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 "AudioOutputDeviceAlsa.h"
#include "AudioOutputDeviceFactory.h"

namespace LinuxSampler {

    REGISTER_AUDIO_OUTPUT_DRIVER("Alsa",AudioOutputDeviceAlsa);

    /**
     * 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,String> Parameters) : AudioOutputDevice(CreateParameters(Parameters)), Thread(true, 1, 0) {
        pcm_handle           = NULL;
        stream               = SND_PCM_STREAM_PLAYBACK;
        this->uiAlsaChannels = ((DeviceCreationParameterInt*)this->Parameters["channels"])->ValueAsInt();
        this->uiSamplerate   = ((DeviceCreationParameterInt*)this->Parameters["samplerate"])->ValueAsInt();
        this->FragmentSize   = ((DeviceCreationParameterInt*)this->Parameters["fragmentsize"])->ValueAsInt();
        uint Fragments       = ((DeviceCreationParameterInt*)this->Parameters["fragments"])->ValueAsInt();
        String Card          = this->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));
    }

    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;
        }
    }

    std::map<String,DeviceCreationParameter*> AudioOutputDeviceAlsa::CreateParameters(std::map<String,String> Parameters) {
        std::map<String,DeviceCreationParameter*> result;
        result["card"]         = new ParameterCard(this, Parameters["card"]);                 // additional parameter, individually for this driver
        result["fragments"]    = new ParameterFragments(this, Parameters["fragments"]);       // additional parameter, individually for this driver
        result["fragmentsize"] = new ParameterFragmentSize(this, Parameters["fragmentsize"]); // additional parameter, individually for this driver
        return result;
    }

    /**
     *  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::Description() {
        return "Advanced Linux Sound Architecture";
    }

    String AudioOutputDeviceAlsa::Version() {
       String s = "$Revision: 1.5 $";
       return s.substr(11, s.size() - 13); // cut dollar signs, spaces and CVS macro keyword
    }

    std::map<String,DeviceCreationParameter*> AudioOutputDeviceAlsa::AvailableParameters() {
        // FIXME: not a good solution to get the commot parameters (ACTIVE,SAMPERATE,CHANNELS which have to be offered by all audio output drivers)
        std::map<String,DeviceCreationParameter*> available_parameters = AudioOutputDevice::AvailableParameters();
        static ParameterCard         param_card(NULL);
        static ParameterFragments    param_fragments(NULL);
        static ParameterFragmentSize param_fragmentsize(NULL);
        available_parameters["card"]         = &param_card;         // additional parameter, individually for this driver
        available_parameters["fragments"]    = &param_fragments;    // additional parameter, individually for this driver
        available_parameters["fragmentsize"] = &param_fragmentsize; // additional parameter, individually for this driver
        return available_parameters;
    }

