/*
    Copyright (C) 2000-2006 Paul Davis

    This program is free software; you can redistribute it and/or modify
    it under the terms of the GNU General Public License as published by
    the Free Software Foundation; either version 2 of the License, or
    (at your option) any later version.

    This program is distributed in the hope that it will be useful,
    but WITHOUT ANY WARRANTY; without even the implied warranty of
    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
    GNU General Public License for more details.

    You should have received a copy of the GNU General Public License
    along with this program; if not, write to the Free Software
    Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/

#include <cstdio>
#include <unistd.h>
#include <cmath>
#include <cerrno>
#include <cassert>
#include <string>
#include <climits>
#include <fcntl.h>
#include <cstdlib>
#include <ctime>

#include "pbd/gstdio_compat.h"
#include "pbd/error.h"
#include "pbd/xml++.h"
#include "pbd/memento_command.h"
#include "pbd/enumwriter.h"
#include "pbd/stateful_diff_command.h"

#include "ardour/analyser.h"
#include "ardour/audio_buffer.h"
#include "ardour/audio_diskstream.h"
#include "ardour/audio_port.h"
#include "ardour/audioengine.h"
#include "ardour/audiofilesource.h"
#include "ardour/audioplaylist.h"
#include "ardour/audioregion.h"
#include "ardour/butler.h"
#include "ardour/debug.h"
#include "ardour/io.h"
#include "ardour/playlist_factory.h"
#include "ardour/profile.h"
#include "ardour/region_factory.h"
#include "ardour/session.h"
#include "ardour/session_playlists.h"
#include "ardour/sndfile_helpers.h"
#include "ardour/source_factory.h"
#include "ardour/track.h"
#include "ardour/types.h"
#include "ardour/utils.h"

#include "pbd/i18n.h"
#include <locale.h>

using namespace std;
using namespace ARDOUR;
using namespace PBD;

Sample* AudioDiskstream::_mixdown_buffer       = 0;
gain_t* AudioDiskstream::_gain_buffer          = 0;

AudioDiskstream::AudioDiskstream (Session &sess, const string &name, Diskstream::Flag flag)
	: Diskstream(sess, name, flag)
	, channels (new ChannelList)
{
	/* prevent any write sources from being created */

	in_set_state = true;
	use_new_playlist ();
	in_set_state = false;

	if (flag & Destructive) {
		use_destructive_playlist ();
	}
}

AudioDiskstream::AudioDiskstream (Session& sess, const XMLNode& node)
	: Diskstream(sess, node)
	, channels (new ChannelList)
{
	in_set_state = true;
	init ();

	if (set_state (node, Stateful::loading_state_version)) {
		in_set_state = false;
		throw failed_constructor();
	}

	in_set_state = false;

	if (destructive()) {
		use_destructive_playlist ();
	}
}

void
AudioDiskstream::init ()
{
	/* there are no channels at this point, so these
	   two calls just get speed_buffer_size and wrap_buffer
	   size setup without duplicating their code.
	*/

	set_block_size (_session.get_block_size());
	allocate_temporary_buffers ();
}

AudioDiskstream::~AudioDiskstream ()
{
	DEBUG_TRACE (DEBUG::Destruction, string_compose ("Audio Diskstream %1 destructor\n", _name));

	{
		RCUWriter<ChannelList> writer (channels);
		boost::shared_ptr<ChannelList> c = writer.get_copy();

		for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
			delete *chan;
		}

		c->clear();
	}

	channels.flush ();
}

void
AudioDiskstream::allocate_working_buffers()
{
	/* with varifill buffer refilling, we compute the read size in bytes (to optimize
	   for disk i/o bandwidth) and then convert back into samples. These buffers
	   need to reflect the maximum size we could use, which is 4MB reads, or 2M samples
	   using 16 bit samples.
	*/
	_mixdown_buffer       = new Sample[2*1048576];
	_gain_buffer          = new gain_t[2*1048576];
}

void
AudioDiskstream::free_working_buffers()
{
	delete [] _mixdown_buffer;
	delete [] _gain_buffer;
	_mixdown_buffer       = 0;
	_gain_buffer          = 0;
}

void
AudioDiskstream::non_realtime_input_change ()
{
	bool need_write_sources = false;

