->capture_val = capture_captured;
|
|
(*chan)->capture_transition_buf->increment_write_ptr(1);
|
|
}
|
|
else {
|
|
// bad!
|
|
fatal << string_compose (_("programmer error: %1"), X_("capture_transition_buf is full when stopping record! inconceivable!")) << endmsg;
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
CaptureInfo* ci = new CaptureInfo;
|
|
|
|
ci->start = capture_start_frame;
|
|
ci->frames = capture_captured;
|
|
|
|
/* XXX theoretical race condition here. Need atomic exchange ?
|
|
However, the circumstances when this is called right
|
|
now (either on record-disable or transport_stopped)
|
|
mean that no actual race exists. I think ...
|
|
We now have a capture_info_lock, but it is only to be used
|
|
to synchronize in the transport_stop and the capture info
|
|
accessors, so that invalidation will not occur (both non-realtime).
|
|
*/
|
|
|
|
DEBUG_TRACE (DEBUG::CaptureAlignment, string_compose ("Finish capture, add new CI, %1 + %2\n", ci->start, ci->frames));
|
|
|
|
capture_info.push_back (ci);
|
|
capture_captured = 0;
|
|
|
|
/* now we've finished a capture, reset first_recordable_frame for next time */
|
|
first_recordable_frame = max_framepos;
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::set_record_enabled (bool yn)
|
|
{
|
|
if (!recordable() || !_session.record_enabling_legal() || _io->n_ports().n_audio() == 0 || record_safe ()) {
|
|
return;
|
|
}
|
|
|
|
/* can't rec-enable in destructive mode if transport is before start */
|
|
|
|
if (destructive() && yn && _session.transport_frame() < _session.current_start_frame()) {
|
|
return;
|
|
}
|
|
|
|
/* yes, i know that this not proof against race conditions, but its
|
|
good enough. i think.
|
|
*/
|
|
|
|
if (record_enabled() != yn) {
|
|
if (yn) {
|
|
engage_record_enable ();
|
|
} else {
|
|
disengage_record_enable ();
|
|
}
|
|
|
|
RecordEnableChanged (); /* EMIT SIGNAL */
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::set_record_safe (bool yn)
|
|
{
|
|
if (!recordable() || !_session.record_enabling_legal() || _io->n_ports().n_audio() == 0) {
|
|
return;
|
|
}
|
|
|
|
/* can't rec-safe in destructive mode if transport is before start ????
|
|
REQUIRES REVIEW */
|
|
|
|
if (destructive() && yn && _session.transport_frame() < _session.current_start_frame()) {
|
|
return;
|
|
}
|
|
|
|
/* yes, i know that this not proof against race conditions, but its
|
|
good enough. i think.
|
|
*/
|
|
|
|
if (record_safe () != yn) {
|
|
if (yn) {
|
|
engage_record_safe ();
|
|
} else {
|
|
disengage_record_safe ();
|
|
}
|
|
|
|
RecordSafeChanged (); /* EMIT SIGNAL */
|
|
}
|
|
}
|
|
|
|
bool
|
|
AudioDiskstream::prep_record_enable ()
|
|
{
|
|
if (!recordable() || !_session.record_enabling_legal() || _io->n_ports().n_audio() == 0 || record_safe ()) { // REQUIRES REVIEW "|| record_safe ()"
|
|
return false;
|
|
}
|
|
|
|
/* can't rec-enable in destructive mode if transport is before start */
|
|
|
|
if (destructive() && _session.transport_frame() < _session.current_start_frame()) {
|
|
return false;
|
|
}
|
|
|
|
bool rolling = _session.transport_speed() != 0.0f;
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
capturing_sources.clear ();
|
|
|
|
if (Config->get_monitoring_model() == HardwareMonitoring) {
|
|
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
(*chan)->source.request_input_monitoring (!(_session.config.get_auto_input() && rolling));
|
|
