drivers/audio/AudioOutputDeviceAlsa.cpp

/***************************************************************************
 *                                                                         *
 *   LinuxSampler - modular, streaming capable sampler                     *
 *                                                                         *
 *   Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck   *
 *                                                                         *
 *   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., 59 Temple Place, Suite 330, Boston,                 *
 *   MA  02111-1307  USA                                                   *
 ***************************************************************************/

#include "../InputOutputDevice.h"
#include "AudioOutputDeviceAlsa.h"
#include "AudioOutputDeviceFactory.h"

namespace LinuxSampler {

    REGISTER_AUDIO_OUTPUT_DRIVER(AudioOutputDeviceAlsa);

    /* Common parameters for now they'll have to be registered here. */
    REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterActive);
    REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterSampleRate);
    REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterChannels);

    /* Driver specific parameters */
    REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterCard);
    REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterFragments);
    REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterFragmentSize);

    /**
     * Create and initialize Alsa audio output device with given parameters.
     *
     * @param Parameters - optional parameters
     * @throws AudioOutputException  if output device cannot be opened
     */
    AudioOutputDeviceAlsa::AudioOutputDeviceAlsa(std::map<String,DeviceCreationParameter*> Parameters) : AudioOutputDevice(Parameters), Thread(true, 1, 0) {
        pcm_handle           = NULL;
        stream               = SND_PCM_STREAM_PLAYBACK;
        this->uiAlsaChannels = ((DeviceCreationParameterInt*)Parameters["channels"])->ValueAsInt();
        this->uiSamplerate   = ((DeviceCreationParameterInt*)Parameters["samplerate"])->ValueAsInt();
        this->FragmentSize   = ((DeviceCreationParameterInt*)Parameters["fragmentsize"])->ValueAsInt();
        uint Fragments       = ((DeviceCreationParameterInt*)Parameters["fragments"])->ValueAsInt();
        String Card          = Parameters["card"]->Value();

        dmsg(1,("Checking if hw parameters supported...\n"));
        if (HardwareParametersSupported(Card, uiAlsaChannels, uiSamplerate, Fragments, FragmentSize)) {
            pcm_name = "hw:" + Card;
        }
        else {
            printf("Warning: your soundcard doesn't support chosen hardware parameters; ");
            printf("trying to compensate support lack with plughw...");
            fflush(stdout);
            pcm_name = "plughw:" + Card;
        }
        dmsg(1,("HW check completed.\n"));

        int err;

        snd_pcm_hw_params_alloca(&hwparams);  // Allocate the snd_pcm_hw_params_t structure on the stack.

        /* Open PCM. The last parameter of this function is the mode. */
        /* If this is set to 0, the standard mode is used. Possible   */
        /* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC.       */
        /* If SND_PCM_NONBLOCK is used, read / write access to the    */
        /* PCM device will return immediately. If SND_PCM_ASYNC is    */
        /* specified, SIGIO will be emitted whenever a period has     */
        /* been completely processed by the soundcard.                */
        if ((err = snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0)) < 0) {
            throw AudioOutputException(String("Error opening PCM device ") + pcm_name + ": " + snd_strerror(err));
        }

        if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) {
            throw AudioOutputException(String("Error, cannot initialize hardware parameter structure: ") + snd_strerror(err));
        }

        /* Set access type. This can be either    */
        /* SND_PCM_ACCESS_RW_INTERLEAVED or       */
        /* SND_PCM_ACCESS_RW_NONINTERLEAVED.      */
        if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
            throw AudioOutputException(String("Error snd_pcm_hw_params_set_access: ") + snd_strerror(err));
        }

        /* Set sample format */
        #if WORDS_BIGENDIAN
        if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE)) < 0)
        #else // little endian
        if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
        #endif
        {
            throw AudioOutputException(String("Error setting sample format: ") + snd_strerror(err));
        }

        int dir = 0;

        /* Set sample rate. If the exact rate is not supported */
        /* by the hardware, use nearest possible rate.         */
        #if ALSA_MAJOR > 0
        if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &uiSamplerate, &dir)) < 0)
        #else
        if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, uiSamplerate, &dir)) < 0)
        #endif
        {
            throw AudioOutputException(String("Error setting sample rate: ") + snd_strerror(err));
        }

        if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, uiAlsaChannels)) < 0) {
            throw AudioOutputException(String("Error setting number of channels: ") + snd_strerror(err));
        }

