// SPDX-License-Identifier: GPL-2.0 /* * ALSA Virtual Soundcard * * alsa-pcm.c - AVIRT ALSA PCM interface * * Copyright (C) 2010-2018 Fiberdyne Systems Pty Ltd */ #include "core_internal.h" #define DO_AUDIOPATH_CB(ap, callback, substream, ...) \ do { \ if ((ap)->pcm_ops->callback) { \ return (ap)->pcm_ops->callback((substream), \ ##__VA_ARGS__); \ } \ } while (0) /** * avirt_pcm_period_elapsed - PCM buffer complete callback * @substreamid: pointer to ALSA PCM substream * * This should be called from a child Audio Path once it has finished processing * the pcm buffer */ void avirt_pcm_period_elapsed(struct snd_pcm_substream *substream) { // Notify ALSA middle layer of the elapsed period boundary snd_pcm_period_elapsed(substream); } EXPORT_SYMBOL_GPL(avirt_pcm_period_elapsed); /******************************************************************************* * ALSA PCM Callbacks ******************************************************************************/ /** * pcm_open - Implements 'open' callback for PCM middle layer * @substream: pointer to ALSA PCM substream * * This is called when an ALSA PCM substream is opened. The substream device is * configured here. * * Returns 0 on success or error code otherwise. */ static int pcm_open(struct snd_pcm_substream *substream) { struct avirt_audiopath *audiopath; struct avirt_stream *stream; struct snd_pcm_hardware *hw; unsigned int bytes_per_sample = 0, blocksize = 0, chans = 0; char *uid = "ap_fddsp"; // TD MF: Make this dynamic! audiopath = avirt_audiopath_get(uid); CHK_NULL_V(audiopath, -EFAULT, "Cannot find Audio Path uid: '%s'!", uid); substream->private_data = audiopath; blocksize = audiopath->blocksize; // Copy the hw params from the audiopath to the pcm hw = &substream->runtime->hw; memcpy(hw, audiopath->hw, sizeof(struct snd_pcm_hardware)); pr_info("%s %d %d", __func__, blocksize, hw->periods_max); if (hw->formats == SNDRV_PCM_FMTBIT_S16_LE) { bytes_per_sample = 2; } else { pr_err("[%s] PCM only supports SNDRV_PCM_FMTBIT_S16_LE", __func__); return -EINVAL; } stream = __avirt_stream_find_by_device(substream->pcm->device); if (IS_ERR_VALUE(stream) || !stream) return PTR_ERR(stream); // Setup remaining hw properties chans = stream->channels; hw->channels_min = chans; hw->channels_max = chans; hw->buffer_bytes_max = blocksize * hw->periods_max * bytes_per_sample * chans; hw->period_bytes_min = blocksize * bytes_per_sample * chans; hw->period_bytes_max = blocksize * bytes_per_sample * chans; // Do additional Audio Path 'open' callback DO_AUDIOPATH_CB(audiopath, open, substream); return 0; } /** * pcm_close - Implements 'close' callback for PCM middle layer * @substream: pointer to ALSA PCM substream * * This is called when a PCM substream is closed. * * Returns 0 on success or error code otherwise. */ static int pcm_close(struct snd_pcm_substream *substream) { DINFO(AP_LOGNAME, ""); // Do additional Audio Path 'close' callback DO_AUDIOPATH_CB(((struct avirt_audiopath *)substream->private_data), close, substream); return 0; } /** * pcm_hw_params - Implements 'hw_params' callback for PCM middle layer * @substream: pointer to ALSA PCM substream * @hw_params: contains the hardware parameters for the PCM * * This is called when the hardware parameters are set, including buffer size, * the period size, the format, etc. The buffer allocation should be done here. * * Returns 0 on success or error code otherwise. */ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { int err; size_t bufsz; struct avirt_audiopath *audiopath; struct avirt_stream *stream; stream = __avirt_stream_find_by_device(substream->pcm->device); if (IS_ERR_VALUE(stream) || !stream) return PTR_ERR(stream); if ((params_channels(hw_params) > stream->channels) || (params_channels(hw_params) < stream->channels)) { pr_err("Requested number of channels not supported.\n"); return -EINVAL; } audiopath = ((struct avirt_audiopath *)substream->private_data); bufsz = params_buffer_bytes(hw_params) * audiopath->hw->periods_max; err = snd_pcm_lib_alloc_vmalloc_buffer(substream, bufsz); if (err <= 0) { pr_err("pcm: buffer allocation failed (%d)\n", err); return err; } // Do additional Audio Path 'hw_params' callback DO_AUDIOPATH_CB(hw_params, substream, hw_params); return 0; } /** * pcm_hw_free - Implements 'hw_free' callback for PCM middle layer * @substream: pointer to ALSA PCM substream * * Release the resources allocated via 'hw_params'. * This function is always called before the 'close' callback . * * Returns 0 on success or error code otherwise. */ static int pcm_hw_free(struct snd_pcm_substream *substream) { DINFO(AP_LOGNAME, ""); CHK_ERR(snd_pcm_lib_free_vmalloc_buffer(substream)); // Do additional Audio Path 'hw_free' callback DO_AUDIOPATH_CB(hw_free, substream); return 0; } /** * pcm_prepare - Implements 'prepare' callback for PCM middle layer * @substream: pointer to ALSA PCM substream * * The