capturing_sources.push_back ((*chan)->write_source);
|
|
Source::Lock lock((*chan)->write_source->mutex());
|
|
(*chan)->write_source->mark_streaming_write_started (lock);
|
|
}
|
|
|
|
} else {
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
capturing_sources.push_back ((*chan)->write_source);
|
|
Source::Lock lock((*chan)->write_source->mutex());
|
|
(*chan)->write_source->mark_streaming_write_started (lock);
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool
|
|
AudioDiskstream::prep_record_disable ()
|
|
{
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
if (Config->get_monitoring_model() == HardwareMonitoring) {
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
(*chan)->source.request_input_monitoring (false);
|
|
}
|
|
}
|
|
capturing_sources.clear ();
|
|
|
|
return true;
|
|
}
|
|
|
|
XMLNode&
|
|
AudioDiskstream::get_state ()
|
|
{
|
|
XMLNode& node (Diskstream::get_state());
|
|
LocaleGuard lg;
|
|
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
node.set_property ("channels", (uint32_t)c->size());
|
|
|
|
if (!capturing_sources.empty() && _session.get_record_enabled()) {
|
|
|
|
XMLNode* cs_child = new XMLNode (X_("CapturingSources"));
|
|
XMLNode* cs_grandchild;
|
|
|
|
for (vector<boost::shared_ptr<AudioFileSource> >::iterator i = capturing_sources.begin(); i != capturing_sources.end(); ++i) {
|
|
cs_grandchild = new XMLNode (X_("file"));
|
|
cs_grandchild->set_property (X_("path"), (*i)->path());
|
|
cs_child->add_child_nocopy (*cs_grandchild);
|
|
}
|
|
|
|
/* store the location where capture will start */
|
|
|
|
Location* pi;
|
|
|
|
if (_session.preroll_record_punch_enabled ()) {
|
|
cs_child->set_property (X_("at"), _session.preroll_record_punch_pos());
|
|
} else if (_session.config.get_punch_in() && ((pi = _session.locations()->auto_punch_location()) != 0)) {
|
|
cs_child->set_property (X_("at"), pi->start());
|
|
} else {
|
|
cs_child->set_property (X_("at"), _session.transport_frame());
|
|
}
|
|
|
|
node.add_child_nocopy (*cs_child);
|
|
}
|
|
|
|
return node;
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::set_state (const XMLNode& node, int version)
|
|
{
|
|
XMLNodeList nlist = node.children();
|
|
XMLNodeIterator niter;
|
|
XMLNode* capture_pending_node = 0;
|
|
LocaleGuard lg;
|
|
|
|
/* prevent write sources from being created */
|
|
|
|
in_set_state = true;
|
|
|
|
for (niter = nlist.begin(); niter != nlist.end(); ++niter) {
|
|
if ((*niter)->name() == IO::state_node_name) {
|
|
deprecated_io_node = new XMLNode (**niter);
|
|
}
|
|
|
|
if ((*niter)->name() == X_("CapturingSources")) {
|
|
capture_pending_node = *niter;
|
|
}
|
|
}
|
|
|
|
if (Diskstream::set_state (node, version)) {
|
|
return -1;
|
|
}
|
|
|
|
uint32_t nchans = 1;
|
|
node.get_property ("channels", nchans);
|
|
|
|
// create necessary extra channels
|
|
// we are always constructed with one and we always need one
|
|
|
|
_n_channels.set(DataType::AUDIO, channels.reader()->size());
|
|
|
|
if (nchans > _n_channels.n_audio()) {
|
|
|
|
add_channel (nchans - _n_channels.n_audio());
|
|
IO::PortCountChanged(_n_channels);
|
|
|
|
} else if (nchans < _n_channels.n_audio()) {
|
|
|
|
remove_channel (_n_channels.n_audio() - nchans);
|
|
}
|
|
|
|
if (!destructive() && capture_pending_node) {
|
|
/* destructive streams have one and only one source per channel,
|
|
and so they never end up in pending capture in any useful
|
|
sense.
