2 Copyright (C) 2013 Paul Davis
4 This program is free software; you can redistribute it and/or modify
5 it under the terms of the GNU General Public License as published by
6 the Free Software Foundation; either version 2 of the License, or
7 (at your option) any later version.
9 This program is distributed in the hope that it will be useful,
10 but WITHOUT ANY WARRANTY; without even the implied warranty of
11 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 GNU General Public License for more details.
14 You should have received a copy of the GNU General Public License
15 along with this program; if not, write to the Free Software
16 Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
20 #ifndef __libardour_audiobackend_h__
21 #define __libardour_audiobackend_h__
29 #include <boost/function.hpp>
31 #include "ardour/types.h"
42 AudioBackend (AudioEngine& e) : engine (e){}
43 virtual ~AudioBackend () {}
45 /** Return the name of this backend.
47 * Should use a well-known, unique term. Expected examples
48 * might include "JACK", "CoreAudio", "ASIO" etc.
50 virtual std::string name() const = 0;
52 /** Return a private, type-free pointer to any data
53 * that might be useful to a concrete implementation
55 virtual void* private_handle() const = 0;
57 /** Return true if the underlying mechanism/API is still available
58 * for us to utilize. return false if some or all of the AudioBackend
59 * API can no longer be effectively used.
61 virtual bool connected() const = 0;
63 /** Return true if the callback from the underlying mechanism/API
64 * (CoreAudio, JACK, ASIO etc.) occurs in a thread subject to realtime
65 * constraints. Return false otherwise.
67 virtual bool is_realtime () const = 0;
69 /* Discovering devices and parameters */
71 /** Returns a collection of strings identifying devices known
72 * to this backend. Any of these strings may be used to identify a
73 * device in other calls to the backend, though any of them may become
74 * invalid at any time.
76 virtual std::vector<std::string> enumerate_devices () const = 0;
77 /** Returns a collection of float identifying sample rates that are
78 * potentially usable with the hardware identified by @param device.
79 * Any of these values may be supplied in other calls to this backend
80 * as the desired sample rate to use with the name device, but the
81 * requested sample rate may turn out to be unavailable, or become invalid
84 virtual std::vector<float> available_sample_rates (const std::string& device) const = 0;
85 /** Returns a collection of uint32 identifying buffer sizes that are
86 * potentially usable with the hardware identified by @param device.
87 * Any of these values may be supplied in other calls to this backend
88 * as the desired buffer size to use with the name device, but the
89 * requested buffer size may turn out to be unavailable, or become invalid
92 virtual std::vector<uint32_t> available_buffer_sizes (const std::string& device) const = 0;
94 /** Returns the maximum number of input channels that are potentially
95 * usable with the hardware identified by @param device. Any number from 1
96 * to the value returned may be supplied in other calls to this backend as
97 * the input channel count to use with the name device, but the requested
98 * count may turn out to be unavailable, or become invalid at any time.
100 virtual uint32_t available_input_channel_count (const std::string& device) const = 0;
102 /** Returns the maximum number of output channels that are potentially
103 * usable with the hardware identified by @param device. Any number from 1
104 * to the value returned may be supplied in other calls to this backend as
105 * the output channel count to use with the name device, but the requested
106 * count may turn out to be unavailable, or become invalid at any time.
108 virtual uint32_t available_output_channel_count (const std::string& device) const = 0;
110 /* Set the hardware parameters.
112 * If called when the current state is stopped or paused,
113 * the changes will not take effect until the state changes to running.
115 * If called while running, the state will change as fast as the
116 * implementation allows.
118 * All set_*() methods return zero on success, non-zero otherwise.
121 /** Set the name of the device to be used
123 virtual int set_device_name (const std::string&) = 0;
124 /** Set the sample rate to be used
126 virtual int set_sample_rate (float) = 0;
127 /** Set the buffer size to be used.
129 * The device is assumed to use a double buffering scheme, so that one
130 * buffer's worth of data can be processed by hardware while software works
131 * on the other buffer. All known suitable audio APIs support this model
132 * (though ALSA allows for alternate numbers of buffers, and CoreAudio
133 * doesn't directly expose the concept).
