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 "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: 1.10 $";
       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 "AudioOutputDeviceAlsa.h"
24			#include "AudioOutputDeviceFactory.h"
25
26			namespace LinuxSampler {
27
28			REGISTER_AUDIO_OUTPUT_DRIVER(AudioOutputDeviceAlsa);
29
30			/* Common parameters for now they'll have to be registered here. */
31			REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterActive);
32			REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterSampleRate);
33			REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterChannels);
34
35			/* Driver specific parameters */
36			REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterCard);
37			REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterFragments);
38			REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterFragmentSize);
39
40			/**
41			* Create and initialize Alsa audio output device with given parameters.
42			*
43			* @param Parameters - optional parameters
44			* @throws AudioOutputException if output device cannot be opened
45			*/
46			AudioOutputDeviceAlsa::AudioOutputDeviceAlsa(std::map<String,DeviceCreationParameter*> Parameters) : AudioOutputDevice(Parameters), Thread(true, 1, 0) {
47			pcm_handle = NULL;
48			stream = SND_PCM_STREAM_PLAYBACK;
49			this->uiAlsaChannels = ((DeviceCreationParameterInt*)Parameters["channels"])->ValueAsInt();
50			this->uiSamplerate = ((DeviceCreationParameterInt*)Parameters["samplerate"])->ValueAsInt();
51			this->FragmentSize = ((DeviceCreationParameterInt*)Parameters["fragmentsize"])->ValueAsInt();
52			uint Fragments = ((DeviceCreationParameterInt*)Parameters["fragments"])->ValueAsInt();
53			String Card = Parameters["card"]->Value();
54
55			dmsg(1,("Checking if hw parameters supported...\n"));
56			if (HardwareParametersSupported(Card, uiAlsaChannels, uiSamplerate, Fragments, FragmentSize)) {
57			pcm_name = "hw:" + Card;
58			}
59			else {
60			printf("Warning: your soundcard doesn't support chosen hardware parameters; ");
61			printf("trying to compensate support lack with plughw...");
62			fflush(stdout);
63			pcm_name = "plughw:" + Card;
64			}
65			dmsg(1,("HW check completed.\n"));
66
67			int err;
68
69			snd_pcm_hw_params_alloca(&hwparams); // Allocate the snd_pcm_hw_params_t structure on the stack.
70
71			/* Open PCM. The last parameter of this function is the mode. */
72			/* If this is set to 0, the standard mode is used. Possible */
73			/* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC. */
74			/* If SND_PCM_NONBLOCK is used, read / write access to the */
75			/* PCM device will return immediately. If SND_PCM_ASYNC is */
76			/* specified, SIGIO will be emitted whenever a period has */
77			/* been completely processed by the soundcard. */
78			if ((err = snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0)) < 0) {
79			throw AudioOutputException(String("Error opening PCM device ") + pcm_name + ": " + snd_strerror(err));
80			}
81
82			if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) {
83			throw AudioOutputException(String("Error, cannot initialize hardware parameter structure: ") + snd_strerror(err));
84			}
85
86			/* Set access type. This can be either */
87			/* SND_PCM_ACCESS_RW_INTERLEAVED or */
88			/* SND_PCM_ACCESS_RW_NONINTERLEAVED. */
89			if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
90			throw AudioOutputException(String("Error snd_pcm_hw_params_set_access: ") + snd_strerror(err));
91			}
92
93			/* Set sample format */
94			#if WORDS_BIGENDIAN
95			if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE)) < 0)
96			#else // little endian