	{
		Glib::Threads::Mutex::Lock lm (state_lock);

		if (input_change_pending.type == IOChange::NoChange) {
			return;
		}

		boost::shared_ptr<ChannelList> cr = channels.reader();
		if (!cr->empty() && !cr->front()->write_source) {
			need_write_sources = true;
		}

		if (input_change_pending.type & IOChange::ConfigurationChanged) {
			RCUWriter<ChannelList> writer (channels);
			boost::shared_ptr<ChannelList> c = writer.get_copy();

			_n_channels.set(DataType::AUDIO, c->size());

			if (_io->n_ports().n_audio() > _n_channels.n_audio()) {
				add_channel_to (c, _io->n_ports().n_audio() - _n_channels.n_audio());
			} else if (_io->n_ports().n_audio() < _n_channels.n_audio()) {
				remove_channel_from (c, _n_channels.n_audio() - _io->n_ports().n_audio());
			}

			need_write_sources = true;
		}

		if (input_change_pending.type & IOChange::ConnectionsChanged) {
			get_input_sources ();
			set_capture_offset ();
			set_align_style_from_io ();
		}

		input_change_pending = IOChange::NoChange;

		/* implicit unlock */
	}

	if (need_write_sources) {
		reset_write_sources (false);
	}

	/* now refill channel buffers */

	if (speed() != 1.0f || speed() != -1.0f) {
		seek ((framepos_t) (_session.transport_frame() * (double) speed()));
	} else {
		seek (_session.transport_frame());
	}
}

void
AudioDiskstream::non_realtime_locate (framepos_t location)
{
	/* now refill channel buffers */

	if (speed() != 1.0f || speed() != -1.0f) {
		seek ((framepos_t) (location * (double) speed()), true);
	} else {
		seek (location, true);
	}
}

void
AudioDiskstream::get_input_sources ()
{
	boost::shared_ptr<ChannelList> c = channels.reader();

	uint32_t n;
	ChannelList::iterator chan;
	uint32_t ni = _io->n_ports().n_audio();
	vector<string> connections;

	for (n = 0, chan = c->begin(); chan != c->end() && n < ni; ++chan, ++n) {

		connections.clear ();

		if ((_io->nth (n).get()) && (_io->nth (n)->get_connections (connections) == 0)) {
			if (!(*chan)->source.name.empty()) {
				// _source->disable_metering ();
			}
			(*chan)->source.name = string();
		} else {
			(*chan)->source.name = connections[0];
		}
	}
}

int
AudioDiskstream::find_and_use_playlist (const string& name)
{
	boost::shared_ptr<AudioPlaylist> playlist;

	if ((playlist = boost::dynamic_pointer_cast<AudioPlaylist> (_session.playlists->by_name (name))) == 0) {
		playlist = boost::dynamic_pointer_cast<AudioPlaylist> (PlaylistFactory::create (DataType::AUDIO, _session, name));
	}

	if (!playlist) {
		error << string_compose(_("AudioDiskstream: Playlist \"%1\" isn't an audio playlist"), name) << endmsg;
		return -1;
	}

	return use_playlist (playlist);
}

int
AudioDiskstream::use_playlist (boost::shared_ptr<Playlist> playlist)
{
	assert(boost::dynamic_pointer_cast<AudioPlaylist>(playlist));

	Diskstream::use_playlist(playlist);

	return 0;
}

int
AudioDiskstream::use_new_playlist ()
{
	string newname;
	boost::shared_ptr<AudioPlaylist> playlist;

	if (!in_set_state && destructive()) {
		return 0;
	}

	if (_playlist) {
		newname = Playlist::bump_name (_playlist->name(), _session);
	} else {
		newname = Playlist::bump_name (_name, _session);
	}

	if ((playlist = boost::dynamic_pointer_cast<AudioPlaylist> (PlaylistFactory::create (DataType::AUDIO, _session, newname, hidden()))) != 0) {

		return use_playlist (playlist);