        /* Set number of periods. Periods used to be called fragments. */
        if ((err = snd_pcm_hw_params_set_periods(pcm_handle, hwparams, Fragments, dir)) < 0) {
            throw AudioOutputException(String("Error setting number of periods: ") + snd_strerror(err));
        }

        /* Set buffer size (in frames). The resulting latency is given by */
        /* latency = periodsize * periods / (rate * bytes_per_frame)     */
        if ((err = snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (FragmentSize * Fragments))) < 0) {
            throw AudioOutputException(String("Error setting buffersize: ") + snd_strerror(err));
        }

        /* Apply HW parameter settings to */
        /* PCM device and prepare device  */
        if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) {
            throw AudioOutputException(String("Error setting HW params: ") + snd_strerror(err));
        }

        if (snd_pcm_sw_params_malloc(&swparams) != 0) {
            throw AudioOutputException(String("Error in snd_pcm_sw_params_malloc: ") + snd_strerror(err));
        }

        if (snd_pcm_sw_params_current(pcm_handle, swparams) != 0) {
            throw AudioOutputException(String("Error in snd_pcm_sw_params_current: ") + snd_strerror(err));
        }

        if (snd_pcm_sw_params_set_stop_threshold(pcm_handle, swparams, 0xffffffff) != 0) {
            throw AudioOutputException(String("Error in snd_pcm_sw_params_set_stop_threshold: ") + snd_strerror(err));
        }

        if (snd_pcm_sw_params(pcm_handle, swparams) != 0) {
            throw AudioOutputException(String("Error in snd_pcm_sw_params: ") + snd_strerror(err));
        }

        if ((err = snd_pcm_prepare(pcm_handle)) < 0) {
            throw AudioOutputException(String("Error snd_pcm_prepare: ") + snd_strerror(err));
        }

        // allocate Alsa output buffer
        pAlsaOutputBuffer = new int16_t[uiAlsaChannels * FragmentSize];

        // create audio channels for this audio device to which the sampler engines can write to
        for (int i = 0; i < uiAlsaChannels; i++) this->Channels.push_back(new AudioChannel(FragmentSize));

        if (((DeviceCreationParameterBool*)Parameters["active"])->ValueAsBool()) {
                Play();
        }
    }

    AudioOutputDeviceAlsa::~AudioOutputDeviceAlsa() {
        //dmsg(0,("Stopping Alsa Thread..."));
        //StopThread();  //FIXME: commented out due to a bug in thread.cpp (StopThread() doesn't return at all)
        //dmsg(0,("OK\n"));

        //FIXME: currently commented out due to segfault
        //snd_pcm_close(pcm_handle);

        // destroy all audio channels
        for (int c = 0; c < Channels.size(); c++) delete Channels[c];

        if (pAlsaOutputBuffer) {
            //FIXME: currently commented out due to segfault
            //delete[] pOutputBuffer;
        }
    }

    /**
     *  Checks if sound card supports the chosen parameters.
     *
     *  @returns  true if hardware supports it
     */
    bool AudioOutputDeviceAlsa::HardwareParametersSupported(String card, uint channels, int samplerate, uint numfragments, uint fragmentsize) {
        pcm_name = "hw:" + card;
        if (snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0) < 0) return false;
        snd_pcm_hw_params_alloca(&hwparams);
        if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) {
            snd_pcm_close(pcm_handle);
            return false;
        }
        if (snd_pcm_hw_params_test_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
            snd_pcm_close(pcm_handle);
            return false;
        }
        #if WORDS_BIGENDIAN
        if (snd_pcm_hw_params_test_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE) < 0)
        #else // little endian
        if (snd_pcm_hw_params_test_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE) < 0)
        #endif
        {
            snd_pcm_close(pcm_handle);
            return false;
        }
        int dir = 0;
        if (snd_pcm_hw_params_test_rate(pcm_handle, hwparams, samplerate, dir) < 0) {
            snd_pcm_close(pcm_handle);
            return false;
        }
        if (snd_pcm_hw_params_test_channels(pcm_handle, hwparams, channels) < 0) {
            snd_pcm_close(pcm_handle);
            return false;
        }
        if (snd_pcm_hw_params_test_periods(pcm_handle, hwparams, numfragments, dir) < 0) {
            snd_pcm_close(pcm_handle);
            return false;
        }
        if (snd_pcm_hw_params_test_buffer_size(pcm_handle, hwparams, (fragmentsize * numfragments)) < 0) {
            snd_pcm_close(pcm_handle);
            return false;
        }