    /**
     * 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	53	/***************************************************************************
2			* *
3			* LinuxSampler - modular, streaming capable sampler *
4			* *
5	schoenebeck	56	* Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck *
6	schoenebeck	53	* *
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	schoenebeck	123	#include "AudioOutputDeviceFactory.h"
25	schoenebeck	53
26			namespace LinuxSampler {
27
28	senkov	136	REGISTER_AUDIO_OUTPUT_DRIVER("Alsa",AudioOutputDeviceAlsa);
29	schoenebeck	123
30	schoenebeck	53	/**
31	schoenebeck	123	* Create and initialize Alsa audio output device with given parameters.
32	schoenebeck	53	*
33	schoenebeck	123	* @param Parameters - optional parameters
34	schoenebeck	53	* @throws AudioOutputException if output device cannot be opened
35			*/
36	schoenebeck	123	AudioOutputDeviceAlsa::AudioOutputDeviceAlsa(std::map<String,String> Parameters) : AudioOutputDevice(CreateParameters(Parameters)), Thread(true, 1, 0) {
37	schoenebeck	53	pcm_handle = NULL;
38			stream = SND_PCM_STREAM_PLAYBACK;
39	senkov	136	this->uiAlsaChannels = ((DeviceCreationParameterInt*)this->Parameters["channels"])->ValueAsInt();
40			this->uiSamplerate = ((DeviceCreationParameterInt*)this->Parameters["samplerate"])->ValueAsInt();
41			this->FragmentSize = ((DeviceCreationParameterInt*)this->Parameters["fragmentsize"])->ValueAsInt();
42			uint Fragments = ((DeviceCreationParameterInt*)this->Parameters["fragments"])->ValueAsInt();
43			String Card = this->Parameters["card"]->Value();
44	schoenebeck	53
45	schoenebeck	123	dmsg(1,("Checking if hw parameters supported...\n"));
46			if (HardwareParametersSupported(Card, uiAlsaChannels, uiSamplerate, Fragments, FragmentSize)) {
47	schoenebeck	53	pcm_name = "hw:" + Card;
48			}
49			else {
50			printf("Warning: your soundcard doesn't support chosen hardware parameters; ");
51			printf("trying to compensate support lack with plughw...");
52			fflush(stdout);
53			pcm_name = "plughw:" + Card;
54			}
55	schoenebeck	123	dmsg(1,("HW check completed.\n"));
56	schoenebeck	53
57			int err;
58
59			snd_pcm_hw_params_alloca(&hwparams); // Allocate the snd_pcm_hw_params_t structure on the stack.
60
61			/* Open PCM. The last parameter of this function is the mode. */
62			/* If this is set to 0, the standard mode is used. Possible */
63			/* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC. */
64			/* If SND_PCM_NONBLOCK is used, read / write access to the */
65			/* PCM device will return immediately. If SND_PCM_ASYNC is */
66			/* specified, SIGIO will be emitted whenever a period has */
67			/* been completely processed by the soundcard. */
68			if ((err = snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0)) < 0) {
69			throw AudioOutputException(String("Error opening PCM device ") + pcm_name + ": " + snd_strerror(err));
70			}
71
72			if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) {
73			throw AudioOutputException(String("Error, cannot initialize hardware parameter structure: ") + snd_strerror(err));
74			}
75
76			/* Set access type. This can be either */
77			/* SND_PCM_ACCESS_RW_INTERLEAVED or */
78			/* SND_PCM_ACCESS_RW_NONINTERLEAVED. */
79			if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
80			throw AudioOutputException(String("Error snd_pcm_hw_params_set_access: ") + snd_strerror(err));
81			}
82
83			/* Set sample format */
84			#if WORDS_BIGENDIAN
85			if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE)) < 0)
86			#else // little endian
87			if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
88			#endif
89			{
90			throw AudioOutputException(String("Error setting sample format: ") + snd_strerror(err));
91			}
92
93			int dir = 0;
94
95			/* Set sample rate. If the exact rate is not supported */
96			/* by the hardware, use nearest possible rate. */
97			#if ALSA_MAJOR > 0
98	schoenebeck	123	if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &uiSamplerate, &dir)) < 0)
99	schoenebeck	53	#else
100	schoenebeck	123	if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, uiSamplerate, &dir)) < 0)
101	schoenebeck	53	#endif
102			{
103			throw AudioOutputException(String("Error setting sample rate: ") + snd_strerror(err));
104			}
105
106	schoenebeck	123	if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, uiAlsaChannels)) < 0) {
107	schoenebeck	53	throw AudioOutputException(String("Error setting number of channels: ") + snd_strerror(err));
