/*************************************************************************** * * * LinuxSampler - modular, streaming capable sampler * * * * Copyright (C) 2003 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 "alsaio.h" AlsaIO::AlsaIO() : AudioIO(), Thread(true, 1, 0) { pcm_handle = NULL; pOutputBuffer = NULL; stream = SND_PCM_STREAM_PLAYBACK; } int AlsaIO::Initialize(uint Channels, uint Samplerate, uint Fragments, uint FragmentSize, String Card) { this->uiChannels = Channels; this->uiSamplerate = Samplerate; this->uiMaxSamplesPerCycle = FragmentSize; this->bInterleaved = true; if (HardwareParametersSupported(Channels, Samplerate, 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; } 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) { fprintf(stderr, "Error opening PCM device %s: %s\n", pcm_name.c_str(), snd_strerror(err)); return -1; } if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) { fprintf(stderr, "Error, cannot initialize hardware parameter structure: %s.\n", snd_strerror(err)); return -1; } /* 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) { fprintf(stderr, "Error snd_pcm_hw_params_set_access: %s.\n", snd_strerror(err)); return -1; } /* 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 fprintf(stderr, "Error setting sample format. : %s\n", snd_strerror(err)); return -1; } 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, &Samplerate, &dir)) < 0) { #else if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, Samplerate, &dir)) < 0) { #endif fprintf(stderr, "Error setting sample rate. : %s\n", snd_strerror(err)); return -1; } if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, Channels)) < 0) { fprintf(stderr, "Error setting number of channels. : %s\n", snd_strerror(err)); return -1; } /* Set number of periods. Periods used to be called fragments. */ if ((err = snd_pcm_hw_params_set_periods(pcm_handle, hwparams, Fragments, dir)) < 0) { fprintf(stderr, "Error setting number of periods. : %s\n", snd_strerror(err)); return -1; } /* 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) { fprintf(stderr, "Error setting buffersize. : %s\n", snd_strerror(err)); return -1; } /* Apply HW parameter settings to */ /* PCM device and prepare device */ if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) { fprintf(stderr, "Error setting HW params. : %s\n", snd_strerror(err)); return -1; } if (snd_pcm_sw_params_malloc(&swparams) != 0) { fprintf(stderr, "Error in snd_pcm_sw_params_malloc. : %s\n", snd_strerror(err)); return -1; } if (snd_pcm_sw_params_current(pcm_handle, swparams) != 0) { fprintf(stderr, "Error in snd_pcm_sw_params_current. : %s\n", snd_strerror(err)); return -1; } if (snd_pcm_sw_params_set_stop_threshold(pcm_handle, swparams, 0xffffffff) != 0) { fprintf(stderr, "Error in snd_pcm_sw_params_set_stop_threshold. : %s\n", snd_strerror(err)); return -1; } if (snd_pcm_sw_params(pcm_handle, swparams) != 0) { fprintf(stderr, "Error in snd_pcm_sw_params. : %s\n", snd_strerror(err)); return -1; } if ((err = snd_pcm_prepare(pcm_handle)) < 0) { fprintf(stderr, "Error snd_pcm_prepare : %s\n", snd_strerror(err)); return -1; } // allocate the audio output buffer pOutputBuffer = new int16_t[Channels * FragmentSize]; this->bInitialized = true; return 0; } /** * Checks if sound card supports the chosen parameters. * * @returns true if hardware supports it */ bool AlsaIO::HardwareParametersSupported(uint channels, int samplerate, uint numfragments, uint fragmentsize) { pcm_name = "hw:0,0"; 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 AlsaIO::Activate() { this->StartThread(); } int AlsaIO::Main() { if (!pEngine) { fprintf(stderr, "AlsaIO: No Sampler Engine assigned, exiting.\n"); exit(EXIT_FAILURE); } if (!bInitialized) { fprintf(stderr, "AlsaIO: Not yet intitialized, exiting.\n"); exit(EXIT_FAILURE); } while (true) { // let the engine render audio for the current audio fragment pEngine->RenderAudio(uiMaxSamplesPerCycle); // check clipping in the audio sum, convert to sample_type // (from 32bit to 16bit sample) and copy to output buffer float sample_point; uint o = 0; for (uint s = 0; s < uiMaxSamplesPerCycle; s++) { for (uint c = 0; c < uiChannels; c++) { sample_point = pEngine->GetAudioSumBuffer(c)[s] * pEngine->Volume; if (sample_point < -32768.0) sample_point = -32768.0; if (sample_point > 32767.0) sample_point = 32767.0; this->pOutputBuffer[o++] = (int32_t) sample_point; } } // output sound int res = Output(); if (res < 0) { fprintf(stderr, "AlsaIO: 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 */ int AlsaIO::Output() { int err = snd_pcm_writei(pcm_handle, pOutputBuffer, uiMaxSamplesPerCycle); if (err < 0) { fprintf(stderr, "Error snd_pcm_writei failed. : %s\n", snd_strerror(err)); return -1; } return 0; } void AlsaIO::Close() { if (bInitialized) { //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")); if (pcm_handle) { //FIXME: currently commented out due to segfault //snd_pcm_close(pcm_handle); pcm_handle = NULL; } if (pOutputBuffer) { //FIXME: currently commented out due to segfault //delete[] pOutputBuffer; pOutputBuffer = NULL; } bInitialized = false; } } void* AlsaIO::GetInterleavedOutputBuffer() { return pOutputBuffer; } void* AlsaIO::GetChannelOutputBufer(uint Channel) { fprintf(stderr, "AlsaIO::GetChannelOutputBufer(): Only interleaved access allowed so far, exiting.\n"); exit(EXIT_FAILURE); // just to avoid compiler warnings return NULL; }