|
|
*/
|
|
use_pending_capture_data (*capture_pending_node);
|
|
}
|
|
|
|
in_set_state = false;
|
|
|
|
/* make sure this is clear before we do anything else */
|
|
|
|
capturing_sources.clear ();
|
|
|
|
/* write sources are handled when we handle the input set
|
|
up of the IO that owns this DS (::non_realtime_input_change())
|
|
*/
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::use_new_write_source (uint32_t n)
|
|
{
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
if (!recordable()) {
|
|
return 1;
|
|
}
|
|
|
|
if (n >= c->size()) {
|
|
error << string_compose (_("AudioDiskstream: channel %1 out of range"), n) << endmsg;
|
|
return -1;
|
|
}
|
|
|
|
ChannelInfo* chan = (*c)[n];
|
|
|
|
try {
|
|
if ((chan->write_source = _session.create_audio_source_for_session (
|
|
n_channels().n_audio(), write_source_name(), n, destructive())) == 0) {
|
|
throw failed_constructor();
|
|
}
|
|
}
|
|
|
|
catch (failed_constructor &err) {
|
|
error << string_compose (_("%1:%2 new capture file not initialized correctly"), _name, n) << endmsg;
|
|
chan->write_source.reset ();
|
|
return -1;
|
|
}
|
|
|
|
/* do not remove destructive files even if they are empty */
|
|
|
|
chan->write_source->set_allow_remove_if_empty (!destructive());
|
|
|
|
return 0;
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::reset_write_sources (bool mark_write_complete, bool /*force*/)
|
|
{
|
|
ChannelList::iterator chan;
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
uint32_t n;
|
|
|
|
if (!_session.writable() || !recordable()) {
|
|
return;
|
|
}
|
|
|
|
capturing_sources.clear ();
|
|
|
|
for (chan = c->begin(), n = 0; chan != c->end(); ++chan, ++n) {
|
|
|
|
if (!destructive()) {
|
|
|
|
if ((*chan)->write_source) {
|
|
|
|
if (mark_write_complete) {
|
|
Source::Lock lock((*chan)->write_source->mutex());
|
|
(*chan)->write_source->mark_streaming_write_completed (lock);
|
|
(*chan)->write_source->done_with_peakfile_writes ();
|
|
}
|
|
|
|
if ((*chan)->write_source->removable()) {
|
|
(*chan)->write_source->mark_for_remove ();
|
|
(*chan)->write_source->drop_references ();
|
|
}
|
|
|
|
(*chan)->write_source.reset ();
|
|
}
|
|
|
|
use_new_write_source (n);
|
|
|
|
if (record_enabled()) {
|
|
capturing_sources.push_back ((*chan)->write_source);
|
|
}
|
|
|
|
} else {
|
|
|
|
if ((*chan)->write_source == 0) {
|
|
use_new_write_source (n);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (destructive() && !c->empty ()) {
|
|
|
|
/* we now have all our write sources set up, so create the
|
|
playlist's single region.
|
|
*/
|
|
|
|
if (_playlist->empty()) {
|
|
setup_destructive_playlist ();
|
|
}
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::set_block_size (pframes_t /*nframes*/)
|
|
{
|
|
if (_session.get_block_size() > speed_buffer_size) {
|
|
speed_buffer_size = _session.get_block_size();
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
if ((*chan)->speed_buffer)
|
|
delete [] (*chan)->speed_buffer;
|
|
(*chan)->speed_buffer = new Sample[speed_buffer_size];
|
|
}
|
|
}
|
|
allocate_temporary_buffers ();
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::allocate_temporary_buffers ()
|
|
{
|
|
/* make sure the wrap buffer is at least large enough to deal
|
|
with the speeds up to 1.2, to allow for micro-variation
|
|
when slaving to MTC, Timecode etc.