135 virtual int set_buffer_size (uint32_t) = 0;
136 /** Set the preferred underlying hardware sample format
138 * This does not change the sample format (32 bit float) read and
139 * written to the device via the Port API.
141 virtual int set_sample_format (SampleFormat) = 0;
142 /** Set the preferred underlying hardware data layout.
143 * If @param yn is true, then the hardware will interleave
144 * samples for successive channels; otherwise, the hardware will store
145 * samples for a single channel contiguously.
147 * Setting this does not change the fact that all data streams
148 * to and from Ports are mono (essentially, non-interleaved)
150 virtual int set_interleaved (bool yn) = 0;
151 /** Set the number of input channels that should be used
153 virtual int set_input_channels (uint32_t) = 0;
154 /** Set the number of output channels that should be used
156 virtual int set_output_channels (uint32_t) = 0;
157 /** Set the (additional) input latency that cannot be determined via
158 * the implementation's underlying code (e.g. latency from
159 * external D-A/D-A converters. Units are samples.
161 virtual int set_systemic_input_latency (uint32_t) = 0;
162 /** Set the (additional) output latency that cannot be determined via
163 * the implementation's underlying code (e.g. latency from
164 * external D-A/D-A converters. Units are samples.
166 virtual int set_systemic_output_latency (uint32_t) = 0;
168 /* Retrieving parameters */
170 virtual std::string device_name () const = 0;
171 virtual float sample_rate () const = 0;
172 virtual uint32_t buffer_size () const = 0;
173 virtual SampleFormat sample_format () const = 0;
174 virtual bool interleaved () const = 0;
175 virtual uint32_t input_channels () const = 0;
176 virtual uint32_t output_channels () const = 0;
177 virtual uint32_t systemic_input_latency () const = 0;
178 virtual uint32_t systemic_output_latency () const = 0;
180 /* Basic state control */
182 /** Start using the device named in the most recent call
183 * to set_device(), with the parameters set by various
184 * the most recent calls to set_sample_rate() etc. etc.
186 * At some undetermined time after this function is successfully called,
187 * the backend will start calling the ::process_callback() method of
188 * the AudioEngine referenced by @param engine. These calls will
189 * occur in a thread created by and/or under the control of the backend.
191 * Return zero if successful, negative values otherwise.
193 virtual int start () = 0;
195 /** Stop using the device currently in use.
197 * If the function is successfully called, no subsequent calls to the
198 * process_callback() of @param engine will be made after the function
199 * returns, until parameters are reset and start() are called again.
201 * The backend is considered to be un-configured after a successful
202 * return, and requires calls to set hardware parameters before it can be
203 * start()-ed again. See pause() for a way to avoid this. stop() should
204 * only be used when reconfiguration is required OR when there are no
205 * plans to use the backend in the future with a reconfiguration.
207 * Return zero if successful, 1 if the device is not in use, negative values on error
209 virtual int stop () = 0;
211 /** Temporarily cease using the device named in the most recent call to set_parameters().
213 * If the function is successfully called, no subsequent calls to the
214 * process_callback() of @param engine will be made after the function
215 * returns, until start() is called again.
217 * The backend will retain its existing parameter configuration after a successful
218 * return, and does NOT require any calls to set hardware parameters before it can be
221 * Return zero if successful, 1 if the device is not in use, negative values on error
223 virtual int pause () = 0;
225 /** While remaining connected to the device, and without changing its
226 * configuration, start (or stop) calling the process_callback() of @param engine
227 * without waiting for the device. Once process_callback() has returned, it
228 * will be called again immediately, thus allowing for faster-than-realtime
231 * All registered ports remain in existence and all connections remain
232 * unaltered. However, any physical ports should NOT be used by the
233 * process_callback() during freewheeling - the data behaviour is undefined.
235 * If @param start_stop is true, begin this behaviour; otherwise cease this
236 * behaviour if it currently occuring, and return to calling
237 * process_callback() of @param engine by waiting for the device.
