893 lines
25 KiB
C
893 lines
25 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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*
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* Copyright (C) 2010 Thomas Martitz
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* Copyright (c) 2020 Solomon Peachy
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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/*
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* Based, but heavily modified, on the example given at
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* http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html
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*
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* This driver uses the so-called unsafe async callback method.
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*
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* To make the async callback safer, an alternative stack is installed, since
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* it's run from a signal hanlder (which otherwise uses the user stack).
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*
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* TODO: Rewrite this to properly use multithreading and/or direct mmap()
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*/
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#include "autoconf.h"
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#include <stdlib.h>
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#include <stdbool.h>
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#include <alsa/asoundlib.h>
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//#define LOGF_ENABLE
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#include "system.h"
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#include "debug.h"
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#include "kernel.h"
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#include "panic.h"
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#include "pcm.h"
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#include "pcm-internal.h"
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#include "pcm_mixer.h"
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#include "pcm_sampr.h"
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#include "audiohw.h"
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#include "pcm-alsa.h"
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#include "logf.h"
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#include <pthread.h>
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#include <signal.h>
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/* plughw:0,0 works with both, however "default" is recommended.
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* default doesnt seem to work with async callback but doesn't break
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* with multple applications running */
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#define DEFAULT_PLAYBACK_DEVICE "plughw:0,0"
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#define DEFAULT_CAPTURE_DEVICE "default"
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#if MIX_FRAME_SAMPLES < 512
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#error "MIX_FRAME_SAMPLES needs to be at least 512!"
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#elif MIX_FRAME_SAMPLES < 1024
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#warning "MIX_FRAME_SAMPLES <1024 may cause dropouts!"
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#endif
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static const snd_pcm_access_t access_ = SND_PCM_ACCESS_RW_INTERLEAVED; /* access mode */
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#if defined(HAVE_ALSA_32BIT)
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static const snd_pcm_format_t format = SND_PCM_FORMAT_S32_LE; /* sample format */
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typedef int32_t sample_t;
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#else
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static const snd_pcm_format_t format = SND_PCM_FORMAT_S16; /* sample format */
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typedef int16_t sample_t;
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#endif
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static const int channels = 2; /* count of channels */
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static unsigned int real_sample_rate;
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static unsigned int last_sample_rate;
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static snd_pcm_t *handle = NULL;
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static snd_pcm_sframes_t buffer_size;
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static snd_pcm_sframes_t period_size;
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static sample_t *frames = NULL;
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static const void *pcm_data = 0;
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static size_t pcm_size = 0;
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static unsigned int xruns = 0;
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static snd_async_handler_t *ahandler = NULL;
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static pthread_mutex_t pcm_mtx;
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static char signal_stack[SIGSTKSZ];
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static const char *playback_dev = DEFAULT_PLAYBACK_DEVICE;
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#ifdef HAVE_RECORDING
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static void *pcm_data_rec = DEFAULT_CAPTURE_DEVICE;
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static const char *capture_dev = NULL;
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static snd_pcm_stream_t current_alsa_mode; /* SND_PCM_STREAM_PLAYBACK / _CAPTURE */
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#endif
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static const char *current_alsa_device;
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void pcm_alsa_set_playback_device(const char *device)
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{
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playback_dev = device;
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}
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#ifdef HAVE_RECORDING
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void pcm_alsa_set_capture_device(const char *device)
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{
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capture_dev = device;
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}
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#endif
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static int set_hwparams(snd_pcm_t *handle)
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{
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int err;
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unsigned int srate;
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snd_pcm_hw_params_t *params;
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snd_pcm_hw_params_malloc(¶ms);
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/* Size playback buffers based on sample rate.
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Buffer size must be at least 4x period size!