	} else {
		return -1;
	}
}

int
AudioDiskstream::use_copy_playlist ()
{
	assert(audio_playlist());

	if (destructive()) {
		return 0;
	}

	if (_playlist == 0) {
		error << string_compose(_("AudioDiskstream %1: there is no existing playlist to make a copy of!"), _name) << endmsg;
		return -1;
	}

	string newname;
	boost::shared_ptr<AudioPlaylist> playlist;

	newname = Playlist::bump_name (_playlist->name(), _session);

	if ((playlist = boost::dynamic_pointer_cast<AudioPlaylist>(PlaylistFactory::create (audio_playlist(), newname))) != 0) {
		return use_playlist (playlist);
	} else {
		return -1;
	}
}

void
AudioDiskstream::setup_destructive_playlist ()
{
	SourceList srcs;
	boost::shared_ptr<ChannelList> c = channels.reader();

	for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {
		srcs.push_back ((*chan)->write_source);
	}

	/* a single full-sized region */

	assert (!srcs.empty ());

	PropertyList plist;
	plist.add (Properties::name, _name.val());
	plist.add (Properties::start, 0);
	plist.add (Properties::length, max_framepos - srcs.front()->natural_position());

	boost::shared_ptr<Region> region (RegionFactory::create (srcs, plist));
	_playlist->add_region (region, srcs.front()->natural_position());

	/* apply region properties and update write sources */
	use_destructive_playlist();
}

void
AudioDiskstream::use_destructive_playlist ()
{
	/* this is called from the XML-based constructor or ::set_destructive. when called,
	   we already have a playlist and a region, but we need to
	   set up our sources for write. we use the sources associated
	   with the (presumed single, full-extent) region.
	*/

	boost::shared_ptr<Region> rp;
	{
		const RegionList& rl (_playlist->region_list_property().rlist());
		if (rl.size() > 0) {
			assert((rl.size() == 1));
			rp = rl.front();
		}
	}

	if (!rp) {
		reset_write_sources (false, true);
		return;
	}

	boost::shared_ptr<AudioRegion> region = boost::dynamic_pointer_cast<AudioRegion> (rp);

	if (region == 0) {
		throw failed_constructor();
	}

	/* be sure to stretch the region out to the maximum length (non-musical)*/

	region->set_length (max_framepos - region->position(), 0);

	uint32_t n;
	ChannelList::iterator chan;
	boost::shared_ptr<ChannelList> c = channels.reader();

	for (n = 0, chan = c->begin(); chan != c->end(); ++chan, ++n) {
		(*chan)->write_source = boost::dynamic_pointer_cast<AudioFileSource>(region->source (n));
		assert((*chan)->write_source);
		(*chan)->write_source->set_allow_remove_if_empty (false);

		/* this might be false if we switched modes, so force it */

		(*chan)->write_source->set_destructive (true);
	}

	/* the source list will never be reset for a destructive track */
}

void
AudioDiskstream::prepare_record_status(framepos_t capture_start_frame)
{
	if (recordable() && destructive()) {
		boost::shared_ptr<ChannelList> c = channels.reader();
		for (ChannelList::iterator chan = c->begin(); chan != c->end(); ++chan) {

			RingBufferNPT<CaptureTransition>::rw_vector transitions;
			(*chan)->capture_transition_buf->get_write_vector (&transitions);

			if (transitions.len[0] > 0) {
				transitions.buf[0]->type = CaptureStart;
				transitions.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 start!  inconceivable!")
					<< endmsg;
			}
		}
	}
}


/** Do some record stuff [not described in this comment!]
 *
 *  Also:
 *    - Setup playback_distance with the nframes, or nframes adjusted
 *      for current varispeed, if appropriate.
 *    - Setup current_playback_buffer in each ChannelInfo to point to data
 *      that someone can read playback_distance worth of data from.
 */
int
AudioDiskstream::process (BufferSet& bufs, framepos_t transport_frame, pframes_t nframes, framecnt_t& playback_distance, bool need_disk_signal)
{
	uint32_t n;
	boost::shared_ptr<ChannelList> c = channels.reader();
	ChannelList::iterator chan;
	framecnt_t rec_offset = 0;
	framecnt_t rec_nframes = 0;
	bool collect_playback = false;
	bool can_record = _session.actively_recording ();

	playback_distance = 0;

	if (!_io || !_io->active()) {
		return 0;
	}

	check_record_status (transport_frame, can_record);

	if (nframes == 0) {
		return 0;
	}

	Glib::Threads::Mutex::Lock sm (state_lock, Glib::Threads::TRY_LOCK);

	if (!sm.locked()) {
		return 1;
	}

	adjust_capture_position = 0;

	for (chan = c->begin(); chan != c->end(); ++chan) {
		(*chan)->current_capture_buffer = 0;
		(*chan)->current_playback_buffer = 0;
	}

