diff options
author | Paul Davis <paul@linuxaudiosystems.com> | 2013-07-23 22:18:22 -0400 |
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committer | Paul Davis <paul@linuxaudiosystems.com> | 2013-07-23 22:18:22 -0400 |
commit | 17588cc9be3522c806d4c9b734714e0b164b5ae2 (patch) | |
tree | 4732a396c89418b313e818e8c9cab4a6bad572c1 /libs/ardour | |
parent | 18d352bb6ef1b9b878d2bca2c400d7e221ac6640 (diff) |
pre-initial version of AudioBackend API (still under heavy development)
Diffstat (limited to 'libs/ardour')
-rw-r--r-- | libs/ardour/ardour/audio_backend.h | 242 |
1 files changed, 242 insertions, 0 deletions
diff --git a/libs/ardour/ardour/audio_backend.h b/libs/ardour/ardour/audio_backend.h new file mode 100644 index 0000000000..d286fbd9c2 --- /dev/null +++ b/libs/ardour/ardour/audio_backend.h @@ -0,0 +1,242 @@ +/* + Copyright (C) 2013 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. + +*/ + +#ifndef __ardour_audiobackend_h__ +#define __ardour_audiobackend_h__ + +#include <string> +#include <vector> + +#include <stdint.h> +#include <stdlib.h> + +namespace ARDOUR { + +class AudioEngine; + +class AudioBackend { + public: + + enum State { + Stopped = 0x1, + Running = 0x2, + Paused = 0x4, + Freewheeling = 0x8, + }; + + AudioBackend (AudioEngine& e) : engine (e), _state (Stopped) {} + virtual ~AudioBackend () {} + + /** return true if the underlying mechanism/API is still available + * for us to utilize. return false if some or all of the AudioBackend + * API can no longer be effectively used. + */ + virtual bool connected() const = 0; + + /** return true if the callback from the underlying mechanism/API + * (CoreAudio, JACK, ASIO etc.) occurs in a thread subject to realtime + * constraints. Return false otherwise. + */ + virtual bool is_realtime () const = 0; + + /* Discovering devices and parameters */ + + /** Returns a collection of strings identifying devices known + * to this backend. Any of these strings may be used to identify a + * device in other calls to the backend, though any of them may become + * invalid at any time. + */ + virtual std::vector<std::string> enumerate_devices () const = 0; + /** Returns a collection of float identifying sample rates that are + * potentially usable with the hardware identified by @param device. + * Any of these values may be supplied in other calls to this backend + * as the desired sample rate to use with the name device, but the + * requested sample rate may turn out to be unavailable, or become invalid + * at any time. + */ + virtual std::vector<float> available_sample_rates (const std::string& device) const = 0; + /** Returns a collection of uint32 identifying buffer sizes that are + * potentially usable with the hardware identified by @param device. + * Any of these values may be supplied in other calls to this backend + * as the desired buffer size to use with the name device, but the + * requested buffer size may turn out to be unavailable, or become invalid + * at any time. + */ + virtual std::vector<uint32_t> available_buffer_sizes (const std::string& device) const = 0; + + struct Parameters { + std::string device_name; + float sample_rate; + uint32_t buffer_size; + uint32_t systemic_input_latency; + uint32_t systemic_output_latency; + uint32_t input_channels; + uint32_t output_channels; + }; + + virtual int set_parameters (const Parameters&) = 0; + virtual int get_parameters (Parameters&) const = 0; + + /* Basic state control */ + + /** Start using the device named in the most recent call + * to set_parameters(), with the parameters also provided + * to that call. + * + * At some undetermined time after this function is successfully called, + * the backend will start calling the ::process_callback() method of + * the AudioEngine referenced by @param engine. These calls will + * occur in a thread created by and/or under the control of the backend. + * + * Return zero if successful, negative values otherwise. + */ + virtual int start () = 0; + + /** Stop using the device named in the most recent call to set_parameters(). + * + * If the function is successfully called, no subsequent calls to the + * process_callback() of @param engine will be made after the function + * returns, until set_parameters() and start() are called again. + * + * The backend is considered to be un-configured after a successful + * return, and requires a call to set_parameters() before it can be + * start()-ed again. See pause() for a way to avoid this. stop() should + * only be used when reconfiguration is required OR when there are no + * plans to use the backend in the future with a reconfiguration. + * + * Return zero if successful, 