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Note these are in FRAMES, and are sized to be about 8.5ms
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for the buffer and 2.1ms for the period
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*/
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if (pcm_sampr > SAMPR_96) {
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buffer_size = MIX_FRAME_SAMPLES * 4 * 4;
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period_size = MIX_FRAME_SAMPLES * 4;
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} else if (pcm_sampr > SAMPR_48) {
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buffer_size = MIX_FRAME_SAMPLES * 2 * 4;
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period_size = MIX_FRAME_SAMPLES * 2;
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} else {
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buffer_size = MIX_FRAME_SAMPLES * 4;
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period_size = MIX_FRAME_SAMPLES;
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}
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/* choose all parameters */
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err = snd_pcm_hw_params_any(handle, params);
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if (err < 0)
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{
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panicf("Broken configuration for playback: no configurations available: %s", snd_strerror(err));
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goto error;
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}
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/* set the interleaved read/write format */
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err = snd_pcm_hw_params_set_access(handle, params, access_);
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if (err < 0)
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{
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panicf("Access type not available for playback: %s", snd_strerror(err));
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goto error;
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}
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/* set the sample format */
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err = snd_pcm_hw_params_set_format(handle, params, format);
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if (err < 0)
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{
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logf("Sample format not available for playback: %s", snd_strerror(err));
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goto error;
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}
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/* set the count of channels */
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err = snd_pcm_hw_params_set_channels(handle, params, channels);
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if (err < 0)
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{
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logf("Channels count (%i) not available for playbacks: %s", channels, snd_strerror(err));
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goto error;
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}
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/* set the stream rate */
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srate = pcm_sampr;
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err = snd_pcm_hw_params_set_rate_near(handle, params, &srate, 0);
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if (err < 0)
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{
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logf("Rate %luHz not available for playback: %s", pcm_sampr, snd_strerror(err));
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goto error;
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}
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real_sample_rate = srate;
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if (real_sample_rate != pcm_sampr)
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{
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logf("Rate doesn't match (requested %luHz, get %dHz)", pcm_sampr, real_sample_rate);
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err = -EINVAL;
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goto error;
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}
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/* set the buffer size */
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err = snd_pcm_hw_params_set_buffer_size_near(handle, params, &buffer_size);
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if (err < 0)
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{
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logf("Unable to set buffer size %ld for playback: %s", buffer_size, snd_strerror(err));
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goto error;
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}
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/* set the period size */
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err = snd_pcm_hw_params_set_period_size_near (handle, params, &period_size, NULL);
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if (err < 0)
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{
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logf("Unable to set period size %ld for playback: %s", period_size, snd_strerror(err));
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goto error;
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}
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if (frames) free(frames);
