/*************************************************************************** * __________ __ ___. * Open \______ \ ____ ____ | | _\_ |__ _______ ___ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ * \/ \/ \/ \/ \/ * * Copyright (C) 2010 Thomas Martitz * Copyright (c) 2020 Solomon Peachy * * 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 software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY * KIND, either express or implied. * ****************************************************************************/ /* * Based, but heavily modified, on the example given at * http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html * * This driver uses the so-called unsafe async callback method and hardcoded device * names. It fails when the audio device is busy by other apps. * * To make the async callback safer, an alternative stack is installed, since * it's run from a signal hanlder (which otherwise uses the user stack). If * tick tasks are run from a signal handler too, please install * an alternative stack for it too. * * TODO: Rewrite this to do it properly with multithreading * * Alternatively, a version using polling in a tick task is provided. While * supposedly safer, it appears to use more CPU (however I didn't measure it * accurately, only looked at htop). At least, in this mode the "default" * device works which doesnt break with other apps running. * device works which doesnt break with other apps running. */ #include "autoconf.h" #include #include #include //#define LOGF_ENABLE #include "system.h" #include "debug.h" #include "kernel.h" #include "panic.h" #include "pcm.h" #include "pcm-internal.h" #include "pcm_mixer.h" #include "pcm_sampr.h" #include "audiohw.h" #include "pcm-alsa.h" #include "logf.h" #include #include #define USE_ASYNC_CALLBACK /* plughw:0,0 works with both, however "default" is recommended. * default doesnt seem to work with async callback but doesn't break * with multple applications running */ static char device[] = "plughw:0,0"; /* playback device */ static const snd_pcm_access_t access_ = SND_PCM_ACCESS_RW_INTERLEAVED; /* access mode */ #if defined(SONY_NWZ_LINUX) || defined(HAVE_FIIO_LINUX_CODEC) /* Sony NWZ must use 32-bit per sample */ static const snd_pcm_format_t format = SND_PCM_FORMAT_S32_LE; /* sample format */ typedef long sample_t; #else static const snd_pcm_format_t format = SND_PCM_FORMAT_S16; /* sample format */ typedef short sample_t; #endif static const int channels = 2; /* count of channels */ static unsigned int real_sample_rate = 0; static unsigned int last_sample_rate = 0; static snd_pcm_t *handle = NULL; static snd_pcm_sframes_t buffer_size; static snd_pcm_sframes_t period_size; static sample_t *frames = NULL; static const void *pcm_data = 0; static size_t pcm_size = 0; #ifdef USE_ASYNC_CALLBACK static snd_async_handler_t *ahandler; static pthread_mutex_t pcm_mtx; static char signal_stack[SIGSTKSZ]; #else static int recursion; #endif static int set_hwparams(snd_pcm_t *handle) { int err; unsigned int srate; snd_pcm_hw_params_t *params; snd_pcm_hw_params_malloc(¶ms); /* Size playback buffers based on sample rate */ if (pcm_sampr > SAMPR_96) { buffer_size = MIX_FRAME_SAMPLES * 32 * 4; /* ~64k */ period_size = MIX_FRAME_SAMPLES * 4 * 4; /* ~16k */ } else if (pcm_sampr > SAMPR_48) { buffer_size = MIX_FRAME_SAMPLES * 32 * 2; /* ~32k */ period_size = MIX_FRAME_SAMPLES * 4 * 2; /* ~8k */ } else { buffer_size = MIX_FRAME_SAMPLES * 32; /* ~16k */ period_size = MIX_FRAME_SAMPLES * 4; /* ~4k */ } /* choose all parameters */ err = snd_pcm_hw_params_any(handle, params); if (err < 0) { panicf("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err)); goto error; } /* set the interleaved read/write format */ err = snd_pcm_hw_params_set_access(handle, params, access_); if (err < 0) { panicf("Access type not available for playback: %s\n", snd_strerror(err)); goto error; } /* set the sample format */ err = snd_pcm_hw_params_set_format(handle, params, format); if (err < 0) { logf("Sample format not available for playback: %s\n", snd_strerror(err)); goto error; } /* set the count of channels */ err = snd_pcm_hw_params_set_channels(handle, params, channels); if (err < 0) { logf("Channels count (%i) not available for playbacks: %s\n", channels, snd_strerror(err)); goto error; } /* set the stream rate */ srate = pcm_sampr; err = snd_pcm_hw_params_set_rate_near(handle, params, &srate, 0); if (err < 0) { logf("Rate %iHz not available for playback: %s\n", pcm_sampr, snd_strerror(err)); goto error; } real_sample_rate = srate; if (real_sample_rate != pcm_sampr) { logf("Rate doesn't match (requested %iHz, get %iHz)\n", pcm_sampr, real_sample_rate); err = -EINVAL; goto error; } /* set the buffer size */ err = snd_pcm_hw_params_set_buffer_size_near(handle, params, &buffer_size); if (err < 0) { logf("Unable to set buffer size %ld for playback: %s\n", buffer_size, snd_strerror(err)); goto error; } /* set the period size */ err = snd_pcm_hw_params_set_period_size_near (handle, params, &period_size, NULL); if (err < 0) { logf("Unable to set period size %ld for playback: %s\n", period_size, snd_strerror(err)); goto error; } if (frames) free(frames); frames = calloc(1, period_size * channels * sizeof(sample_t)); /* write the parameters to device */ err = snd_pcm_hw_params(handle, params); if (err < 0) { logf("Unable to set hw params for playback: %s\n", snd_strerror(err)); goto error; } err = 0; /* success */ error: snd_pcm_hw_params_free(params); return err; } /* Set sw params: playback start threshold and low buffer watermark */ static int set_swparams(snd_pcm_t *handle) { int err; snd_pcm_sw_params_t *swparams; snd_pcm_sw_params_malloc(&swparams); /* get the current swparams */ err = snd_pcm_sw_params_current(handle, swparams); if (err < 0) { logf("Unable to determine current swparams for playback: %s\n", snd_strerror(err)); goto error; } /* start the transfer when the buffer is half full */ err = snd_pcm_sw_params_set_start_threshold(handle, swparams, buffer_size / 2); if (err < 0) { logf("Unable to set start threshold mode for playback: %s\n", snd_strerror(err)); goto error; } /* allow the transfer when at least period_size samples can be processed */ err = snd_pcm_sw_params_set_avail_min(handle, swparams, period_size); if (err < 0) { logf("Unable to set avail min for playback: %s\n", snd_strerror(err)); goto error; } /* write the parameters to the playback device */ err = snd_pcm_sw_params(handle, swparams); if (err < 0) { logf("Unable to set sw params for playback: %s\n", snd_strerror(err)); goto error; } err = 0; /* success */ error: snd_pcm_sw_params_free(swparams); return err; } /* Digital volume explanation: * with very good approximation (<0.1dB) the convertion from dB to multiplicative * factor, for dB>=0, is 2^(dB/3). We can then notice that if we write dB=3*k+r * then this is 2^k*2^(r/3) so we only need to look at r=0,1,2. For r=0 this is * 1, for r=1 we have 2^(1/3)~=1.25 so we approximate by 1+1/4, and 2^(2/3)~=1.5 * so we approximate by 1+1/2. To go from negative to nonnegative we notice that * 48 dB => 63095 factor ~= 2^16 so we virtually pre-multiply everything by 2^(-16) * and add 48dB to the input volume. We cannot go lower -43dB because several * values between -48dB and -43dB would require a fractional multiplier, which is * stupid to implement for such very low volume. */ static int dig_vol_mult_l = 2 ^ 16; /* multiplicative factor to apply to each sample */ static int dig_vol_mult_r = 2 ^ 16; /* multiplicative factor to apply to each sample */ void pcm_alsa_set_digital_volume(int vol_db_l, int vol_db_r) { if(vol_db_l > 0 || vol_db_r > 0 || vol_db_l < -43 || vol_db_r < -43) panicf("invalid pcm alsa volume"); if(format != SND_PCM_FORMAT_S32_LE) panicf("this function assumes 32-bit sample size"); vol_db_l += 48; /* -42dB .. 0dB => 5dB .. 48dB */ vol_db_r += 48; /* -42dB .. 0dB => 5dB .. 