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schoenebeck |
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/*************************************************************************** |
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* * |
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* LinuxSampler - modular, streaming capable sampler * |
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* * |
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* Copyright (C) 2003 by Benno Senoner and Christian Schoenebeck * |
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* * |
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* This program is free software; you can redistribute it and/or modify * |
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* it under the terms of the GNU General Public License as published by * |
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* the Free Software Foundation; either version 2 of the License, or * |
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* (at your option) any later version. * |
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* * |
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* This program is distributed in the hope that it will be useful, * |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of * |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * |
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* GNU General Public License for more details. * |
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* * |
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* You should have received a copy of the GNU General Public License * |
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* along with this program; if not, write to the Free Software * |
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, * |
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* MA 02111-1307 USA * |
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***************************************************************************/ |
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#include "alsaio.h" |
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AlsaIO::AlsaIO() : AudioIO(), Thread(true, 1, 0) { |
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pcm_handle = NULL; |
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pOutputBuffer = NULL; |
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stream = SND_PCM_STREAM_PLAYBACK; |
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} |
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int AlsaIO::Initialize(uint Channels, uint Samplerate, uint Fragments, uint FragmentSize) { |
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this->uiChannels = Channels; |
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this->uiSamplerate = Samplerate; |
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this->uiMaxSamplesPerCycle = FragmentSize; |
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this->bInterleaved = true; |
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if (HardwareParametersSupported(Channels, Samplerate, Fragments, FragmentSize)) { |
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pcm_name = "hw:0,0"; |
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} |
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else { |
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printf("Warning: your soundcard doesn't support chosen hardware parameters; "); |
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printf("trying to compensate support lack with plughw..."); |
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fflush(stdout); |
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pcm_name = "plughw:0,0"; |
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} |
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int err; |
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snd_pcm_hw_params_alloca(&hwparams); // Allocate the snd_pcm_hw_params_t structure on the stack. |
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/* Open PCM. The last parameter of this function is the mode. */ |
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/* If this is set to 0, the standard mode is used. Possible */ |
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/* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC. */ |
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/* If SND_PCM_NONBLOCK is used, read / write access to the */ |
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/* PCM device will return immediately. If SND_PCM_ASYNC is */ |
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/* specified, SIGIO will be emitted whenever a period has */ |
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/* been completely processed by the soundcard. */ |
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if ((err = snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0)) < 0) { |
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fprintf(stderr, "Error opening PCM device %s: %s\n", pcm_name.c_str(), snd_strerror(err)); |
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return -1; |
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} |
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if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) { |
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fprintf(stderr, "Error, cannot initialize hardware parameter structure: %s.\n", snd_strerror(err)); |
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return -1; |
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} |
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/* Set access type. This can be either */ |
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/* SND_PCM_ACCESS_RW_INTERLEAVED or */ |
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/* SND_PCM_ACCESS_RW_NONINTERLEAVED. */ |
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if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { |
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fprintf(stderr, "Error snd_pcm_hw_params_set_access: %s.\n", snd_strerror(err)); |
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return -1; |
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} |
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/* Set sample format */ |
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#if WORDS_BIGENDIAN |
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if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE)) < 0) { |
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#else // little endian |
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if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0) { |
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#endif |
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fprintf(stderr, "Error setting sample format. : %s\n", snd_strerror(err)); |
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return -1; |
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} |
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int dir = 0; |
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/* Set sample rate. If the exact rate is not supported */ |
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/* by the hardware, use nearest possible rate. */ |
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#if ALSA_MAJOR > 0 |
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if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &Samplerate, &dir)) < 0) { |
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#else |
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if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, Samplerate, &dir)) < 0) { |
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#endif |
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fprintf(stderr, "Error setting sample rate. : %s\n", snd_strerror(err)); |
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return -1; |
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} |
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if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, Channels)) < 0) { |
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fprintf(stderr, "Error setting number of channels. : %s\n", snd_strerror(err)); |
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return -1; |
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} |
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/* Set number of periods. Periods used to be called fragments. */ |
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if ((err = snd_pcm_hw_params_set_periods(pcm_handle, hwparams, Fragments, dir)) < 0) { |
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fprintf(stderr, "Error setting number of periods. : %s\n", snd_strerror(err)); |
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return -1; |
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} |
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/* Set buffer size (in frames). The resulting latency is given by */ |
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/* latency = periodsize * periods / (rate * bytes_per_frame) */ |
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if ((err = snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (FragmentSize * Fragments))) < 0) { |
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fprintf(stderr, "Error setting buffersize. : %s\n", snd_strerror(err)); |
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return -1; |
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} |
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/* Apply HW parameter settings to */ |
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/* PCM device and prepare device */ |
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if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) { |
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fprintf(stderr, "Error setting HW params. : %s\n", snd_strerror(err)); |
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return -1; |
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} |
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if (snd_pcm_sw_params_malloc(&swparams) != 0) { |
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fprintf(stderr, "Error in snd_pcm_sw_params_malloc. : %s\n", snd_strerror(err)); |
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return -1; |
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} |
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if (snd_pcm_sw_params_current(pcm_handle, swparams) != 0) { |
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fprintf(stderr, "Error in snd_pcm_sw_params_current. : %s\n", snd_strerror(err)); |
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return -1; |
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} |
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if (snd_pcm_sw_params_set_stop_threshold(pcm_handle, swparams, 0xffffffff) != 0) { |
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fprintf(stderr, "Error in snd_pcm_sw_params_set_stop_threshold. : %s\n", snd_strerror(err)); |
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return -1; |