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frames = calloc(1, period_size * channels * sizeof(sample_t));
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/* write the parameters to device */
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err = snd_pcm_hw_params(handle, params);
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if (err < 0)
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{
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logf("Unable to set hw params for playback: %s", snd_strerror(err));
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goto error;
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}
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err = 0; /* success */
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error:
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snd_pcm_hw_params_free(params);
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return err;
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}
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/* Set sw params: playback start threshold and low buffer watermark */
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static int set_swparams(snd_pcm_t *handle)
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{
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int err;
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snd_pcm_sw_params_t *swparams;
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snd_pcm_sw_params_malloc(&swparams);
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/* get the current swparams */
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err = snd_pcm_sw_params_current(handle, swparams);
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if (err < 0)
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{
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logf("Unable to determine current swparams for playback: %s", snd_strerror(err));
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goto error;
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}
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/* start the transfer when the buffer is half full */
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err = snd_pcm_sw_params_set_start_threshold(handle, swparams, buffer_size / 2);
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if (err < 0)
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{
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logf("Unable to set start threshold mode for playback: %s", snd_strerror(err));
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goto error;
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}
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/* allow the transfer when at least period_size samples can be processed */
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err = snd_pcm_sw_params_set_avail_min(handle, swparams, period_size);
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if (err < 0)
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{
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logf("Unable to set avail min for playback: %s", snd_strerror(err));
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goto error;
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}
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/* write the parameters to the playback device */
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err = snd_pcm_sw_params(handle, swparams);
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if (err < 0)
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{
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logf("Unable to set sw params for playback: %s", snd_strerror(err));
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goto error;
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}
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err = 0; /* success */
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error:
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snd_pcm_sw_params_free(swparams);
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return err;
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}
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#if defined(HAVE_ALSA_32BIT)
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/* Digital volume explanation:
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* with very good approximation (<0.1dB) the convertion from dB to multiplicative
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* factor, for dB>=0, is 2^(dB/3). We can then notice that if we write dB=3*k+r
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* then this is 2^k*2^(r/3) so we only need to look at r=0,1,2. For r=0 this is
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* 1, for r=1 we have 2^(1/3)~=1.25 so we approximate by 1+1/4, and 2^(2/3)~=1.5
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* so we approximate by 1+1/2. To go from negative to nonnegative we notice that
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* 48 dB => 63095 factor ~= 2^16 so we virtually pre-multiply everything by 2^(-16)
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* and add 48dB to the input volume. We cannot go lower -43dB because several
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* values between -48dB and -43dB would require a fractional multiplier, which is
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* stupid to implement for such very low volume. */
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static int dig_vol_mult_l = 2 << 16; /* multiplicative factor to apply to each sample */
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static int dig_vol_mult_r = 2 << 16; /* multiplicative factor to apply to each sample */
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void pcm_set_mixer_volume(int vol_db_l, int vol_db_r)