108			}
109
110			/* Set number of periods. Periods used to be called fragments. */
111			if ((err = snd_pcm_hw_params_set_periods(pcm_handle, hwparams, Fragments, dir)) < 0) {
112			throw AudioOutputException(String("Error setting number of periods: ") + snd_strerror(err));
113			}
114
115			/* Set buffer size (in frames). The resulting latency is given by */
116			/* latency = periodsize * periods / (rate * bytes_per_frame) */
117			if ((err = snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (FragmentSize * Fragments))) < 0) {
118			throw AudioOutputException(String("Error setting buffersize: ") + snd_strerror(err));
119			}
120
121			/* Apply HW parameter settings to */
122			/* PCM device and prepare device */
123			if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) {
124			throw AudioOutputException(String("Error setting HW params: ") + snd_strerror(err));
125			}
126
127			if (snd_pcm_sw_params_malloc(&swparams) != 0) {
128			throw AudioOutputException(String("Error in snd_pcm_sw_params_malloc: ") + snd_strerror(err));
129			}
130
131			if (snd_pcm_sw_params_current(pcm_handle, swparams) != 0) {
132			throw AudioOutputException(String("Error in snd_pcm_sw_params_current: ") + snd_strerror(err));
133			}
134
135			if (snd_pcm_sw_params_set_stop_threshold(pcm_handle, swparams, 0xffffffff) != 0) {
136			throw AudioOutputException(String("Error in snd_pcm_sw_params_set_stop_threshold: ") + snd_strerror(err));
137			}
138
139			if (snd_pcm_sw_params(pcm_handle, swparams) != 0) {
140			throw AudioOutputException(String("Error in snd_pcm_sw_params: ") + snd_strerror(err));
141			}
142
143			if ((err = snd_pcm_prepare(pcm_handle)) < 0) {
144			throw AudioOutputException(String("Error snd_pcm_prepare: ") + snd_strerror(err));
145			}
146
147			// allocate Alsa output buffer
148	schoenebeck	123	pAlsaOutputBuffer = new int16_t[uiAlsaChannels * FragmentSize];
149	schoenebeck	53
150			// create audio channels for this audio device to which the sampler engines can write to
151	schoenebeck	123	for (int i = 0; i < uiAlsaChannels; i++) this->Channels.push_back(new AudioChannel(FragmentSize));
152	schoenebeck	53	}
153
154			AudioOutputDeviceAlsa::~AudioOutputDeviceAlsa() {
155			//dmsg(0,("Stopping Alsa Thread..."));
156			//StopThread(); //FIXME: commented out due to a bug in thread.cpp (StopThread() doesn't return at all)
157			//dmsg(0,("OK\n"));
158
159			//FIXME: currently commented out due to segfault
160			//snd_pcm_close(pcm_handle);
161
162			// destroy all audio channels
163			for (int c = 0; c < Channels.size(); c++) delete Channels[c];
164
165			if (pAlsaOutputBuffer) {
166			//FIXME: currently commented out due to segfault
167			//delete[] pOutputBuffer;
168			}
169			}
170
171	schoenebeck	123	std::map<String,DeviceCreationParameter*> AudioOutputDeviceAlsa::CreateParameters(std::map<String,String> Parameters) {
172			std::map<String,DeviceCreationParameter*> result;
173	senkov	136	result["card"] = new ParameterCard(this, Parameters["card"]); // additional parameter, individually for this driver
174			result["fragments"] = new ParameterFragments(this, Parameters["fragments"]); // additional parameter, individually for this driver
175			result["fragmentsize"] = new ParameterFragmentSize(this, Parameters["fragmentsize"]); // additional parameter, individually for this driver
176	schoenebeck	123	return result;
177			}
178
179	schoenebeck	53	/**
180			* Checks if sound card supports the chosen parameters.
181			*
182			* @returns true if hardware supports it
183			*/
184	schoenebeck	123	bool AudioOutputDeviceAlsa::HardwareParametersSupported(String card, uint channels, int samplerate, uint numfragments, uint fragmentsize) {
185			pcm_name = "hw:" + card;
186	schoenebeck	53	if (snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0) < 0) return false;
187			snd_pcm_hw_params_alloca(&hwparams);
188			if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) {
189			snd_pcm_close(pcm_handle);
190			return false;
191			}
192			if (snd_pcm_hw_params_test_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
193			snd_pcm_close(pcm_handle);
194			return false;
195			}
196			#if WORDS_BIGENDIAN
197			if (snd_pcm_hw_params_test_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE) < 0)
198			#else // little endian
199			if (snd_pcm_hw_params_test_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE) < 0)