        snd_pcm_close(pcm_handle);
        return true;
    }

    void AudioOutputDeviceAlsa::Play() {
        StartThread();
    }

    bool AudioOutputDeviceAlsa::IsPlaying() {
        return IsRunning(); // if Thread is running
    }

    void AudioOutputDeviceAlsa::Stop() {
        StopThread();
    }

    void AudioOutputDeviceAlsa::AcquireChannels(uint Channels) {
        if (Channels > uiAlsaChannels) {
            // just create mix channel(s)
            for (int i = uiAlsaChannels; i < Channels; i++) {
                AudioChannel* pNewChannel = new AudioChannel(this->Channels[i % uiAlsaChannels]);
                this->Channels.push_back(pNewChannel);
            }
        }
    }

    uint AudioOutputDeviceAlsa::MaxSamplesPerCycle() {
        return FragmentSize;
    }

    uint AudioOutputDeviceAlsa::SampleRate() {
        return uiSamplerate;
    }

    String AudioOutputDeviceAlsa::Name() {
        return "Alsa";
    }

    String AudioOutputDeviceAlsa::Driver() {
        return Name();
    }

    String AudioOutputDeviceAlsa::Description() {
        return "Advanced Linux Sound Architecture";
    }

    String AudioOutputDeviceAlsa::Version() {
       String s = "$Revision: 1.9 $";
       return s.substr(11, s.size() - 13); // cut dollar signs, spaces and CVS macro keyword
    }

    /**
     * Entry point for the thread.
     */
    int AudioOutputDeviceAlsa::Main() {
        while (true) {

            // let all connected engines render 'FragmentSize' sample points
            RenderAudio(FragmentSize);

            // convert from DSP value range (-1.0..+1.0) to 16 bit integer value
            // range (-32768..+32767), check clipping  and copy to Alsa output buffer
            // (note: we use interleaved output method to Alsa)
            for (int c = 0; c < uiAlsaChannels; c++) {
                float* in  = Channels[c]->Buffer();
                for (int i = 0, o = c; i < FragmentSize; i++ , o += uiAlsaChannels) {
                    float sample_point = in[i] * 32768.0f;
                    if (sample_point < -32768.0) sample_point = -32768.0;
                    if (sample_point >  32767.0) sample_point =  32767.0;
                    pAlsaOutputBuffer[o] = (int16_t) sample_point;
                }
            }

            // output sound
            int res = Output();
            if (res < 0) {
                fprintf(stderr, "Alsa: Audio output error, exiting.\n");
                exit(EXIT_FAILURE);
            }
        }
    }

    /**
     *  Will be called after every audio fragment cycle, to output the audio data
     *  of the current fragment to the soundcard.
     *
     *  @returns  0 on success, a value < 0 on error
     */
    int AudioOutputDeviceAlsa::Output() {
        int err = snd_pcm_writei(pcm_handle, pAlsaOutputBuffer, FragmentSize);
        if (err < 0) {
            fprintf(stderr, "Error snd_pcm_writei failed: %s\n", snd_strerror(err));
            return -1;
        }
        return 0;
    }