|
|
*/
|
|
|
|
double const sp = max (fabs (_actual_speed), 1.2);
|
|
framecnt_t required_wrap_size = (framecnt_t) ceil (_session.get_block_size() * sp) + 2;
|
|
|
|
if (required_wrap_size > wrap_buffer_size) {
|
|
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
if ((*chan)->playback_wrap_buffer) {
|
|
delete [] (*chan)->playback_wrap_buffer;
|
|
}
|
|
(*chan)->playback_wrap_buffer = new Sample[required_wrap_size];
|
|
if ((*chan)->capture_wrap_buffer) {
|
|
delete [] (*chan)->capture_wrap_buffer;
|
|
}
|
|
(*chan)->capture_wrap_buffer = new Sample[required_wrap_size];
|
|
}
|
|
|
|
wrap_buffer_size = required_wrap_size;
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::request_input_monitoring (bool yn)
|
|
{
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
(*chan)->source.request_input_monitoring (yn);
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::set_align_style_from_io ()
|
|
{
|
|
bool have_physical = false;
|
|
|
|
if (_alignment_choice != Automatic) {
|
|
return;
|
|
}
|
|
|
|
if (_io == 0) {
|
|
return;
|
|
}
|
|
|
|
get_input_sources ();
|
|
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
if ((*chan)->source.is_physical ()) {
|
|
have_physical = true;
|
|
break;
|
|
}
|
|
}
|
|
|
|
#ifdef MIXBUS
|
|
// compensate for latency when bouncing from master or mixbus.
|
|
// we need to use "ExistingMaterial" to pick up the master bus' latency
|
|
// see also Route::direct_feeds_according_to_reality
|
|
IOVector ios;
|
|
ios.push_back (_io);
|
|
if (_session.master_out() && ios.fed_by (_session.master_out()->output())) {
|
|
have_physical = true;
|
|
}
|
|
for (uint32_t n = 0; n < NUM_MIXBUSES && !have_physical; ++n) {
|
|
if (_session.get_mixbus (n) && ios.fed_by (_session.get_mixbus(n)->output())) {
|
|
have_physical = true;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
if (have_physical) {
|
|
set_align_style (ExistingMaterial);
|
|
} else {
|
|
set_align_style (CaptureTime);
|
|
}
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::add_channel_to (boost::shared_ptr<ChannelList> c, uint32_t how_many)
|
|
{
|
|
while (how_many--) {
|
|
c->push_back (new ChannelInfo(
|
|
_session.butler()->audio_diskstream_playback_buffer_size(),
|
|
_session.butler()->audio_diskstream_capture_buffer_size(),
|
|
speed_buffer_size, wrap_buffer_size));
|
|
interpolation.add_channel_to (
|
|
_session.butler()->audio_diskstream_playback_buffer_size(),
|
|
speed_buffer_size);
|
|
}
|
|
|
|
_n_channels.set(DataType::AUDIO, c->size());
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::add_channel (uint32_t how_many)
|
|
{
|
|
RCUWriter<ChannelList> writer (channels);
|
|
boost::shared_ptr<ChannelList> c = writer.get_copy();
|
|
|
|
return add_channel_to (c, how_many);
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::remove_channel_from (boost::shared_ptr<ChannelList> c, uint32_t how_many)
|
|
{
|
|
while (how_many-- && !c->empty()) {
|
|
delete c->back();
|
|
c->pop_back();
|
|
interpolation.remove_channel_from ();
|
|
}
|
|
|
|
_n_channels.set(DataType::AUDIO, c->size());
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::remove_channel (uint32_t how_many)
|
|
{
|
|
RCUWriter<ChannelList> writer (channels);
|
|
boost::shared_ptr<ChannelList> c = writer.get_copy();
|
|
|
|
return remove_channel_from (c, how_many);
|
|
}
|
|
|
|
float
|
|
AudioDiskstream::playback_buffer_load () const
|
|
{
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
if (c->empty ()) {
|
|
return 1.0;
|
|
}
|
|
|
|
return (float) ((double) c->front()->playback_buf->read_space()/
|
|
(double) c->front()->playback_buf->bufsize());
|
|
}
|
|
|
|
float
|
|
AudioDiskstream::capture_buffer_load () const
|
|
{
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
if (c->empty ()) {
|
|
return 1.0;
|
|
}
|
|
|
|
return (float) ((double) c->front()->capture_buf->write_space()/
|
|
(double) c->front()->capture_buf->bufsize());
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::use_pending_capture_data (XMLNode& node)
|
|
{
|
|
XMLProperty const * prop;
|
|
XMLNodeList nlist = node.children();
|
|
XMLNodeIterator niter;
|
|
boost::shared_ptr<AudioFileSource> fs;
|
|
boost::shared_ptr<AudioFileSource> first_fs;
|
|
SourceList pending_sources;
|
|
framepos_t position;
|
|
|
|
if ((prop = node.property (X_("at"))) == 0) {
|
|
return -1;
|
|
}
|
|
|
|
if (sscanf (prop->value().c_str(), "%" PRIu64, &position) != 1) {
|
|
return -1;
|
|
}
|
|
|
|
for (niter = nlist.begin(); niter != nlist.end(); ++niter) {
|
|
if ((*niter)->name() == X_("file")) {
|
|
|
|
if ((prop = (*niter)->property (X_("path"))) == 0) {
|
|
continue;
|
|
}
|
|
|
|
// This protects sessions from errant CapturingSources in stored sessions
|
|
GStatBuf sbuf;
|
|
if (g_stat (prop->value().c_str(), &sbuf)) {
|
|
continue;
|
|
}
|
|
|
|
/* XXX as of June 2014, we always record to mono
|
|
files. Since this Source is being created as part of
|
|
crash recovery, we know that we need the first
|
|
channel (the final argument to the SourceFactory
|
|
call below). If we ever support non-mono files for
|
|
capture, this will need rethinking.