200			#endif
201			{
202			snd_pcm_close(pcm_handle);
203			return false;
204			}
205			int dir = 0;
206			if (snd_pcm_hw_params_test_rate(pcm_handle, hwparams, samplerate, dir) < 0) {
207			snd_pcm_close(pcm_handle);
208			return false;
209			}
210			if (snd_pcm_hw_params_test_channels(pcm_handle, hwparams, channels) < 0) {
211			snd_pcm_close(pcm_handle);
212			return false;
213			}
214			if (snd_pcm_hw_params_test_periods(pcm_handle, hwparams, numfragments, dir) < 0) {
215			snd_pcm_close(pcm_handle);
216			return false;
217			}
218			if (snd_pcm_hw_params_test_buffer_size(pcm_handle, hwparams, (fragmentsize * numfragments)) < 0) {
219			snd_pcm_close(pcm_handle);
220			return false;
221			}
222
223			snd_pcm_close(pcm_handle);
224			return true;
225			}
226
227			void AudioOutputDeviceAlsa::Play() {
228			StartThread();
229			}
230
231			bool AudioOutputDeviceAlsa::IsPlaying() {
232			return IsRunning(); // if Thread is running
233			}
234
235			void AudioOutputDeviceAlsa::Stop() {
236			StopThread();
237			}
238
239			void AudioOutputDeviceAlsa::AcquireChannels(uint Channels) {
240			if (Channels > uiAlsaChannels) {
241			// just create mix channel(s)
242			for (int i = uiAlsaChannels; i < Channels; i++) {
243			AudioChannel* pNewChannel = new AudioChannel(this->Channels[i % uiAlsaChannels]);
244			this->Channels.push_back(pNewChannel);
245			}
246			}
247			}
248
249			uint AudioOutputDeviceAlsa::MaxSamplesPerCycle() {
250			return FragmentSize;
251			}
252
253			uint AudioOutputDeviceAlsa::SampleRate() {
254			return uiSamplerate;
255			}
256
257	schoenebeck	123	String AudioOutputDeviceAlsa::Description() {
258			return "Advanced Linux Sound Architecture";
259			}
260
261			String AudioOutputDeviceAlsa::Version() {
262	senkov	136	String s = "$Revision: 1.5 $";
263	schoenebeck	123	return s.substr(11, s.size() - 13); // cut dollar signs, spaces and CVS macro keyword
264			}
265
266			std::map<String,DeviceCreationParameter*> AudioOutputDeviceAlsa::AvailableParameters() {
267			// FIXME: not a good solution to get the commot parameters (ACTIVE,SAMPERATE,CHANNELS which have to be offered by all audio output drivers)
268			std::map<String,DeviceCreationParameter*> available_parameters = AudioOutputDevice::AvailableParameters();
269			static ParameterCard param_card(NULL);
270			static ParameterFragments param_fragments(NULL);
271			static ParameterFragmentSize param_fragmentsize(NULL);
272	senkov	136	available_parameters["card"] = &param_card; // additional parameter, individually for this driver
273			available_parameters["fragments"] = &param_fragments; // additional parameter, individually for this driver
274			available_parameters["fragmentsize"] = &param_fragmentsize; // additional parameter, individually for this driver
275	schoenebeck	123	return available_parameters;
276			}
277
278			/**
279			* Entry point for the thread.
280			*/
281	schoenebeck	53	int AudioOutputDeviceAlsa::Main() {
282			while (true) {
283
284			// let all connected engines render 'FragmentSize' sample points
285			RenderAudio(FragmentSize);
286
287			// convert from DSP value range (-1.0..+1.0) to 16 bit integer value
288			// range (-32768..+32767), check clipping and copy to Alsa output buffer
289			// (note: we use interleaved output method to Alsa)
290			for (int c = 0; c < uiAlsaChannels; c++) {
291			float* in = Channels[c]->Buffer();
292			for (int i = 0, o = c; i < FragmentSize; i++ , o += uiAlsaChannels) {
293			float sample_point = in[i] * 32768.0f;
294			if (sample_point < -32768.0) sample_point = -32768.0;
295			if (sample_point > 32767.0) sample_point = 32767.0;
296			pAlsaOutputBuffer[o] = (int16_t) sample_point;
297			}
298			}
299
300			// output sound
301			int res = Output();
302			if (res < 0) {
303			fprintf(stderr, "Alsa: Audio output error, exiting.\n");
304			exit(EXIT_FAILURE);
305			}
306			}
307			}
308
309			/**
310			* Will be called after every audio fragment cycle, to output the audio data
311			* of the current fragment to the soundcard.
312			*
313			* @returns 0 on success, a value < 0 on error
314			*/
315			int AudioOutputDeviceAlsa::Output() {
316			int err = snd_pcm_writei(pcm_handle, pAlsaOutputBuffer, FragmentSize);
317			if (err < 0) {
318			fprintf(stderr, "Error snd_pcm_writei failed: %s\n", snd_strerror(err));
319			return -1;
320			}
321			return 0;
322			}
323
324			} // namespace LinuxSampler
325