	// Safeguard against situations where process() goes haywire when autopunching
	// and last_recordable_frame < first_recordable_frame

	if (last_recordable_frame < first_recordable_frame) {
		last_recordable_frame = max_framepos;
	}

	if (record_enabled()) {

		Evoral::OverlapType ot = Evoral::coverage (first_recordable_frame, last_recordable_frame, transport_frame, transport_frame + nframes);
		// XXX should this be transport_frame + nframes - 1 ? coverage() expects its parameter ranges to include their end points
		// XXX also, first_recordable_frame & last_recordable_frame may both be == max_framepos: coverage() will return OverlapNone in that case. Is thak OK?
		calculate_record_range (ot, transport_frame, nframes, rec_nframes, rec_offset);

		DEBUG_TRACE (DEBUG::CaptureAlignment, string_compose ("%1: this time record %2 of %3 frames, offset %4\n", _name, rec_nframes, nframes, rec_offset));

		if (rec_nframes && !was_recording) {
			capture_captured = 0;
			was_recording = true;
		}
	}

	if (can_record && !_last_capture_sources.empty()) {
		_last_capture_sources.clear ();
	}

	if (rec_nframes) {

		uint32_t limit = _io->n_ports ().n_audio();

		/* one or more ports could already have been removed from _io, but our
		   channel setup hasn't yet been updated. prevent us from trying to
		   use channels that correspond to missing ports. note that the
		   process callback (from which this is called) is always atomic
		   with respect to port removal/addition.
		*/

		for (n = 0, chan = c->begin(); chan != c->end() && n < limit; ++chan, ++n) {

			ChannelInfo* chaninfo (*chan);

			chaninfo->capture_buf->get_write_vector (&chaninfo->capture_vector);

			if (rec_nframes <= (framecnt_t) chaninfo->capture_vector.len[0]) {

				chaninfo->current_capture_buffer = chaninfo->capture_vector.buf[0];

				/* note: grab the entire port buffer, but only copy what we were supposed to
				   for recording, and use rec_offset
				*/

				boost::shared_ptr<AudioPort> const ap = _io->audio (n);
				assert(ap);
				assert(rec_nframes <= (framecnt_t) ap->get_audio_buffer(nframes).capacity());

				Sample *buf = bufs.get_audio (n).data(rec_offset);
				memcpy (chaninfo->current_capture_buffer, buf, sizeof (Sample) * rec_nframes);

			} else {

				framecnt_t total = chaninfo->capture_vector.len[0] + chaninfo->capture_vector.len[1];

				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 ();

		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);
			}

			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, 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());
	char buf[64] = "";
	LocaleGuard lg;

	boost::shared_ptr<ChannelList> c = channels.reader();
	snprintf (buf, sizeof(buf), "%u", (unsigned int) c->size());
	node.add_property ("channels", buf);

	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->add_property (X_("path"), (*i)->path());
			cs_child->add_child_nocopy (*cs_grandchild);
		}

		/* store the location where capture will start */

		Location* pi;

		if (_session.config.get_punch_in() && ((pi = _session.locations()->auto_punch_location()) != 0)) {
			snprintf (buf, sizeof (buf), "%" PRId64, pi->start());
		} else {
			snprintf (buf, sizeof (buf), "%" PRId64, _session.transport_frame());
		}

		cs_child->add_property (X_("at"), buf);
		node.add_child_nocopy (*cs_child);
	}

	return node;
}

int
AudioDiskstream::set_state (const XMLNode& node, int version)
{
	XMLProperty const * prop;
	XMLNodeList nlist = node.children();
	XMLNodeIterator niter;
	uint32_t nchans = 1;
	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;
	}

	if ((prop = node.property ("channels")) != 0) {
		nchans = atoi (prop->value().c_str());
	}

	// 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;
}

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;
}

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;
}