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authorMichele Calgaro <michele.calgaro@yahoo.it>2020-12-06 19:28:06 +0900
committerMichele Calgaro <michele.calgaro@yahoo.it>2020-12-06 19:28:06 +0900
commit00d4f92b717fbcbed6f9eee361975d6ee5380d59 (patch)
tree043b5970d66e539e1fbf6dde03440d6569e34c4e /flow/audioiomas.cpp
parent2f53bfe61c8ee78ff36ac6c66ae714b01e407b33 (diff)
downloadarts-00d4f92b717fbcbed6f9eee361975d6ee5380d59.tar.gz
arts-00d4f92b717fbcbed6f9eee361975d6ee5380d59.zip
Renaming of files in preparation for code style tools.
Signed-off-by: Michele Calgaro <michele.calgaro@yahoo.it>
Diffstat (limited to 'flow/audioiomas.cpp')
-rw-r--r--flow/audioiomas.cpp619
1 files changed, 619 insertions, 0 deletions
diff --git a/flow/audioiomas.cpp b/flow/audioiomas.cpp
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--- /dev/null
+++ b/flow/audioiomas.cpp
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+ /*
+
+ Copyright (C) 2001-2003 Stefan Westerfeld
+ stefan@space.twc.de
+
+ This library is free software; you can redistribute it and/or
+ modify it under the terms of the GNU Library General Public
+ License as published by the Free Software Foundation; either
+ version 2 of the License, or (at your option) any later version.
+
+ This library 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
+ Library General Public License for more details.
+
+ You should have received a copy of the GNU Library General Public License
+ along with this library; see the file COPYING.LIB. If not, write to
+ the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
+ Boston, MA 02110-1301, USA.
+
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+/*
+ * Only compile this AudioIO class if we have MAS
+ */
+#ifdef HAVE_LIBMAS
+
+extern "C" {
+#include <mas/mas.h>
+#include <mas/mas_getset.h>
+#include <mas/mas_source.h>
+}
+
+#include <sys/types.h>
+#include <sys/ioctl.h>
+#include <sys/time.h>
+#include <sys/stat.h>
+
+#include <assert.h>
+#include <errno.h>
+#include <fcntl.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <iostream>
+#include <algorithm>
+
+#include "debug.h"
+#include "audioio.h"
+#include "audiosubsys.h"
+#include "iomanager.h"
+#include "dispatcher.h"
+
+namespace Arts {
+
+ class AudioIOMAS : public AudioIO, public TimeNotify {
+ protected:
+ mas_channel_t audio_channel;
+ mas_port_t mix_sink;
+ mas_port_t srate_source, srate_sink;
+ mas_port_t audio_source, audio_sink;
+ mas_port_t endian_sink, endian_source;
+ mas_port_t sbuf_source, sbuf_sink;
+ mas_port_t squant_sink, squant_source;
+ mas_port_t open_source; /* (!) */
+ mas_device_t endian;
+ mas_device_t srate;
+ mas_device_t squant;
+ mas_device_t sbuf;
+ mas_data *data;
+ mas_package package;
+ int32 mas_error;
+
+ std::list<mas_channel_t> allocated_channels;
+ std::list<mas_port_t> allocated_ports;
+ std::list<mas_device_t> allocated_devices;
+
+ double lastUpdate, bytesPerSec;
+ int readBufferAvailable;
+ int writeBufferAvailable;
+
+ double currentTime();
+ void updateBufferSizes();
+
+#ifdef WORDS_BIGENDIAN
+ static const int defaultFormat = 17;
+#else
+ static const int defaultFormat = 16;
+#endif
+ bool close_with_error(const std::string& text);
+ public:
+ AudioIOMAS();
+
+ // Timer callback
+ void notifyTime();
+
+ void setParam(AudioParam param, int& value);
+ int getParam(AudioParam param);
+
+ bool open();
+ void close();
+ int read(void *buffer, int size);
+ int write(void *buffer, int size);