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{
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if(vol_db_l > 0 || vol_db_r > 0 || vol_db_l < -43 || vol_db_r < -43)
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panicf("invalid pcm alsa volume %d %d", vol_db_l, vol_db_r);
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if(format != SND_PCM_FORMAT_S32_LE)
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panicf("this function assumes 32-bit sample size");
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vol_db_l += 48; /* -42dB .. 0dB => 5dB .. 48dB */
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vol_db_r += 48; /* -42dB .. 0dB => 5dB .. 48dB */
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/* NOTE if vol_dB = 5 then vol_shift = 1 but r = 1 so we do vol_shift - 1 >= 0
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* otherwise vol_dB >= 0 implies vol_shift >= 2 so vol_shift - 2 >= 0 */
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int vol_shift_l = vol_db_l / 3;
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int vol_shift_r = vol_db_r / 3;
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int r_l = vol_db_l % 3;
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int r_r = vol_db_r % 3;
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if(r_l == 0)
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dig_vol_mult_l = 1 << vol_shift_l;
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else if(r_l == 1)
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dig_vol_mult_l = 1 << vol_shift_l | 1 << (vol_shift_l - 2);
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else
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dig_vol_mult_l = 1 << vol_shift_l | 1 << (vol_shift_l - 1);
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logf("l: %d dB -> factor = %d", vol_db_l - 48, dig_vol_mult_l);
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if(r_r == 0)
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dig_vol_mult_r = 1 << vol_shift_r;
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else if(r_r == 1)
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dig_vol_mult_r = 1 << vol_shift_r | 1 << (vol_shift_r - 2);
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else
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dig_vol_mult_r = 1 << vol_shift_r | 1 << (vol_shift_r - 1);
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logf("r: %d dB -> factor = %d", vol_db_r - 48, dig_vol_mult_r);
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}
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#endif
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/* copy pcm samples to a spare buffer, suitable for snd_pcm_writei() */
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static bool copy_frames(bool first)
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{
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ssize_t nframes, frames_left = period_size;
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bool new_buffer = false;
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while (frames_left > 0)
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{
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if (!pcm_size)
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{
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new_buffer = true;
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#ifdef HAVE_RECORDING
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switch (current_alsa_mode)
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{
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case SND_PCM_STREAM_PLAYBACK:
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#endif
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if (!pcm_play_dma_complete_callback(PCM_DMAST_OK, &pcm_data, &pcm_size))
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{
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return false;
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}
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#ifdef HAVE_RECORDING
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break;
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case SND_PCM_STREAM_CAPTURE:
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if (!pcm_play_dma_complete_callback(PCM_DMAST_OK, &pcm_data, &pcm_size))
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{
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return false;
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}
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break;
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default:
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break;
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}
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#endif
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}
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/* Note: This assumes stereo 16-bit */
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if (pcm_size % 4)
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panicf("Wrong pcm_size");
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/* the compiler will optimize this test away */
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nframes = MIN((ssize_t)pcm_size/4, frames_left);
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#ifdef HAVE_RECORDING
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switch (current_alsa_mode)
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{
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case SND_PCM_STREAM_PLAYBACK:
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#endif
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#if defined(HAVE_ALSA_32BIT)