|
|
*/
|
|
|
|
try {
|
|
fs = boost::dynamic_pointer_cast<AudioFileSource> (SourceFactory::createForRecovery (DataType::AUDIO, _session, prop->value(), 0));
|
|
}
|
|
|
|
catch (failed_constructor& err) {
|
|
error << string_compose (_("%1: cannot restore pending capture source file %2"),
|
|
_name, prop->value())
|
|
<< endmsg;
|
|
return -1;
|
|
}
|
|
|
|
pending_sources.push_back (fs);
|
|
|
|
if (first_fs == 0) {
|
|
first_fs = fs;
|
|
}
|
|
|
|
fs->set_captured_for (_name.val());
|
|
}
|
|
}
|
|
|
|
if (pending_sources.size() == 0) {
|
|
/* nothing can be done */
|
|
return 1;
|
|
}
|
|
|
|
if (pending_sources.size() != _n_channels.n_audio()) {
|
|
error << string_compose (_("%1: incorrect number of pending sources listed - ignoring them all"), _name)
|
|
<< endmsg;
|
|
return -1;
|
|
}
|
|
|
|
try {
|
|
|
|
boost::shared_ptr<AudioRegion> wf_region;
|
|
boost::shared_ptr<AudioRegion> region;
|
|
|
|
/* First create the whole file region */
|
|
|
|
PropertyList plist;
|
|
|
|
plist.add (Properties::start, 0);
|
|
plist.add (Properties::length, first_fs->length (first_fs->timeline_position()));
|
|
plist.add (Properties::name, region_name_from_path (first_fs->name(), true));
|
|
|
|
wf_region = boost::dynamic_pointer_cast<AudioRegion> (RegionFactory::create (pending_sources, plist));
|
|
|
|
wf_region->set_automatic (true);
|
|
wf_region->set_whole_file (true);
|
|
wf_region->special_set_position (position);
|
|
|
|
/* Now create a region that isn't the whole file for adding to
|
|
* the playlist */
|
|
|
|
region = boost::dynamic_pointer_cast<AudioRegion> (RegionFactory::create (pending_sources, plist));
|
|
|
|
_playlist->add_region (region, position);
|
|
}
|
|
|
|
catch (failed_constructor& err) {
|
|
error << string_compose (
|
|
_("%1: cannot create whole-file region from pending capture sources"),
|
|
_name) << endmsg;
|
|
|
|
return -1;
|
|
}
|
|
|
|
|
|
return 0;
|
|
}
|
|
|
|
#ifdef XXX_OLD_DESTRUCTIVE_API_XXX
|
|
int
|
|
AudioDiskstream::set_non_layered (bool yn)
|
|
{
|
|
if (yn != non_layered()) {
|
|
|
|
if (yn) {
|
|
_flags = Flag (_flags | NonLayered);
|
|
} else {
|
|
_flags = Flag (_flags & ~NonLayered);
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
AudioDiskstream::set_destructive (bool yn)
|
|
{
|
|
if (yn != destructive()) {
|
|
|
|
if (yn) {
|
|
bool bounce_ignored;
|
|
/* requestor should already have checked this and
|
|
bounced if necessary and desired
|
|
*/
|
|
if (!can_become_destructive (bounce_ignored)) {
|
|
return -1;
|
|
}
|
|
_flags = Flag (_flags | Destructive);
|
|
use_destructive_playlist ();
|
|
} else {
|
|
_flags = Flag (_flags & ~Destructive);
|
|
reset_write_sources (true, true);
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
bool
|
|
AudioDiskstream::can_become_destructive (bool& requires_bounce) const
|
|
{
|
|
if (Profile->get_trx()) {
|
|
return false;
|
|
}
|
|
|
|
if (!_playlist) {
|
|
requires_bounce = false;
|
|
return false;
|
|
}
|
|
|
|
/* if no regions are present: easy */
|
|
|
|
if (_playlist->n_regions() == 0) {
|
|
requires_bounce = false;
|
|
return true;
|
|
}
|
|
|
|
/* is there only one region ? */