97			if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
98			#endif
99			{
100			throw AudioOutputException(String("Error setting sample format: ") + snd_strerror(err));
101			}
102
103			int dir = 0;
104
105			/* Set sample rate. If the exact rate is not supported */
106			/* by the hardware, use nearest possible rate. */
107			#if ALSA_MAJOR > 0
108			if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &uiSamplerate, &dir)) < 0)
109			#else
110			if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, uiSamplerate, &dir)) < 0)
111			#endif
112			{
113			throw AudioOutputException(String("Error setting sample rate: ") + snd_strerror(err));
114			}
115
116			if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, uiAlsaChannels)) < 0) {
117			throw AudioOutputException(String("Error setting number of channels: ") + snd_strerror(err));
118			}
119
120			/* Set number of periods. Periods used to be called fragments. */
121			if ((err = snd_pcm_hw_params_set_periods(pcm_handle, hwparams, Fragments, dir)) < 0) {
122			throw AudioOutputException(String("Error setting number of periods: ") + snd_strerror(err));
123			}
124
125			/* Set buffer size (in frames). The resulting latency is given by */
126			/* latency = periodsize * periods / (rate * bytes_per_frame) */
127			if ((err = snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (FragmentSize * Fragments))) < 0) {
128			throw AudioOutputException(String("Error setting buffersize: ") + snd_strerror(err));
129			}
130
131			/* Apply HW parameter settings to */
132			/* PCM device and prepare device */
133			if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) {
134			throw AudioOutputException(String("Error setting HW params: ") + snd_strerror(err));
135			}
136
137			if (snd_pcm_sw_params_malloc(&swparams) != 0) {
138			throw AudioOutputException(String("Error in snd_pcm_sw_params_malloc: ") + snd_strerror(err));
139			}
140
141			if (snd_pcm_sw_params_current(pcm_handle, swparams) != 0) {
142			throw AudioOutputException(String("Error in snd_pcm_sw_params_current: ") + snd_strerror(err));
143			}
144
145			if (snd_pcm_sw_params_set_stop_threshold(pcm_handle, swparams, 0xffffffff) != 0) {
146			throw AudioOutputException(String("Error in snd_pcm_sw_params_set_stop_threshold: ") + snd_strerror(err));
147			}
148
149			if (snd_pcm_sw_params(pcm_handle, swparams) != 0) {
150			throw AudioOutputException(String("Error in snd_pcm_sw_params: ") + snd_strerror(err));
151			}
152
153			if ((err = snd_pcm_prepare(pcm_handle)) < 0) {
154			throw AudioOutputException(String("Error snd_pcm_prepare: ") + snd_strerror(err));
155			}
156
157			// allocate Alsa output buffer
158			pAlsaOutputBuffer = new int16_t[uiAlsaChannels * FragmentSize];
159
160			// create audio channels for this audio device to which the sampler engines can write to
161			for (int i = 0; i < uiAlsaChannels; i++) this->Channels.push_back(new AudioChannel(FragmentSize));
162
163			if (((DeviceCreationParameterBool*)Parameters["active"])->ValueAsBool()) {
164			Play();
165			}
166			}
167
168			AudioOutputDeviceAlsa::~AudioOutputDeviceAlsa() {
169			//dmsg(0,("Stopping Alsa Thread..."));
170			//StopThread(); //FIXME: commented out due to a bug in thread.cpp (StopThread() doesn't return at all)
171			//dmsg(0,("OK\n"));
172
173			//FIXME: currently commented out due to segfault
174			//snd_pcm_close(pcm_handle);
175
176			// destroy all audio channels
177			for (int c = 0; c < Channels.size(); c++) delete Channels[c];
178
179			if (pAlsaOutputBuffer) {
180			//FIXME: currently commented out due to segfault
181			//delete[] pOutputBuffer;
182			}
183			}
184
185			/**
186			* Checks if sound card supports the chosen parameters.