} // namespace LinuxSampler

1	schoenebeck	200	/***************************************************************************
2			* *
3			* LinuxSampler - modular, streaming capable sampler *
4			* *
5			* Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck *
6			* *
7			* This program is free software; you can redistribute it and/or modify *
8			* it under the terms of the GNU General Public License as published by *
9			* the Free Software Foundation; either version 2 of the License, or *
10			* (at your option) any later version. *
11			* *
12			* This program is distributed in the hope that it will be useful, *
13			* but WITHOUT ANY WARRANTY; without even the implied warranty of *
14			* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the *
15			* GNU General Public License for more details. *
16			* *
17			* You should have received a copy of the GNU General Public License *
18			* along with this program; if not, write to the Free Software *
19			* Foundation, Inc., 59 Temple Place, Suite 330, Boston, *
20			* MA 02111-1307 USA *
21			***************************************************************************/
22
23			#include "../InputOutputDevice.h"
24			#include "AudioOutputDeviceAlsa.h"
25			#include "AudioOutputDeviceFactory.h"
26
27			namespace LinuxSampler {
28
29			REGISTER_AUDIO_OUTPUT_DRIVER(AudioOutputDeviceAlsa);
30
31			/* Common parameters for now they'll have to be registered here. */
32			REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterActive);
33			REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterSampleRate);
34			REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterChannels);
35
36			/* Driver specific parameters */
37			REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterCard);
38			REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterFragments);
39			REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterFragmentSize);
40
41			/**
42			* Create and initialize Alsa audio output device with given parameters.
43			*
44			* @param Parameters - optional parameters
45			* @throws AudioOutputException if output device cannot be opened
46			*/
47			AudioOutputDeviceAlsa::AudioOutputDeviceAlsa(std::map<String,DeviceCreationParameter*> Parameters) : AudioOutputDevice(Parameters), Thread(true, 1, 0) {
48			pcm_handle = NULL;
49			stream = SND_PCM_STREAM_PLAYBACK;
50			this->uiAlsaChannels = ((DeviceCreationParameterInt*)Parameters["channels"])->ValueAsInt();
51			this->uiSamplerate = ((DeviceCreationParameterInt*)Parameters["samplerate"])->ValueAsInt();
52			this->FragmentSize = ((DeviceCreationParameterInt*)Parameters["fragmentsize"])->ValueAsInt();
53			uint Fragments = ((DeviceCreationParameterInt*)Parameters["fragments"])->ValueAsInt();
54			String Card = Parameters["card"]->Value();
55
56			dmsg(1,("Checking if hw parameters supported...\n"));
57			if (HardwareParametersSupported(Card, uiAlsaChannels, uiSamplerate, Fragments, FragmentSize)) {
58			pcm_name = "hw:" + Card;
59			}
60			else {
61			printf("Warning: your soundcard doesn't support chosen hardware parameters; ");
62			printf("trying to compensate support lack with plughw...");
63			fflush(stdout);
64			pcm_name = "plughw:" + Card;
65			}
66			dmsg(1,("HW check completed.\n"));
67
68			int err;
69
70			snd_pcm_hw_params_alloca(&hwparams); // Allocate the snd_pcm_hw_params_t structure on the stack.
71
72			/* Open PCM. The last parameter of this function is the mode. */
73			/* If this is set to 0, the standard mode is used. Possible */
74			/* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC. */
75			/* If SND_PCM_NONBLOCK is used, read / write access to the */
76			/* PCM device will return immediately. If SND_PCM_ASYNC is */
77			/* specified, SIGIO will be emitted whenever a period has */
78			/* been completely processed by the soundcard. */
79			if ((err = snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0)) < 0) {
80			throw AudioOutputException(String("Error opening PCM device ") + pcm_name + ": " + snd_strerror(err));
81			}
82
83			if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) {
84			throw AudioOutputException(String("Error, cannot initialize hardware parameter structure: ") + snd_strerror(err));
85			}
86
87			/* Set access type. This can be either */