|
|
|
|
if (_playlist->n_regions() != 1) {
|
|
requires_bounce = true;
|
|
return false;
|
|
}
|
|
|
|
boost::shared_ptr<Region> first;
|
|
{
|
|
const RegionList& rl (_playlist->region_list_property().rlist());
|
|
assert((rl.size() == 1));
|
|
first = rl.front();
|
|
|
|
}
|
|
|
|
if (!first) {
|
|
requires_bounce = false;
|
|
return true;
|
|
}
|
|
|
|
/* do the source(s) for the region cover the session start position ? */
|
|
|
|
if (first->position() != _session.current_start_frame()) {
|
|
// what is the idea here? why start() ??
|
|
if (first->start() > _session.current_start_frame()) {
|
|
requires_bounce = true;
|
|
return false;
|
|
}
|
|
}
|
|
|
|
/* currently RouteTimeAxisView::set_track_mode does not
|
|
* implement bounce. Existing regions cannot be converted.
|
|
*
|
|
* so let's make sure this region is already set up
|
|
* as tape-track (spanning the complete range)
|
|
*/
|
|
if (first->length() != max_framepos - first->position()) {
|
|
requires_bounce = true;
|
|
return false;
|
|
}
|
|
|
|
/* is the source used by only 1 playlist ? */
|
|
|
|
boost::shared_ptr<AudioRegion> afirst = boost::dynamic_pointer_cast<AudioRegion> (first);
|
|
|
|
assert (afirst);
|
|
|
|
if (_session.playlists->source_use_count (afirst->source()) > 1) {
|
|
requires_bounce = true;
|
|
return false;
|
|
}
|
|
|
|
requires_bounce = false;
|
|
return true;
|
|
}
|
|
#endif
|
|
|
|
void
|
|
AudioDiskstream::adjust_playback_buffering ()
|
|
{
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
(*chan)->resize_playback (_session.butler()->audio_diskstream_playback_buffer_size());
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::adjust_capture_buffering ()
|
|
{
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
|
|
for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
|
|
(*chan)->resize_capture (_session.butler()->audio_diskstream_capture_buffer_size());
|
|
}
|
|
}
|
|
|
|
bool
|
|
AudioDiskstream::ChannelSource::is_physical () const
|
|
{
|
|
if (name.empty()) {
|
|
return false;
|
|
}
|
|
|
|
return AudioEngine::instance()->port_is_physical (name);
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::ChannelSource::request_input_monitoring (bool yn) const
|
|
{
|
|
if (name.empty()) {
|
|
return;
|
|
}
|
|
|
|
return AudioEngine::instance()->request_input_monitoring (name, yn);
|
|
}
|
|
|
|
AudioDiskstream::ChannelInfo::ChannelInfo (framecnt_t playback_bufsize, framecnt_t capture_bufsize, framecnt_t speed_size, framecnt_t wrap_size)
|
|
{
|
|
current_capture_buffer = 0;
|
|
current_playback_buffer = 0;
|
|
curr_capture_cnt = 0;
|
|
|
|
speed_buffer = new Sample[speed_size];
|
|
playback_wrap_buffer = new Sample[wrap_size];
|
|
capture_wrap_buffer = new Sample[wrap_size];
|
|
|
|
playback_buf = new RingBufferNPT<Sample> (playback_bufsize);
|
|
capture_buf = new RingBufferNPT<Sample> (capture_bufsize);
|
|
capture_transition_buf = new RingBufferNPT<CaptureTransition> (256);
|
|
|
|
/* touch the ringbuffer buffers, which will cause
|
|
them to be mapped into locked physical RAM if
|
|
we're running with mlockall(). this doesn't do
|
|
much if we're not.
