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	/***************************************************************************
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	String s = "$Revision: 1.10 $";
277	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