+ };
+
+ REGISTER_AUDIO_IO(AudioIOMAS,"mas","MAS Audio Input/Output");
+
+};
+
+using namespace std;
+using namespace Arts;
+
+AudioIOMAS::AudioIOMAS()
+{
+ /*
+ * default parameters
+ */
+ param(samplingRate) = 44100;
+ paramStr(deviceName) = ""; // TODO
+ param(fragmentSize) = 4096;
+ param(fragmentCount) = 7;
+ param(channels) = 2;
+ param(direction) = 2;
+ param(format) = defaultFormat;
+}
+
+namespace {
+ int masInitCount = 0;
+}
+
+// Opens the audio device
+bool AudioIOMAS::open()
+{
+ string& _error = paramStr(lastError);
+ string& _deviceName = paramStr(deviceName);
+ int& _channels = param(channels);
+ int& _fragmentSize = param(fragmentSize);
+ int& _fragmentCount = param(fragmentCount);
+ int& _samplingRate = param(samplingRate);
+ int& _format = param(format);
+
+ /* FIXME: do we need to free what we allocate with mas_init() in close() */
+ if (!masInitCount)
+ {
+ mas_error = mas_init();
+
+ if (mas_error < 0)
+ return close_with_error("error connecting to MAS server");
+ }
+ masInitCount++;
+
+ if (param(direction) != 2)
+ {
+ _error = "unsupported direction (currently no full duplex support)";
+ return false;
+ }
+
+ /*
+ * data path
+ *
+ * audio_sink
+ * audio_channel: data_channel ("artsd")
+ * audio_source
+ * |
+ * V
+ * endian_sink
+ * endian: instantiate_device ("endian")
+ * open_source = endian_source
+ * |
+ * V
+ * [squant_sink]
+ * [squant]
+ * [squant_source]
+ * |
+ * V
+ * [srate_sink]
+ * [srate]
+ * [srate_source]
+ * |
+ * V
+ * sbuf_sink
+ * sbuf
+ * sbuf_source
+ * |
+ * V
+ * mix_sink: port ("default_mix_sink")
+ */
+
+ // audio_channel, source & sink
+ mas_error = mas_make_data_channel("artsd", &audio_channel, &audio_source, &audio_sink);
+ if (mas_error < 0)
+ return close_with_error("error initializing MAS data channel");
+
+ allocated_channels.push_back(audio_channel);
+ allocated_ports.push_back(audio_source);
+ allocated_ports.push_back(audio_sink);
+
+ // endian, source & sink
+ mas_error = mas_asm_instantiate_device( "endian", 0, 0, &endian );
+ if ( mas_error < 0 )
+ return close_with_error("error initantiating MAS endian device");
+
+ allocated_devices.push_back(endian);
+
+ mas_error = mas_asm_get_port_by_name( endian, "sink", &endian_sink );
+ if ( mas_error < 0 )
+ return close_with_error("error getting MAS endian device sink port");
+
+ allocated_ports.push_back(endian_sink);
+
+ mas_error = mas_asm_get_port_by_name( endian, "source", &endian_source );
+ if ( mas_error < 0 )
+ return close_with_error("error getting MAS endian device source port");
+
+ allocated_ports.push_back(endian_source);
+
+ char ratestring[16], resolutionstring[16];
+ sprintf (ratestring, "%u", _samplingRate);
+ sprintf (resolutionstring, "%u", _format);
+
+ mas_data_characteristic* dc;
+
+ dc = (mas_data_characteristic *)MAS_NEW( dc );
+ masc_setup_dc( dc, 6 );
+ masc_append_dc_key_value( dc, "format", (_format==8) ? "ulinear":"linear" );
+
+ masc_append_dc_key_value( dc, "resolution", resolutionstring );
+ masc_append_dc_key_value( dc, "sampling rate", ratestring );
+ masc_append_dc_key_value( dc, "channels", "2" );
+ masc_append_dc_key_value( dc, "endian", "little" );
+
+ mas_error = mas_asm_connect_source_sink( audio_source, endian_sink, dc );
+ if ( mas_error < 0 )
+ return close_with_error("error connecting MAS net audio source to endian sink");
+
+ /* The next device is 'if needed' only. After the following if()