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if (format == SND_PCM_FORMAT_S32_LE)
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{
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/* We have to convert 16-bit to 32-bit, the need to multiply the
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* sample by some value so the sound is not too low */
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const int16_t *pcm_ptr = pcm_data;
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sample_t *sample_ptr = &frames[2*(period_size-frames_left)];
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for (int i = 0; i < nframes; i++)
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{
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*sample_ptr++ = *pcm_ptr++ * dig_vol_mult_l;
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*sample_ptr++ = *pcm_ptr++ * dig_vol_mult_r;
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}
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}
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else
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#endif
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{
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/* Rockbox and PCM have same format: memcopy */
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memcpy(&frames[2*(period_size-frames_left)], pcm_data, nframes * 4);
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}
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#ifdef HAVE_RECORDING
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break;
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case SND_PCM_STREAM_CAPTURE:
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memcpy(pcm_data_rec, &frames[2*(period_size-frames_left)], nframes * 4);
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break;
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default:
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break;
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}
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#endif
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pcm_data += nframes*4;
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pcm_size -= nframes*4;
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frames_left -= nframes;
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if (new_buffer && !first)
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{
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new_buffer = false;
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#ifdef HAVE_RECORDING
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switch (current_alsa_mode)
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{
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case SND_PCM_STREAM_PLAYBACK:
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#endif
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pcm_play_dma_status_callback(PCM_DMAST_STARTED);
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#ifdef HAVE_RECORDING
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break;
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case SND_PCM_STREAM_CAPTURE:
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pcm_rec_dma_status_callback(PCM_DMAST_STARTED);
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break;
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default:
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break;
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}
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#endif
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}
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}
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return true;
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}
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static void async_callback(snd_async_handler_t *ahandler)
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{
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int err;
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if (!ahandler) return;
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snd_pcm_t *handle = snd_async_handler_get_pcm(ahandler);
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if (!handle) return;
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if (pthread_mutex_trylock(&pcm_mtx) != 0)
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return;
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snd_pcm_state_t state = snd_pcm_state(handle);
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if (state == SND_PCM_STATE_XRUN)
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{
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xruns++;
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logf("initial underrun!");
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err = snd_pcm_recover(handle, -EPIPE, 0);
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if (err < 0) {
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logf("XRUN Recovery error: %s", snd_strerror(err));
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goto abort;
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}
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}
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else if (state == SND_PCM_STATE_DRAINING)
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{
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logf("draining...");
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goto abort;
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}
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else if (state == SND_PCM_STATE_SETUP)
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{
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goto abort;
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}