88			/* SND_PCM_ACCESS_RW_INTERLEAVED or */
89			/* SND_PCM_ACCESS_RW_NONINTERLEAVED. */
90			if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
91			throw AudioOutputException(String("Error snd_pcm_hw_params_set_access: ") + snd_strerror(err));
92			}
93
94			/* Set sample format */
95			#if WORDS_BIGENDIAN
96			if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE)) < 0)
97			#else // little endian
98			if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
99			#endif
100			{
101			throw AudioOutputException(String("Error setting sample format: ") + snd_strerror(err));
102			}
103
104			int dir = 0;
105
106			/* Set sample rate. If the exact rate is not supported */
107			/* by the hardware, use nearest possible rate. */
108			#if ALSA_MAJOR > 0
109			if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &uiSamplerate, &dir)) < 0)
110			#else
111			if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, uiSamplerate, &dir)) < 0)
112			#endif
113			{
114			throw AudioOutputException(String("Error setting sample rate: ") + snd_strerror(err));
115			}
116
117			if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, uiAlsaChannels)) < 0) {
118			throw AudioOutputException(String("Error setting number of channels: ") + snd_strerror(err));
119			}
120
121			/* Set number of periods. Periods used to be called fragments. */
122			if ((err = snd_pcm_hw_params_set_periods(pcm_handle, hwparams, Fragments, dir)) < 0) {
123			throw AudioOutputException(String("Error setting number of periods: ") + snd_strerror(err));
124			}
125
126			/* Set buffer size (in frames). The resulting latency is given by */
127			/* latency = periodsize * periods / (rate * bytes_per_frame) */
128			if ((err = snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (FragmentSize * Fragments))) < 0) {
129			throw AudioOutputException(String("Error setting buffersize: ") + snd_strerror(err));
130			}
131
132			/* Apply HW parameter settings to */
133			/* PCM device and prepare device */
134			if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) {
135			throw AudioOutputException(String("Error setting HW params: ") + snd_strerror(err));
136			}
137
138			if (snd_pcm_sw_params_malloc(&swparams) != 0) {
139			throw AudioOutputException(String("Error in snd_pcm_sw_params_malloc: ") + snd_strerror(err));
140			}
141
142			if (snd_pcm_sw_params_current(pcm_handle, swparams) != 0) {
143			throw AudioOutputException(String("Error in snd_pcm_sw_params_current: ") + snd_strerror(err));
144			}
145
146			if (snd_pcm_sw_params_set_stop_threshold(pcm_handle, swparams, 0xffffffff) != 0) {
147			throw AudioOutputException(String("Error in snd_pcm_sw_params_set_stop_threshold: ") + snd_strerror(err));
148			}
149
150			if (snd_pcm_sw_params(pcm_handle, swparams) != 0) {
151			throw AudioOutputException(String("Error in snd_pcm_sw_params: ") + snd_strerror(err));
152			}
153
154			if ((err = snd_pcm_prepare(pcm_handle)) < 0) {
155			throw AudioOutputException(String("Error snd_pcm_prepare: ") + snd_strerror(err));
156			}
157
158			// allocate Alsa output buffer
159			pAlsaOutputBuffer = new int16_t[uiAlsaChannels * FragmentSize];
160
161			// create audio channels for this audio device to which the sampler engines can write to
162			for (int i = 0; i < uiAlsaChannels; i++) this->Channels.push_back(new AudioChannel(FragmentSize));
163
164			if (((DeviceCreationParameterBool*)Parameters["active"])->ValueAsBool()) {
165			Play();
166			}
167			}
168
169			AudioOutputDeviceAlsa::~AudioOutputDeviceAlsa() {
170			//dmsg(0,("Stopping Alsa Thread..."));
171			//StopThread(); //FIXME: commented out due to a bug in thread.cpp (StopThread() doesn't return at all)
172			//dmsg(0,("OK\n"));
173
174			//FIXME: currently commented out due to segfault
175			//snd_pcm_close(pcm_handle);
176
177			// destroy all audio channels
178			for (int c = 0; c < Channels.size(); c++) delete Channels[c];
179
180			if (pAlsaOutputBuffer) {
181			//FIXME: currently commented out due to segfault
182			//delete[] pOutputBuffer;
183			}
184			}
185
186			/**
187			* Checks if sound card supports the chosen parameters.
188			*
189			* @returns true if hardware supports it
190			*/
191			bool AudioOutputDeviceAlsa::HardwareParametersSupported(String card, uint channels, int samplerate, uint numfragments, uint fragmentsize) {