+ statement, open_source will contain the current unconnected
+ source in the path (will be either endian_source or
+ squant_source in this case)
+ */
+ open_source = endian_source;
+
+ if ( _format != 16 )
+ {
+ arts_debug("MAS output: Sample resolution is not 16 bit/sample, instantiating squant device.");
+
+ // squant, source & sink
+ mas_error = mas_asm_instantiate_device( "squant", 0, 0, &squant );
+ if ( mas_error < 0 )
+ return close_with_error("error creating MAS squant device");
+
+ allocated_devices.push_back(squant);
+
+ mas_error = mas_asm_get_port_by_name( squant, "sink", &squant_sink );
+ if ( mas_error < 0 )
+ return close_with_error("error getting MAS squant device sink port");
+
+ allocated_ports.push_back(squant_sink);
+
+ mas_error = mas_asm_get_port_by_name( squant, "source", &squant_source );
+ if ( mas_error < 0 )
+ return close_with_error("error getting MAS squant device source port");
+
+ allocated_ports.push_back(squant_source);
+
+ arts_debug( "MAS output: Connecting endian -> squant.");
+
+ masc_strike_dc( dc );
+ masc_setup_dc( dc, 6 );
+ masc_append_dc_key_value( dc,"format",(_format==8) ? "ulinear":"linear" );
+ masc_append_dc_key_value( dc, "resolution", resolutionstring );
+ masc_append_dc_key_value( dc, "sampling rate", ratestring );
+ masc_append_dc_key_value( dc, "channels", "2" );
+ masc_append_dc_key_value( dc, "endian", "host" );
+
+ mas_error = mas_asm_connect_source_sink( endian_source, squant_sink, dc );
+ if ( mas_error < 0 )
+ return close_with_error("error connecting MAS endian output to squant device");
+
+ /* sneaky: the squant device is optional -> pretend it isn't there */
+ open_source = squant_source;
+ }
+
+
+ /* Another 'if necessary' device, as above */
+ if ( _samplingRate != 44100 )
+ {
+ arts_debug ("MAS output: Sample rate is not 44100, instantiating srate device.");
+
+ // srate, source & sink
+ mas_error = mas_asm_instantiate_device( "srate", 0, 0, &srate );
+ if ( mas_error < 0 )
+ return close_with_error("error initantiating MAS srate device");
+
+ allocated_devices.push_back(srate);
+
+ mas_error = mas_asm_get_port_by_name( srate, "sink", &srate_sink );
+ if ( mas_error < 0 )
+ return close_with_error("error getting MAS srate sink port");
+
+ allocated_ports.push_back(srate_sink);
+
+ mas_error = mas_asm_get_port_by_name( srate, "source", &srate_source );
+ if ( mas_error < 0 )
+ return close_with_error("error getting MAS srate source port");
+
+ allocated_ports.push_back(srate_source);
+
+ arts_debug( "MAS output: Connecting to srate.");
+
+ masc_strike_dc( dc );
+ masc_setup_dc( dc, 6 );
+ masc_append_dc_key_value( dc, "format", "linear" );
+ masc_append_dc_key_value( dc, "resolution", "16" );
+ masc_append_dc_key_value( dc, "sampling rate", ratestring );
+ masc_append_dc_key_value( dc, "channels", "2" );
+ masc_append_dc_key_value( dc, "endian", "host" );
+
+ mas_error = mas_asm_connect_source_sink( open_source, srate_sink, dc );
+ if ( mas_error < 0 )
+ return close_with_error("error connecting to MAS srate device");
+
+ open_source = srate_source;
+ }
+
+ // sbuf, source & sink
+ mas_error = mas_asm_instantiate_device( "sbuf", 0, 0, &sbuf );
+ if ( mas_error < 0 )
+ return close_with_error("error initantiating MAS sbuf device");
+
+ allocated_devices.push_back(sbuf);
+
+ mas_error = mas_asm_get_port_by_name( sbuf, "sink", &sbuf_sink );
+ if ( mas_error < 0 )
+ return close_with_error("error getting MAS sbuf device sink port");