48dB */ /* NOTE if vol_dB = 5 then vol_shift = 1 but r = 1 so we do vol_shift - 1 >= 0 * otherwise vol_dB >= 0 implies vol_shift >= 2 so vol_shift - 2 >= 0 */ int vol_shift_l = vol_db_l / 3; int vol_shift_r = vol_db_r / 3; int r_l = vol_db_l % 3; int r_r = vol_db_r % 3; if(r_l == 0) dig_vol_mult_l = 1 << vol_shift_l; else if(r_l == 1) dig_vol_mult_l = 1 << vol_shift_l | 1 << (vol_shift_l - 2); else dig_vol_mult_l = 1 << vol_shift_l | 1 << (vol_shift_l - 1); logf("l: %d dB -> factor = %d\n", vol_db_l - 48, dig_vol_mult_l); if(r_r == 0) dig_vol_mult_r = 1 << vol_shift_r; else if(r_r == 1) dig_vol_mult_r = 1 << vol_shift_r | 1 << (vol_shift_r - 2); else dig_vol_mult_r = 1 << vol_shift_r | 1 << (vol_shift_r - 1); logf("r: %d dB -> factor = %d\n", vol_db_r - 48, dig_vol_mult_r); } /* copy pcm samples to a spare buffer, suitable for snd_pcm_writei() */ static bool fill_frames(void) { ssize_t copy_n, frames_left = period_size; bool new_buffer = false; while (frames_left > 0) { if (!pcm_size) { new_buffer = true; if (!pcm_play_dma_complete_callback(PCM_DMAST_OK, &pcm_data, &pcm_size)) { return false; } } if (pcm_size % 4) panicf("Wrong pcm_size"); /* the compiler will optimize this test away */ copy_n = MIN((ssize_t)pcm_size/4, frames_left); if (format == SND_PCM_FORMAT_S32_LE) { /* We have to convert 16-bit to 32-bit, the need to multiply the * sample by some value so the sound is not too low */ const short *pcm_ptr = pcm_data; sample_t *sample_ptr = &frames[2*(period_size-frames_left)]; for (int i = 0; i < copy_n; i++) { *sample_ptr++ = *pcm_ptr++ * dig_vol_mult_l; *sample_ptr++ = *pcm_ptr++ * dig_vol_mult_r; } } else { /* Rockbox and PCM have same format: memcopy */ memcpy(&frames[2*(period_size-frames_left)], pcm_data, copy_n * 4); } pcm_data += copy_n*4; pcm_size -= copy_n*4; frames_left -= copy_n; if (new_buffer) { new_buffer = false; pcm_play_dma_status_callback(PCM_DMAST_STARTED); } } return true; } #ifdef USE_ASYNC_CALLBACK static void async_callback(snd_async_handler_t *ahandler) { snd_pcm_t *handle = snd_async_handler_get_pcm(ahandler); if (pthread_mutex_trylock(&pcm_mtx) != 0) return; #else static void pcm_tick(void) { if (snd_pcm_state(handle) != SND_PCM_STATE_RUNNING) return; #endif while (snd_pcm_avail_update(handle) >= period_size) { if (fill_frames()) { int err = snd_pcm_writei(handle, frames, period_size); if (err < 0 && err != period_size && err != -EAGAIN) { logf("Write error: written %i expected %li\n", err, period_size); break; } } else { logf("%s: No Data.\n", __func__); break; } } #ifdef USE_ASYNC_CALLBACK pthread_mutex_unlock(&pcm_mtx); #endif } static int async_rw(snd_pcm_t *handle) { int err; snd_pcm_sframes_t sample_size; sample_t *samples; #ifdef USE_ASYNC_CALLBACK /* assign alternative stack for the signal handlers */ stack_t ss = { .ss_sp = signal_stack, .ss_size = sizeof(signal_stack), .ss_flags = 0 }; struct sigaction sa; err = sigaltstack(&ss, NULL); if (err < 0) { logf("Unable to install alternative signal stack: %s", strerror(err)); return err; } err = snd_async_add_pcm_handler(&ahandler, handle, async_callback, NULL); if (err < 0) { logf("Unable to register async handler: %s\n", snd_strerror(err)); return err; } /* only modify the stack the handler runs on */ sigaction(SIGIO, NULL, &sa); sa.sa_flags |= SA_ONSTACK; err = sigaction(SIGIO, &sa, NULL); if (err < 0) { logf("Unable to install alternative signal stack: %s", strerror(err)); return err; } #endif /* fill buffer with silence to initiate playback without noisy click */ sample_size = buffer_size; samples = calloc(1, sample_size * channels * sizeof(sample_t)); snd_pcm_format_set_silence(format, samples, sample_size); err = snd_pcm_writei(handle, samples, sample_size); free(samples); if (err < 0) { logf("Initial write error: %s\n", snd_strerror(err)); return err; } if (err != (ssize_t)sample_size) { logf("Initial write error: written %i expected %li\n", err, sample_size); return err; } snd_pcm_state_t state = snd_pcm_state(handle); logf("PCM RW State %d", state); if (state == SND_PCM_STATE_PREPARED) { err = snd_pcm_start(handle); if (err < 0) { logf("Start error: %s\n", snd_strerror(err)); return err; } } else { return state; } return 0; } void cleanup(void) { free(frames); frames = NULL; snd_pcm_close(handle); } void pcm_play_dma_init(void) { int err; logf("PCM DMA Init"); audiohw_preinit(); if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { panicf("%s(): Cannot open device %s: %s\n", __func__, device, snd_strerror(err)); } if ((err = snd_pcm_nonblock(handle, 