192			pcm_name = "hw:" + card;
193			if (snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0) < 0) return false;
194			snd_pcm_hw_params_alloca(&hwparams);
195			if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) {
196			snd_pcm_close(pcm_handle);
197			return false;
198			}
199			if (snd_pcm_hw_params_test_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
200			snd_pcm_close(pcm_handle);
201			return false;
202			}
203			#if WORDS_BIGENDIAN
204			if (snd_pcm_hw_params_test_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE) < 0)
205			#else // little endian
206			if (snd_pcm_hw_params_test_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE) < 0)
207			#endif
208			{
209			snd_pcm_close(pcm_handle);
210			return false;
211			}
212			int dir = 0;
213			if (snd_pcm_hw_params_test_rate(pcm_handle, hwparams, samplerate, dir) < 0) {
214			snd_pcm_close(pcm_handle);
215			return false;
216			}
217			if (snd_pcm_hw_params_test_channels(pcm_handle, hwparams, channels) < 0) {
218			snd_pcm_close(pcm_handle);
219			return false;
220			}
221			if (snd_pcm_hw_params_test_periods(pcm_handle, hwparams, numfragments, dir) < 0) {
222			snd_pcm_close(pcm_handle);
223			return false;
224			}
225			if (snd_pcm_hw_params_test_buffer_size(pcm_handle, hwparams, (fragmentsize * numfragments)) < 0) {
226			snd_pcm_close(pcm_handle);
227			return false;
228			}
229
230			snd_pcm_close(pcm_handle);
231			return true;
232			}
233
234			void AudioOutputDeviceAlsa::Play() {
235			StartThread();
236			}
237
238			bool AudioOutputDeviceAlsa::IsPlaying() {
239			return IsRunning(); // if Thread is running
240			}
241
242			void AudioOutputDeviceAlsa::Stop() {
243			StopThread();
244			}
245
246			void AudioOutputDeviceAlsa::AcquireChannels(uint Channels) {
247			if (Channels > uiAlsaChannels) {
248			// just create mix channel(s)
249			for (int i = uiAlsaChannels; i < Channels; i++) {
250			AudioChannel* pNewChannel = new AudioChannel(this->Channels[i % uiAlsaChannels]);
251			this->Channels.push_back(pNewChannel);
252			}
253			}
254			}
255
256			uint AudioOutputDeviceAlsa::MaxSamplesPerCycle() {
257			return FragmentSize;
258			}
259
260			uint AudioOutputDeviceAlsa::SampleRate() {
261			return uiSamplerate;
262			}
263
264			String AudioOutputDeviceAlsa::Name() {
265			return "Alsa";
266			}
267
268			String AudioOutputDeviceAlsa::Driver() {
269			return Name();
270			}
271
272			String AudioOutputDeviceAlsa::Description() {
273			return "Advanced Linux Sound Architecture";
274			}
275
276			String AudioOutputDeviceAlsa::Version() {
277			String s = "$Revision: 1.9 $";
278			return s.substr(11, s.size() - 13); // cut dollar signs, spaces and CVS macro keyword
279			}
280
281			/**
282			* Entry point for the thread.
283			*/
284			int AudioOutputDeviceAlsa::Main() {
285			while (true) {
286
287			// let all connected engines render 'FragmentSize' sample points
288			RenderAudio(FragmentSize);
289
290			// convert from DSP value range (-1.0..+1.0) to 16 bit integer value
291			// range (-32768..+32767), check clipping and copy to Alsa output buffer
292			// (note: we use interleaved output method to Alsa)
293			for (int c = 0; c < uiAlsaChannels; c++) {
294			float* in = Channels[c]->Buffer();
295			for (int i = 0, o = c; i < FragmentSize; i++ , o += uiAlsaChannels) {
296			float sample_point = in[i] * 32768.0f;
297			if (sample_point < -32768.0) sample_point = -32768.0;
298			if (sample_point > 32767.0) sample_point = 32767.0;
299			pAlsaOutputBuffer[o] = (int16_t) sample_point;
300			}
301			}
302
303			// output sound
304			int res = Output();
305			if (res < 0) {
306			fprintf(stderr, "Alsa: Audio output error, exiting.\n");
307			exit(EXIT_FAILURE);
308			}
309			}
310			}
311
312			/**
313			* Will be called after every audio fragment cycle, to output the audio data
314			* of the current fragment to the soundcard.
315			*
316			* @returns 0 on success, a value < 0 on error
317			*/
318			int AudioOutputDeviceAlsa::Output() {
319			int err = snd_pcm_writei(pcm_handle, pAlsaOutputBuffer, FragmentSize);
320			if (err < 0) {
321			fprintf(stderr, "Error snd_pcm_writei failed: %s\n", snd_strerror(err));
322			return -1;
323			}
324			return 0;
325			}
326
327			} // namespace LinuxSampler
328