+
+ allocated_ports.push_back(sbuf_sink);
+
+ mas_error = mas_asm_get_port_by_name( sbuf, "source", &sbuf_source );
+ if ( mas_error < 0 )
+ return close_with_error("error getting MAS sbuf device source port");
+
+ allocated_ports.push_back(sbuf_source);
+
+ masc_strike_dc( dc );
+ masc_setup_dc( dc, 6 );
+
+ masc_append_dc_key_value( dc, "format", "linear" );
+ masc_append_dc_key_value( dc, "resolution", "16" );
+ masc_append_dc_key_value( dc, "sampling rate", "44100" );
+ masc_append_dc_key_value( dc, "channels", "2" );
+ masc_append_dc_key_value( dc, "endian", "host" );
+
+ arts_debug("MAS output: Connecting to sbuf.");
+
+ mas_error = mas_asm_connect_source_sink( open_source, sbuf_sink, dc );
+ if ( mas_error < 0 )
+ return close_with_error("error connecting to MAS mixer device");
+
+ /* configure sbuf */
+
+ float BUFTIME_MS = _fragmentSize * _fragmentCount;
+ BUFTIME_MS *= 1000.0;
+ BUFTIME_MS /= (float)_channels;
+ if (_format > 8)
+ BUFTIME_MS /= 2.0;
+ BUFTIME_MS /= (float)_samplingRate;
+
+ arts_debug("MAS output: BUFTIME_MS = %f", BUFTIME_MS);
+
+ masc_setup_package( &package, NULL, 0, 0 );
+ masc_pushk_uint32( &package, "buftime_ms", (uint32) BUFTIME_MS );
+ masc_finalize_package( &package );
+ mas_set( sbuf, "buftime_ms", &package );
+ masc_strike_package( &package );
+
+ masc_setup_package( &package, NULL, 0, 0 );
+ masc_pushk_int32( &package, "mc_clkid", 9 );
+ masc_finalize_package( &package );
+ mas_set( sbuf, "mc_clkid", &package );
+ masc_strike_package( &package );
+
+ mas_source_play( sbuf );
+
+ // mix_sink
+ mas_error = mas_asm_get_port_by_name( 0, "default_mix_sink", &mix_sink );
+ if (mas_error < 0)
+ return close_with_error("error finding MAS default sink");
+
+ allocated_ports.push_back(mix_sink);
+
+ arts_debug("MAS output: Connecting sbuf to mix_sink.");
+
+ mas_error = mas_asm_connect_source_sink( sbuf_source, mix_sink, dc );
+ if ( mas_error < 0 )
+ return close_with_error("error connecting to MAS mixer device");
+
+ data = (mas_data *)MAS_NEW( data );
+ masc_setup_data( data, _fragmentSize ); /* we can reuse this */
+ data->length = _fragmentSize;
+ data->allocated_length = data->length;
+ data->header.type = 10;
+
+ arts_debug("MAS output: playing.");
+
+ // Install the timer
+ Dispatcher::the()->ioManager()->addTimer(10, this);
+
+ bytesPerSec = _channels * _samplingRate;
+ if (_format > 8)
+ bytesPerSec *= 2;
+
+ lastUpdate = 0;
+
+ return true;
+}
+
+double AudioIOMAS::currentTime()
+{
+ timeval tv;
+ gettimeofday(&tv,0);
+
+ return (double)tv.tv_sec + (double)tv.tv_usec/1000000.0;
+}
+
+bool AudioIOMAS::close_with_error(const string& text)
+{
+ string& error = paramStr(lastError);
+ error = text;
+ error += masc_strmerror (mas_error);
+ return false;
+}
+
+void AudioIOMAS::close()
+{
+ list<mas_port_t>::iterator pi;
+ for (pi = allocated_ports.begin(); pi != allocated_ports.end(); pi++)
+ mas_free_port (*pi);
+ allocated_ports.clear();
+
+ list<mas_channel_t>::iterator ci;
+ for (ci = allocated_channels.begin(); ci != allocated_channels.end(); ci++)
+ mas_free_channel (*ci);
+ allocated_channels.clear();
+
+ list<mas_device_t>::iterator di;
+ for (di = allocated_devices.begin(); di != allocated_devices.end(); di++)
+ {
+ mas_device_t device = *di;
+ mas_error = mas_asm_terminate_device_instance(device, 0);
+ if (mas_error < 0)
+ arts_warning ("MAS output: error while closing device: %s", masc_strmerror(mas_error));