format rate, sample rate, etc., can be set here. This callback can be * called many times at each setup. This function is also used to handle overrun * and underrun conditions when we try and resync the DMA (if we're using DMA). * * Returns 0 on success or error code otherwise. */ static int pcm_prepare(struct snd_pcm_substream *substream) { // Do additional Audio Path 'prepare' callback DO_AUDIOPATH_CB(((struct avirt_audiopath *)substream->private_data), prepare, substream); return 0; } /** * pcm_trigger - Implements 'trigger' callback for PCM middle layer * @substream: pointer to ALSA PCM substream * @cmd: action to be performed (start or stop) * * This is called when the PCM is started, stopped or paused. The action * indicated action is specified in the second argument, SNDRV_PCM_TRIGGER_XXX * * Returns 0 on success or error code otherwise. */ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) { switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: break; default: pr_err("trigger must be START or STOP"); return -EINVAL; } // Do additional Audio Path 'trigger' callback DO_AUDIOPATH_CB(((struct avirt_audiopath *)substream->private_data), trigger, substream, cmd); return 0; } /** * pcm_pointer - Implements 'pointer' callback for PCM middle layer * @substream: pointer to ALSA PCM substream * * This gets called when the user space needs a DMA buffer index. IO errors will * be generated if the index does not increment, or drives beyond the frame * threshold of the buffer itself. * * Returns the current hardware buffer frame index. */ static snd_pcm_uframes_t pcm_pointer(struct snd_pcm_substream *substream) { // Do additional Audio Path 'pointer' callback DO_AUDIOPATH_CB(((struct avirt_audiopath *)substream->private_data), pointer, substream); return 0; } /** * pcm_pointer - Implements 'get_time_info' callback for PCM middle layer * @substream: pointer to ALSA PCM substream * @system_ts * @audio_ts * @audio_tstamp_config * @audio_tstamp_report * * Generic way to get system timestamp and audio timestamp info * * Returns 0 on success or error code otherwise */ static int pcm_get_time_info( struct snd_pcm_substream *substream, struct timespec *system_ts, struct timespec *audio_ts, struct snd_pcm_audio_tstamp_config *audio_tstamp_config, struct snd_pcm_audio_tstamp_report *audio_tstamp_report) { DO_AUDIOPATH_CB(((struct avirt_audiopath *)substream->private_data), get_time_info, substream, system_ts, audio_ts, audio_tstamp_config, audio_tstamp_report); return 0; } /** * pcm_copy_user - Implements 'copy_user' callback for PCM middle layer * @substream: pointer to ALSA PCM substream * @channel: * @pos: The offset in the DMA * @src: Audio PCM data from the user space * @count: * * This is where we need to copy user audio PCM data into the sound driver * * Returns 0 on success or error code otherwise. * */ static int pcm_copy_user(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void __user *src, snd_pcm_uframes_t count) { // struct snd_pcm_runtime *runtime; // int offset; // runtime = substream->runtime; // offset = frames_to_bytes(runtime, pos); // Do additional Audio Path 'copy_user' callback DO_AUDIOPATH_CB(((struct avirt_audiopath *)substream->private_data), copy_user, substream, channel, pos, src, count); return 0; } /** * pcm_copy_kernel - Implements 'copy_kernel' callback for PCM middle layer * @substream: pointer to ALSA PCM substream * @channel: * @pos: The offset in the DMA * @src: Audio PCM data from the user space * @count: * * This is where we need to copy kernel audio PCM data into the sound driver * * Returns 0 on success or error code otherwise. * */ static int pcm_copy_kernel(struct snd_pcm_substream *substream, int channel, unsigned long pos, void *buf, unsigned long count) { DO_AUDIOPATH_CB(((struct avirt_audiopath *)substream->private_data), copy_kernel, substream, channel, pos, buf, count); return 0; } /** * pcm_ack - Implements 'ack' callback for PCM middle layer * @substream: pointer to ALSA PCM substream * * This is where we need to ack * * Returns 0 on success or error code otherwise. * */ static int pcm_ack(struct snd_pcm_substream *substream) { DO_AUDIOPATH_CB(((struct avirt_audiopath *)substream->private_data), ack, substream); return 0; } static int pcm_silence(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, snd_pcm_uframes_t count) { DO_AUDIOPATH_CB(((struct avirt_audiopath *)substream->private_data), fill_silence, substream, channel, pos, count); return 0; } struct snd_pcm_ops pcm_ops = { .open = pcm_open, .close = pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = pcm_hw_params, .hw_free = pcm_hw_free, .prepare = pcm_prepare, .trigger = pcm_trigger, .pointer = pcm_pointer, .get_time_info = pcm_get_time_info, .fill_silence = pcm_silence, .copy_user = pcm_copy_user, .copy_kernel = pcm_copy_kernel, .page = snd_pcm_lib_get_vmalloc_page, .ack = pcm_ack, };