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#ifdef HAVE_RECORDING
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if (current_alsa_mode == SND_PCM_STREAM_PLAYBACK)
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{
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#endif
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while (snd_pcm_avail_update(handle) >= period_size)
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{
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if (copy_frames(false))
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{
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retry:
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err = snd_pcm_writei(handle, frames, period_size);
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if (err == -EPIPE)
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{
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logf("mid underrun!");
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xruns++;
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err = snd_pcm_recover(handle, -EPIPE, 0);
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if (err < 0) {
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logf("XRUN Recovery error: %s", snd_strerror(err));
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goto abort;
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}
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goto retry;
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}
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else if (err != period_size)
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{
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logf("Write error: written %i expected %li", err, period_size);
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break;
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}
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}
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else
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{
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logf("%s: No Data (%d).", __func__, state);
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break;
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}
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}
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#ifdef HAVE_RECORDING
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}
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else if (current_alsa_mode == SND_PCM_STREAM_CAPTURE)
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{
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while (snd_pcm_avail_update(handle) >= period_size)
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{
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int err = snd_pcm_readi(handle, frames, period_size);
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if (err == -EPIPE)
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{
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logf("rec mid underrun!");
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xruns++;
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err = snd_pcm_recover(handle, -EPIPE, 0);
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if (err < 0) {
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logf("XRUN Recovery error: %s", snd_strerror(err));
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goto abort;
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}
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continue; /* buffer contents trashed, no sense in trying to copy */
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}
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else if (err != period_size)
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{
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logf("Read error: read %i expected %li", err, period_size);
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break;
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}
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|
|
|
/* start the fake DMA transfer */
|
|
if (!copy_frames(false))
|
|
{
|
|
/* do not spam logf */
|
|
/* logf("%s: No Data.", __func__); */
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
#endif
|
|
|
|
if (snd_pcm_state(handle) == SND_PCM_STATE_PREPARED)
|
|
{
|
|
err = snd_pcm_start(handle);
|
|
if (err < 0) {
|
|
logf("cb start error: %s", snd_strerror(err));
|
|
/* Depending on the error we might be SOL */
|
|
}
|
|
}
|
|
|
|
abort:
|
|
pthread_mutex_unlock(&pcm_mtx);
|
|
}
|
|
|
|
static void close_hwdev(void)
|
|
{
|
|
logf("closedev (%p)", handle);
|
|
|
|
if (handle) {
|
|
snd_pcm_drain(handle);
|
|
#ifdef AUDIOHW_MUTE_ON_STOP
|
|
audiohw_mute(true);
|
|
#endif
|
|
if (ahandler) {
|
|
snd_async_del_handler(ahandler);
|
|
ahandler = NULL;
|
|
}
|
|
snd_pcm_close(handle);
|
|
|
|
handle = NULL;
|
|
}
|
|
current_alsa_device = NULL;
|
|
|
|
#ifdef HAVE_RECORDING
|
|
pcm_data_rec = NULL;
|
|
#endif
|
|
}
|
|
|
|
static void alsadev_cleanup(void)
|
|
{
|
|
free(frames);
|
|
frames = NULL;
|
|
close_hwdev();
|
|
}
|
|
|
|
static void open_hwdev(const char *device, snd_pcm_stream_t mode)
|
|
{
|
|
int err;
|
|
|
|
logf("opendev %s (%p)", device, handle);
|
|
|
|
if (handle && device == current_alsa_device
|
|
#ifdef HAVE_RECORDING
|
|
&& current_alsa_mode == mode
|
|
#endif
|
|
)
|
|
{
|
|
return;
|
|
}
|
|
|
|
/* Close old handle */
|
|
close_hwdev();
|
|
|
|
if ((err = snd_pcm_open(&handle, device, mode, 0)) < 0)
|
|
{
|
|
panicf("%s(): Cannot open device %s: %s", __func__, device, snd_strerror(err));
|
|
}
|
|
last_sample_rate = 0;
|
|
|
|
pthread_mutexattr_t attr;
|
|
pthread_mutexattr_init(&attr);
|
|
pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE);
|
|
pthread_mutex_init(&pcm_mtx, &attr);