+
+ mas_free_device(device);
+ }
+ allocated_devices.clear();
+
+ Dispatcher::the()->ioManager()->removeTimer(this);
+}
+
+void AudioIOMAS::updateBufferSizes()
+{
+ double time = currentTime();
+ double waterMark = param(fragmentSize);
+ waterMark *= 1.3;
+
+ if ((time - lastUpdate) * bytesPerSec < waterMark)
+ return;
+
+ lastUpdate = time;
+
+ uint32 inbuf_ms;
+ int32 mas_error;
+
+ mas_error = mas_get( sbuf, "inbuf_ms", 0 , &package );
+ if ( mas_error < 0 )
+ arts_fatal ("MAS output: error getting size of buffer: %s", masc_strmerror(mas_error));
+
+ masc_pull_uint32( &package, &inbuf_ms );
+ masc_strike_package( &package );
+
+ //arts_debug(" inbuf_ms = %u", inbuf_ms);
+
+ float bytes = inbuf_ms;
+ bytes /= 1000.0;
+ bytes *= param(samplingRate);
+ bytes *= param(channels);
+ if(param(format) > 8)
+ bytes *= 2;
+
+ int bytesFree = param(fragmentSize) * param(fragmentCount) - (int)bytes;
+
+ if (bytesFree < param(fragmentSize))
+ bytesFree = 0;
+
+ writeBufferAvailable = bytesFree;
+
+ arts_debug ("MAS output buffer: %6d / %6d bytes used => %6d bytes free",
+ (int)bytes, param(fragmentSize) * param(fragmentCount), writeBufferAvailable);
+}
+
+// This is called on each timer tick
+void AudioIOMAS::notifyTime()
+{
+ updateBufferSizes();
+
+ int& _direction = param(direction);
+ int& _fragmentSize = param(fragmentSize);
+
+ for (;;) {
+ int todo = 0;
+
+ if ((_direction & directionRead) && (getParam(canRead) >= _fragmentSize))
+ todo |= AudioSubSystem::ioRead;
+
+ if ((_direction & directionWrite) && (getParam(canWrite) >= _fragmentSize))
+ todo |= AudioSubSystem::ioWrite;
+
+ if (!todo)
+ return;
+
+ AudioSubSystem::the()->handleIO(todo);
+ }
+}
+
+void AudioIOMAS::setParam(AudioParam p, int& value)
+{
+ switch(p) {
+#if 0
+ case fragmentSize:
+ param(p) = requestedFragmentSize = value;
+ break;
+ case fragmentCount:
+ param(p) = requestedFragmentCount = value;
+ break;
+#endif
+ default:
+ param(p) = value;
+ break;
+ }
+}
+
+int AudioIOMAS::getParam(AudioParam p)
+{
+ int bytes;
+ int count;
+
+ switch(p)
+ {
+#if 0
+ case canRead:
+ if (ioctl(audio_fd, AUDIO_GETINFO, &auinfo) < 0)
+ return (0);
+ bytes = (auinfo.record.samples * bytesPerSample) - bytesRead;
+ if (bytes < 0) {
+ printf("Error: bytes %d < 0, samples=%u, bytesRead=%u\n",
+ bytes, auinfo.record.samples, bytesRead);
+ bytes = 0;
+ }
+ return bytes;
+
+ case canWrite:
+ if (ioctl(audio_fd, AUDIO_GETINFO, &auinfo) < 0)
+ return (0);
+ count = SUN_MAX_BUFFER_SIZE -
+ (bytesWritten - (auinfo.play.samples * bytesPerSample));
+ return count;
+#endif
+ case canWrite:
+ return writeBufferAvailable;
+
+ case autoDetect:
+ /*
+ * Fairly small priority, for we haven't tested this a lot
+ */
+ return 3;
+
+ default:
+ return param(p);
+ }
+}
+
+int AudioIOMAS::read(void *buffer, int size)
+{
+#if 0
+ size = ::read(audio_fd, buffer, size);
+ if (size < 0)
+ return 0;
+
+ bytesRead += size;
+ return size;
+#endif
+ return 0;
+}
+
+int AudioIOMAS::write(void *buffer, int size)
+{
+ static int ts = 0;
+ static int seq = 0;
+ data->header.sequence = seq++;
+ data->header.media_timestamp = ts;
+ ts += size / 4;
+
+ assert(size == data->length);
+ memcpy(data->segment, buffer, size);
+
+ int32 mas_error = mas_send( audio_channel , data );
+ if (mas_error < 0)
+ arts_fatal ("MAS output: problem during mas_send: %s", masc_strmerror(mas_error));
+
+ writeBufferAvailable -= size;
+ return size;
+}
+
+#endif /* HAVE_LIBMAS */