1))) panicf("Could not set non-block mode: %s\n", snd_strerror(err)); if ((err = set_hwparams(handle)) < 0) { panicf("Setting of hwparams failed: %s\n", snd_strerror(err)); } if ((err = set_swparams(handle)) < 0) { panicf("Setting of swparams failed: %s\n", snd_strerror(err)); } pcm_dma_apply_settings(); #ifdef USE_ASYNC_CALLBACK pthread_mutexattr_t attr; pthread_mutexattr_init(&attr); pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE); pthread_mutex_init(&pcm_mtx, &attr); #else tick_add_task(pcm_tick); #endif atexit(cleanup); return; } void pcm_play_lock(void) { #ifdef USE_ASYNC_CALLBACK pthread_mutex_lock(&pcm_mtx); #else if (recursion++ == 0) tick_remove_task(pcm_tick); #endif } void pcm_play_unlock(void) { #ifdef USE_ASYNC_CALLBACK pthread_mutex_unlock(&pcm_mtx); #else if (--recursion == 0) tick_add_task(pcm_tick); #endif } static void pcm_dma_apply_settings_nolock(void) { logf("PCM DMA Settings %d %d", last_sample_rate, pcm_sampr); if (last_sample_rate != pcm_sampr) { last_sample_rate = pcm_sampr; #ifdef AUDIOHW_MUTE_ON_SRATE_CHANGE // XXX AK4450 (xDuoo X3ii) needs to be muted when switching rates. audiohw_mute(true); #endif snd_pcm_drop(handle); set_hwparams(handle); #if defined(HAVE_NWZ_LINUX_CODEC) /* Sony NWZ linux driver uses a nonstandard mecanism to set the sampling rate */ audiohw_set_frequency(pcm_sampr); #endif #ifdef AUDIOHW_MUTE_ON_SRATE_CHANGE audiohw_mute(false); #endif /* (Will be unmuted by pcm resuming) */ } } void pcm_dma_apply_settings(void) { pcm_play_lock(); pcm_dma_apply_settings_nolock(); pcm_play_unlock(); } void pcm_play_dma_pause(bool pause) { logf("PCM DMA pause %d", pause); #ifdef AUDIOHW_MUTE_ON_PAUSE if (pause) audiohw_mute(true); #endif snd_pcm_pause(handle, pause); #ifdef AUDIOHW_MUTE_ON_PAUSE if (!pause) audiohw_mute(false); #endif } void pcm_play_dma_stop(void) { snd_pcm_nonblock(handle, 0); snd_pcm_drain(handle); snd_pcm_nonblock(handle, 1); last_sample_rate = 0; #ifdef AUDIOHW_MUTE_ON_PAUSE audiohw_mute(true); #endif logf("PCM DMA stopped"); } void pcm_play_dma_start(const void *addr, size_t size) { logf("PCM DMA start (%p %d)", addr, size); pcm_dma_apply_settings_nolock(); pcm_data = addr; pcm_size = size; #if !defined(AUDIOHW_MUTE_ON_PAUSE) || !defined(AUDIOHW_MUTE_ON_SRATE_CHANGE) audiohw_mute(false); #endif while (1) { snd_pcm_state_t state = snd_pcm_state(handle); logf("PCM State %d", state); switch (state) { case SND_PCM_STATE_RUNNING: return; case SND_PCM_STATE_XRUN: { logf("Trying to recover from error\n"); int err = snd_pcm_recover(handle, -EPIPE, 0); if (err < 0) logf("Recovery failed: %s\n", snd_strerror(err)); continue; } case SND_PCM_STATE_SETUP: { int err = snd_pcm_prepare(handle); if (err < 0) logf("Prepare error: %s\n", snd_strerror(err)); /* fall through */ } case SND_PCM_STATE_PREPARED: { /* prepared state, we need to fill the buffer with silence before * starting */ int err = async_rw(handle); if (err < 0) { logf("Start error: %s\n", snd_strerror(err)); return; } #ifdef AUDIOHW_MUTE_ON_PAUSE audiohw_mute(false); #endif if (err == 0) return; break; } case SND_PCM_STATE_PAUSED: { /* paused, simply resume */ pcm_play_dma_pause(0); return; } case SND_PCM_STATE_DRAINING: /* run until drained */ continue; default: logf("Unhandled state: %s\n", snd_pcm_state_name(state)); return; } } } size_t pcm_get_bytes_waiting(void) { return pcm_size; } const void * pcm_play_dma_get_peak_buffer(int *count) { uintptr_t addr = (uintptr_t)pcm_data; *count = pcm_size / 4; return (void *)((addr + 3) & ~3); } void pcm_play_dma_postinit(void) { audiohw_postinit(); } void pcm_set_mixer_volume(int volume) { (void)volume; } int pcm_alsa_get_rate(void) { return real_sample_rate; } #ifdef HAVE_RECORDING void pcm_rec_lock(void) { } void pcm_rec_unlock(void) { } void pcm_rec_dma_init(void) { } void pcm_rec_dma_close(void) { } void pcm_rec_dma_start(void *start, size_t size) { (void)start; (void)size; } void pcm_rec_dma_stop(void) { } const void * pcm_rec_dma_get_peak_buffer(void) { return NULL; } void audiohw_set_recvol(int left, int right, int type) { (void)left; (void)right; (void)type; } #endif /* HAVE_RECORDING */