|
|
*/
|
|
|
|
memset (playback_buf->buffer(), 0, sizeof (Sample) * playback_buf->bufsize());
|
|
memset (capture_buf->buffer(), 0, sizeof (Sample) * capture_buf->bufsize());
|
|
memset (capture_transition_buf->buffer(), 0, sizeof (CaptureTransition) * capture_transition_buf->bufsize());
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::ChannelInfo::resize_playback (framecnt_t playback_bufsize)
|
|
{
|
|
delete playback_buf;
|
|
playback_buf = new RingBufferNPT<Sample> (playback_bufsize);
|
|
memset (playback_buf->buffer(), 0, sizeof (Sample) * playback_buf->bufsize());
|
|
}
|
|
|
|
void
|
|
AudioDiskstream::ChannelInfo::resize_capture (framecnt_t capture_bufsize)
|
|
{
|
|
delete capture_buf;
|
|
|
|
capture_buf = new RingBufferNPT<Sample> (capture_bufsize);
|
|
memset (capture_buf->buffer(), 0, sizeof (Sample) * capture_buf->bufsize());
|
|
}
|
|
|
|
AudioDiskstream::ChannelInfo::~ChannelInfo ()
|
|
{
|
|
if (write_source) {
|
|
if (write_source->removable()) {
|
|
/* this is a "stub" write source which exists in the
|
|
Session source list, but is removable. We must emit
|
|
a drop references call because it should not
|
|
continue to exist. If we do not do this, then the
|
|
Session retains a reference to it, it is not
|
|
deleted, and later attempts to create a new source
|
|
file will use wierd naming because it already
|
|
exists.
|
|
|
|
XXX longer term TO-DO: do not add to session source
|
|
list until we write to the source.
|
|
*/
|
|
write_source->drop_references ();
|
|
}
|
|
}
|
|
|
|
write_source.reset ();
|
|
|
|
delete [] speed_buffer;
|
|
speed_buffer = 0;
|
|
|
|
delete [] playback_wrap_buffer;
|
|
playback_wrap_buffer = 0;
|
|
|
|
delete [] capture_wrap_buffer;
|
|
capture_wrap_buffer = 0;
|
|
|
|
delete playback_buf;
|
|
playback_buf = 0;
|
|
|
|
delete capture_buf;
|
|
capture_buf = 0;
|
|
|
|
delete capture_transition_buf;
|
|
capture_transition_buf = 0;
|
|
}
|
|
|
|
|
|
bool
|
|
AudioDiskstream::set_name (string const & name)
|
|
{
|
|
if (_name == name) {
|
|
return true;
|
|
}
|
|
Diskstream::set_name (name);
|
|
|
|
/* get a new write source so that its name reflects the new diskstream name */
|
|
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
ChannelList::iterator i;
|
|
int n = 0;
|
|
|
|
for (n = 0, i = c->begin(); i != c->end(); ++i, ++n) {
|
|
use_new_write_source (n);
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool
|
|
AudioDiskstream::set_write_source_name (const std::string& str) {
|
|
if (_write_source_name == str) {
|
|
return true;
|
|
}
|
|
|
|
Diskstream::set_write_source_name (str);
|
|
|
|
if (_write_source_name == name()) {
|
|
return true;
|
|
}
|
|
boost::shared_ptr<ChannelList> c = channels.reader();
|
|
ChannelList::iterator i;
|
|
int n = 0;
|
|
|
|
for (n = 0, i = c->begin(); i != c->end(); ++i, ++n) {
|
|
use_new_write_source (n);
|
|
}
|
|
return true;
|
|
}
|