|
|
|
|
/* 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)
|
|
{
|
|
panicf("Unable to install alternative signal stack: %s", strerror(err));
|
|
}
|
|
|
|
err = snd_async_add_pcm_handler(&ahandler, handle, async_callback, NULL);
|
|
if (err < 0)
|
|
{
|
|
panicf("Unable to register async handler: %s", snd_strerror(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)
|
|
{
|
|
panicf("Unable to install alternative signal stack: %s", strerror(err));
|
|
}
|
|
|
|
#ifdef HAVE_RECORDING
|
|
current_alsa_mode = mode;
|
|
#else
|
|
(void)mode;
|
|
#endif
|
|
current_alsa_device = device;
|
|
|
|
atexit(alsadev_cleanup);
|
|
}
|
|
|
|
void pcm_play_dma_init(void)
|
|
{
|
|
logf("PCM DMA Init");
|
|
|
|
audiohw_preinit();
|
|
|
|
open_hwdev(playback_dev, SND_PCM_STREAM_PLAYBACK);
|
|
|
|
return;
|
|
}
|
|
|
|
void pcm_play_lock(void)
|
|
{
|
|
pthread_mutex_lock(&pcm_mtx);
|
|
}
|
|
|
|
void pcm_play_unlock(void)
|
|
{
|
|
pthread_mutex_unlock(&pcm_mtx);
|
|
}
|
|
|
|
static void pcm_dma_apply_settings_nolock(void)
|
|
{
|
|
logf("PCM DMA Settings %d %lu", last_sample_rate, pcm_sampr);
|
|
|
|
if (last_sample_rate != pcm_sampr)
|
|
{
|
|
last_sample_rate = pcm_sampr;
|
|
|
|
#ifdef AUDIOHW_MUTE_ON_SRATE_CHANGE
|
|
audiohw_mute(true);
|
|
#endif
|
|
snd_pcm_drop(handle);
|
|
|
|
set_hwparams(handle); // FIXME: check return code?
|
|
set_swparams(handle); // FIXME: check return code?
|
|
|
|
#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
|
|
/* (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_stop(void)
|
|
{
|
|
logf("PCM DMA stop (%d)", snd_pcm_state(handle));
|
|
|
|
int err = snd_pcm_drain(handle);
|
|
if (err < 0)
|
|
if (err < 0)
|
|
logf("Drain failed: %s", snd_strerror(err));
|
|
#ifdef AUDIOHW_MUTE_ON_STOP
|
|
audiohw_mute(true);
|
|
#endif
|
|
}
|
|
|
|
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_STOP) && 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:
|
|
#if defined(AUDIOHW_MUTE_ON_STOP)
|
|
audiohw_mute(false);
|
|
#endif
|
|
return;
|
|
case SND_PCM_STATE_XRUN:
|
|
{
|
|
logf("Trying to recover from underrun");
|
|
int err = snd_pcm_recover(handle, -EPIPE, 0);
|
|
if (err < 0)
|
|
logf("Recovery failed: %s", snd_strerror(err));
|
|
continue;
|
|
}
|
|
case SND_PCM_STATE_SETUP:
|
|
{
|
|
int err = snd_pcm_prepare(handle);
|
|
if (err < 0)
|
|
logf("Prepare error: %s", snd_strerror(err));
|
|
}
|
|
/* fall through */
|
|
case SND_PCM_STATE_PREPARED:
|
|
{
|
|
int err;
|
|
#if 0
|
|
/* fill buffer with silence to initiate playback without noisy click */
|
|
snd_pcm_sframes_t sample_size = buffer_size;
|
|
sample_t *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 != (ssize_t)sample_size)
|
|
{
|
|
logf("Initial write error: written %i expected %li", err, sample_size);
|
|
return;
|
|
}
|
|
#else
|
|
/* Fill buffer with proper sample data */
|
|
while (snd_pcm_avail_update(handle) >= period_size)
|
|
{
|
|
if (copy_frames(true))
|
|
{
|
|
err = snd_pcm_writei(handle, frames, period_size);
|
|
if (err < 0 && err != period_size && err != -EAGAIN)
|
|
{
|
|
logf("Write error: written %i expected %li", err, period_size);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
#endif
|
|
err = snd_pcm_start(handle);
|
|
if (err < 0) {
|
|
logf("start error: %s", snd_strerror(err));
|
|
/* We will recover on the next iteration */
|
|
}
|
|
|
|
break;
|
|
}
|
|
case SND_PCM_STATE_DRAINING:
|
|
/* run until drained */
|
|
continue;
|
|
default:
|
|
logf("Unhandled state: %s", snd_pcm_state_name(state));
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
void pcm_play_dma_postinit(void)
|
|
{
|
|
audiohw_postinit();
|
|
|
|
#ifdef AUDIOHW_NEEDS_INITIAL_UNMUTE
|
|
audiohw_mute(false);
|
|
#endif
|
|
}
|
|
|
|
unsigned int pcm_alsa_get_rate(void)
|
|
{
|
|
return real_sample_rate;
|
|
}
|
|
|
|
unsigned int pcm_alsa_get_xruns(void)
|
|
{
|
|
return xruns;
|
|
}
|
|
|
|
#ifdef HAVE_RECORDING
|
|
void pcm_rec_lock(void)
|
|
{
|
|
pcm_play_lock();
|
|
}
|
|
|
|
void pcm_rec_unlock(void)
|
|
{
|
|
pcm_play_unlock();
|
|
}
|
|
|
|
void pcm_rec_dma_init(void)
|
|
{
|
|
logf("PCM REC DMA Init");
|
|
|
|
open_hwdev(capture_dev, SND_PCM_STREAM_CAPTURE);
|
|
}
|
|
|
|
void pcm_rec_dma_close(void)
|
|
{
|
|
logf("Rec DMA Close");
|
|
// close_hwdev();
|
|
open_hwdev(playback_dev, SND_PCM_STREAM_PLAYBACK);
|
|
}
|
|
|
|
void pcm_rec_dma_start(void *start, size_t size)
|
|
{
|
|
logf("PCM REC DMA start (%p %d)", start, size);
|
|
pcm_dma_apply_settings_nolock();
|
|
pcm_data_rec = start;
|
|
pcm_size = size;
|
|
|
|
if (!handle) return;
|
|
|
|
while (1)
|
|
{
|
|
snd_pcm_state_t state = snd_pcm_state(handle);
|
|
|
|
switch (state)
|
|
{
|
|
case SND_PCM_STATE_RUNNING:
|
|
return;
|
|
case SND_PCM_STATE_XRUN:
|
|
{
|
|
logf("Trying to recover from error");
|
|
int err = snd_pcm_recover(handle, -EPIPE, 0);
|
|
if (err < 0)
|
|
panicf("Recovery failed: %s", snd_strerror(err));
|
|
continue;
|
|
}
|
|
case SND_PCM_STATE_SETUP:
|
|
{
|
|
int err = snd_pcm_prepare(handle);
|
|
if (err < 0)
|
|
panicf("Prepare error: %s", snd_strerror(err));
|
|
}
|
|
/* fall through */
|
|
case SND_PCM_STATE_PREPARED:
|
|
{
|
|
int err = snd_pcm_start(handle);
|
|
if (err < 0)
|
|
panicf("Start error: %s", snd_strerror(err));
|
|
return;
|
|
}
|
|
case SND_PCM_STATE_DRAINING:
|
|
/* run until drained */
|
|
continue;
|
|
default:
|
|
logf("Unhandled state: %s", snd_pcm_state_name(state));
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
void pcm_rec_dma_stop(void)
|
|
{
|
|
logf("Rec DMA Stop");
|
|
close_hwdev();
|
|
}
|
|
|
|
const void * pcm_rec_dma_get_peak_buffer(void)
|
|
{
|
|
uintptr_t addr = (uintptr_t)pcm_data_rec;
|
|
return (void*)((addr + 3) & ~3);
|
|
}
|
|
|
|
#ifdef SIMULATOR
|
|
void audiohw_set_recvol(int left, int right, int type)
|
|
{
|
|
(void)left;
|
|
(void)right;
|
|
(void)type;
|
|
}
|
|
#endif
|
|
|
|
#endif /* HAVE_RECORDING */
|