187			*
188			* @returns true if hardware supports it
189			*/
190			bool AudioOutputDeviceAlsa::HardwareParametersSupported(String card, uint channels, int samplerate, uint numfragments, uint fragmentsize) {
191			pcm_name = "hw:" + card;
192			if (snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0) < 0) return false;
193			snd_pcm_hw_params_alloca(&hwparams);
194			if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) {
195			snd_pcm_close(pcm_handle);
196			return false;
197			}
198			if (snd_pcm_hw_params_test_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
199			snd_pcm_close(pcm_handle);
200			return false;
201			}
202			#if WORDS_BIGENDIAN
203			if (snd_pcm_hw_params_test_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE) < 0)
204			#else // little endian
205			if (snd_pcm_hw_params_test_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE) < 0)
206			#endif
207			{
208			snd_pcm_close(pcm_handle);
209			return false;
210			}
211			int dir = 0;
212			if (snd_pcm_hw_params_test_rate(pcm_handle, hwparams, samplerate, dir) < 0) {
213			snd_pcm_close(pcm_handle);
214			return false;
215			}
216			if (snd_pcm_hw_params_test_channels(pcm_handle, hwparams, channels) < 0) {
217			snd_pcm_close(pcm_handle);
218			return false;
219			}
220			if (snd_pcm_hw_params_test_periods(pcm_handle, hwparams, numfragments, dir) < 0) {
221			snd_pcm_close(pcm_handle);
222			return false;
223			}
224			if (snd_pcm_hw_params_test_buffer_size(pcm_handle, hwparams, (fragmentsize * numfragments)) < 0) {
225			snd_pcm_close(pcm_handle);
226			return false;
227			}
228
229			snd_pcm_close(pcm_handle);
230			return true;
231			}
232
233			void AudioOutputDeviceAlsa::Play() {
234			StartThread();
235			}
236
237			bool AudioOutputDeviceAlsa::IsPlaying() {
238			return IsRunning(); // if Thread is running
239			}
240
241			void AudioOutputDeviceAlsa::Stop() {
242			StopThread();
243			}
244
245			void AudioOutputDeviceAlsa::AcquireChannels(uint Channels) {
246			if (Channels > uiAlsaChannels) {
247			// just create mix channel(s)
248			for (int i = uiAlsaChannels; i < Channels; i++) {
249			AudioChannel* pNewChannel = new AudioChannel(this->Channels[i % uiAlsaChannels]);
250			this->Channels.push_back(pNewChannel);
251			}
252			}
253			}
254
255			uint AudioOutputDeviceAlsa::MaxSamplesPerCycle() {
256			return FragmentSize;
257			}
258
259			uint AudioOutputDeviceAlsa::SampleRate() {
260			return uiSamplerate;
261			}
262
263			String AudioOutputDeviceAlsa::Name() {
264			return "Alsa";
265			}
266
267			String AudioOutputDeviceAlsa::Driver() {
268			return Name();
269			}
270
271			String AudioOutputDeviceAlsa::Description() {
272			return "Advanced Linux Sound Architecture";
273			}
274
275			String AudioOutputDeviceAlsa::Version() {
276	schoenebeck	208	String s = "$Revision: 1.10 $";
277	schoenebeck	200	return s.substr(11, s.size() - 13); // cut dollar signs, spaces and CVS macro keyword
278			}
279
280			/**
281			* Entry point for the thread.
282			*/
283			int AudioOutputDeviceAlsa::Main() {
284			while (true) {
285
286			// let all connected engines render 'FragmentSize' sample points
287			RenderAudio(FragmentSize);
288
289			// convert from DSP value range (-1.0..+1.0) to 16 bit integer value
290			// range (-32768..+32767), check clipping and copy to Alsa output buffer
291			// (note: we use interleaved output method to Alsa)
292			for (int c = 0; c < uiAlsaChannels; c++) {
293			float* in = Channels[c]->Buffer();
294			for (int i = 0, o = c; i < FragmentSize; i++ , o += uiAlsaChannels) {
295			float sample_point = in[i] * 32768.0f;
296			if (sample_point < -32768.0) sample_point = -32768.0;
297			if (sample_point > 32767.0) sample_point = 32767.0;
298			pAlsaOutputBuffer[o] = (int16_t) sample_point;
299			}
300			}
301
302			// output sound
303			int res = Output();
304			if (res < 0) {
305			fprintf(stderr, "Alsa: Audio output error, exiting.\n");
306			exit(EXIT_FAILURE);
307			}
308			}
309			}
310
311			/**
312			* Will be called after every audio fragment cycle, to output the audio data
313			* of the current fragment to the soundcard.
314			*
315			* @returns 0 on success, a value < 0 on error
316			*/
317			int AudioOutputDeviceAlsa::Output() {
318			int err = snd_pcm_writei(pcm_handle, pAlsaOutputBuffer, FragmentSize);
319			if (err < 0) {
320			fprintf(stderr, "Error snd_pcm_writei failed: %s\n", snd_strerror(err));
321			return -1;
322			}
323			return 0;
324			}
325
326			} // namespace LinuxSampler
327