239 * Return zero on success, non-zero otherwise.
241 virtual int freewheel (bool start_stop) = 0;
243 /** return the fraction of the time represented by the current buffer
244 * size that is being used for each buffer process cycle, as a value
247 * E.g. if the buffer size represents 5msec and current processing
248 * takes 1msec, the returned value should be 0.2.
250 * Implementations can feel free to smooth the values returned over
251 * time (e.g. high pass filtering, or its equivalent).
253 virtual float cpu_load() const = 0;
255 /* Transport Control (JACK is the only audio API that currently offers
256 the concept of shared transport control)
259 /** Attempt to change the transport state to TransportRolling.
261 virtual void transport_start () {}
262 /** Attempt to change the transport state to TransportStopped.
264 virtual void transport_stop () {}
265 /** return the current transport state
267 virtual TransportState transport_state () const { return TransportStopped; }
268 /** Attempt to locate the transport to @param pos
270 virtual void transport_locate (framepos_t /*pos*/) {}
271 /** Return the current transport location, in samples measured
272 * from the origin (defined by the transport time master)
274 virtual framepos_t transport_frame() const { return 0; }
276 /** If @param yn is true, become the time master for any inter-application transport
277 * timebase, otherwise cease to be the time master for the same.
279 * Return zero on success, non-zero otherwise
281 * JACK is the only currently known audio API with the concept of a shared
282 * transport timebase.
284 virtual int set_time_master (bool /*yn*/) { return 0; }
286 virtual int usecs_per_cycle () const { return 1000000 * (buffer_size() / sample_rate()); }
287 virtual size_t raw_buffer_size (DataType t);
291 /** return the time according to the sample clock in use, measured in
292 * samples since an arbitrary zero time in the past. The value should
293 * increase monotonically and linearly, without interruption from any
294 * source (including CPU frequency scaling).
296 * It is extremely likely that any implementation will use a DLL, since
297 * this function can be called from any thread, at any time, and must be
298 * able to accurately determine the correct sample time.
300 virtual pframes_t sample_time () = 0;
302 /** return the time according to the sample clock in use when the current
303 * buffer process cycle began.
305 * Can ONLY be called from within a process() callback tree (which
306 * implies that it can only be called by a process thread)
308 virtual pframes_t sample_time_at_cycle_start () = 0;
310 /** return the time since the current buffer process cycle started,
311 * in samples, according to the sample clock in use.
313 * Can ONLY be called from within a process() callback tree (which
314 * implies that it can only be called by a process thread)
316 virtual pframes_t samples_since_cycle_start () = 0;
318 /** return true if it possible to determine the offset in samples of the
319 * first video frame that starts within the current buffer process cycle,
320 * measured from the first sample of the cycle. If returning true,
321 * set @param offset to that offset.
323 * Eg. if it can be determined that the first video frame within the cycle
324 * starts 28 samples after the first sample of the cycle, then this method
325 * should return true and set @param offset to 28.
327 * May be impossible to support outside of JACK, which has specific support
328 * (in some cases, hardware support) for this feature.
330 * Can ONLY be called from within a process() callback tree (which implies
331 * that it can only be called by a process thread)
333 virtual bool get_sync_offset (pframes_t& /*offset*/) const { return false; }
335 /** Create a new thread suitable for running part of the buffer process
336 * cycle (i.e. Realtime scheduling, memory allocation, etc. etc are all
337 * correctly setup), with a stack size given in bytes by specified @param
338 * stacksize. The thread will begin executing @param func, and will exit
339 * when that function returns.
341 virtual int create_process_thread (boost::function<void()> func, pthread_t*, size_t stacksize) = 0;
347 struct AudioBackendInfo {
350 int (*instantiate) (const std::string& arg1, const std::string& arg2);
351 int (*deinstantiate) (void);
353 boost::shared_ptr<AudioBackend> (*backend_factory) (AudioEngine&);
354 boost::shared_ptr<PortEngine> (*portengine_factory) (PortManager&);
359 #endif /* __libardour_audiobackend_h__ */