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} |
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if (snd_pcm_sw_params(pcm_handle, swparams) != 0) { |
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fprintf(stderr, "Error in snd_pcm_sw_params. : %s\n", snd_strerror(err)); |
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return -1; |
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} |
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if ((err = snd_pcm_prepare(pcm_handle)) < 0) { |
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fprintf(stderr, "Error snd_pcm_prepare : %s\n", snd_strerror(err)); |
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return -1; |
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} |
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// allocate the audio output buffer |
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pOutputBuffer = new int16_t[Channels * FragmentSize]; |
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this->bInitialized = true; |
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return 0; |
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} |
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/** |
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* Checks if sound card supports the chosen parameters. |
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* |
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* @returns true if hardware supports it |
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*/ |
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bool AlsaIO::HardwareParametersSupported(uint channels, int samplerate, uint numfragments, uint fragmentsize) { |
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pcm_name = "hw:0,0"; |
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if (snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0) < 0) return false; |
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snd_pcm_hw_params_alloca(&hwparams); |
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if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) { |
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snd_pcm_close(pcm_handle); |
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return false; |
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} |
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if (snd_pcm_hw_params_test_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) { |
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snd_pcm_close(pcm_handle); |
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return false; |
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} |
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#if WORDS_BIGENDIAN |
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if (snd_pcm_hw_params_test_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE) < 0) { |
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#else // little endian |
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if (snd_pcm_hw_params_test_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE) < 0) { |
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#endif |
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snd_pcm_close(pcm_handle); |
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return false; |
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} |
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int dir = 0; |
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if (snd_pcm_hw_params_test_rate(pcm_handle, hwparams, samplerate, dir) < 0) { |
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snd_pcm_close(pcm_handle); |
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return false; |
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} |
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if (snd_pcm_hw_params_test_channels(pcm_handle, hwparams, channels) < 0) { |
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snd_pcm_close(pcm_handle); |
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return false; |
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} |
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if (snd_pcm_hw_params_test_periods(pcm_handle, hwparams, numfragments, dir) < 0) { |
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snd_pcm_close(pcm_handle); |
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return false; |
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} |
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if (snd_pcm_hw_params_test_buffer_size(pcm_handle, hwparams, (fragmentsize * numfragments)) < 0) { |
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snd_pcm_close(pcm_handle); |
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return false; |
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} |
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snd_pcm_close(pcm_handle); |
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return true; |
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} |
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void AlsaIO::Activate() { |
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this->StartThread(); |
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} |
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int AlsaIO::Main() { |
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if (!pEngine) { |
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fprintf(stderr, "AlsaIO: No Sampler Engine assigned, exiting.\n"); |
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exit(EXIT_FAILURE); |
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} |
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if (!bInitialized) { |
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fprintf(stderr, "AlsaIO: Not yet intitialized, exiting.\n"); |
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exit(EXIT_FAILURE); |
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} |
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while (true) { |
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// let the engine render audio for the current audio fragment |
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pEngine->RenderAudio(uiMaxSamplesPerCycle); |
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// check clipping in the audio sum, convert to sample_type |
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// (from 32bit to 16bit sample) and copy to output buffer |
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float sample_point; uint o = 0; |
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for (uint s = 0; s < uiMaxSamplesPerCycle; s++) { |
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for (uint c = 0; c < uiChannels; c++) { |
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sample_point = pEngine->GetAudioSumBuffer(c)[s] * pEngine->Volume; |
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if (sample_point < -32768.0) sample_point = -32768.0; |
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if (sample_point > 32767.0) sample_point = 32767.0; |
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this->pOutputBuffer[o++] = (int32_t) sample_point; |
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} |
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} |
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// output sound |
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int res = Output(); |
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if (res < 0) { |
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fprintf(stderr, "AlsaIO: Audio output error, exiting.\n"); |
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exit(EXIT_FAILURE); |
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} |
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} |
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} |
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/** |
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* Will be called after every audio fragment cycle, to output the audio data |
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* of the current fragment to the soundcard. |
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* |
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* @returns 0 on success |
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*/ |
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int AlsaIO::Output() { |
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int err = snd_pcm_writei(pcm_handle, pOutputBuffer, uiMaxSamplesPerCycle); |
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if (err < 0) { |
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fprintf(stderr, "Error snd_pcm_writei failed. : %s\n", snd_strerror(err)); |
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return -1; |
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} |
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return 0; |
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} |
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void AlsaIO::Close() { |
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if (bInitialized) { |
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//dmsg(0,("Stopping Alsa Thread...")); |
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//StopThread(); //FIXME: commented out due to a bug in thread.cpp (StopThread() doesn't return at all) |
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//dmsg(0,("OK\n")); |
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if (pcm_handle) { |
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//FIXME: currently commented out due to segfault |
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//snd_pcm_close(pcm_handle); |
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pcm_handle = NULL; |
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} |
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if (pOutputBuffer) { |
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//FIXME: currently commented out due to segfault |
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//delete[] pOutputBuffer; |
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pOutputBuffer = NULL; |
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} |
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bInitialized = false; |
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} |
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} |
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void* AlsaIO::GetInterleavedOutputBuffer() { |
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return pOutputBuffer; |
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} |
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void* AlsaIO::GetChannelOutputBufer(uint Channel) { |
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fprintf(stderr, "AlsaIO::GetChannelOutputBufer(): Only interleaved access allowed so far, exiting.\n"); |
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exit(EXIT_FAILURE); |
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// just to avoid compiler warnings |
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return NULL; |
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} |