1 if the device is not in use, negative values on error + */ + virtual int stop () = 0; + + /** Temporarily cease using the device named in the most recent call to set_parameters(). + * + * If the function is successfully called, no subsequent calls to the + * process_callback() of @param engine will be made after the function + * returns, until start() is called again. + * + * The backend will retain its existing parameter configuration after a successful + * return, and requires a call to set_parameters() before it can be + * start()-ed again. See pause() for a way to avoid this. stop() should + * only be used when reconfiguration is required OR when there are no + * plans to use the backend in the future with a reconfiguration. + * + * Return zero if successful, 1 if the device is not in use, negative values on error + */ + virtual int pause () = 0; + + /** While remaining connected to the device, and without changing its + * configuration, start (or stop) calling the process_callback() of @param engine + * without waiting for the device. + * + * If @param start_stop is true, begin this behaviour, otherwise cease this + * behaviour if it currently occuring, and return to calling + * process_callback() of @param engine by waiting for the device. + * + * Return zero on success, non-zero otherwise. + */ + virtual int freewheel (bool start_stop) = 0; + + /** return the fraction of the time represented by the current buffer + * size that is being used for each buffer process cycle, as a value + * from 0.0 to 1.0 + */ + virtual float get_cpu_load() const = 0; + + /* Transport Control (JACK is the only audio API that currently offers + the concept of shared transport control) + */ + + /** Attempt to change the transport state to TransportRolling. + */ + virtual void transport_start () {} + /** Attempt to change the transport state to TransportStopped. + */ + virtual void transport_stop () {} + /** return the current transport state + */ + virtual TransportState transport_state () { return TransportStopped; } + /** Attempt to locate the transport to @param pos + */ + virtual void transport_locate (framepos_t pos) {} + /** Return the current transport location, in samples measured + * from the origin (defined by the transport time master) + */ + virtual framepos_t transport_frame() { return 0; } + + virtual framecnt_t sample_rate () const; + virtual pframes_t samples_per_cycle () const; + virtual int usecs_per_cycle () const { return _usecs_per_cycle; } + virtual size_t raw_buffer_size (DataType t); + + /* Process time */ + + /** return the time according to the sample clock in use, measured in + * samples since an arbitrary zero time in the past. The value should + * increase monotonically and linearly, without interruption from any + * source (including CPU frequency scaling). + * + * It is extremely likely that any implementation will use a DLL, since + * this function can be called from any thread, at any time, and must be + * able to accurately determine the correct sample time. + */ + virtual pframes_t sample_time () = 0; + + /** return the time according to the sample clock in use when the current + * buffer process cycle began. + * + * Can ONLY be called from within a process() callback tree (which + * implies that it can only be called by a process thread) + */ + virtual pframes_t sample_time_at_cycle_start () = 0; + + /** return the time since the current buffer process cycle started, + * in samples, according to the sample clock in use. + * + * Can ONLY be called from within a process() callback tree (which + * implies that it can only be called by a process thread) + */ + virtual pframes_t samples_since_cycle_start () = 0; + + /** return true if it possible to determine the offset in samples of the + * first video frame that starts within the current buffer process cycle, + * measured from the first sample of the cycle. If returning true, + * set @param offset to that offset. + * + * Eg. if it can be determined that the first video frame within the cycle + * starts 28 samples after the first sample of the cycle, then this method + * should return true and set @param offset to 28. + * + * May be impossible to support outside of JACK, which has specific support + * (in some cases, hardware support) for this feature. + * + * Can ONLY be called from within a process() callback tree (which implies + * that it can only be called by a process thread) + */ + virtual bool get_sync_offset (pframes_t& offset) const { return 0; } + + private: + AudioEngine& engine; + Parameters _last_requested_parameters; + State _state; +}; + +} + +#endif /* __ardour_audiobackend_h__ */ + |