src/audiodriver/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 "../drivers/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: 1.8 $";
       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	/***************************************************************************
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 "../drivers/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: 1.8 $";
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