diff options
author | James O'Shannessy <james.oshannessy@fiberdyne.com.au> | 2018-08-27 15:08:14 +1000 |
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committer | Mark Farrugia <mark.farrugia@fiberdyne.com.au> | 2018-10-26 17:27:24 +1100 |
commit | bc8c3a602bceaba0e6d34a1ba8b776b56b00d766 (patch) | |
tree | ae8cec69c910144611e06f272033cc8c2aee7032 /alsa-pcm.c | |
parent | 416c9b0f9b816a6b2eb5c544f21567ad286dd4be (diff) |
Public push of AVIRT.
Follow readme for building in/out of tree for Ubuntu/AGL/etc.
Signed-off-by: James O'Shannessy <james.oshannessy@fiberdyne.com.au>
Diffstat (limited to 'alsa-pcm.c')
-rw-r--r-- | alsa-pcm.c | 381 |
1 files changed, 381 insertions, 0 deletions
diff --git a/alsa-pcm.c b/alsa-pcm.c new file mode 100644 index 0000000..bd705f4 --- /dev/null +++ b/alsa-pcm.c @@ -0,0 +1,381 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * ALSA Virtual Soundcard + * + * alsa-pcm.c - ALSA PCM implementation + * + * Copyright (C) 2010-2018 Fiberdyne Systems Pty Ltd + */ + +#include "core.h" +#include "alsa.h" + +#define DO_AUDIOPATH_CB(callback, substream, ...) \ + do { \ + struct avirt_audiopath *ap; \ + ap = avirt_get_current_audiopath(); \ + CHK_NULL_V(ap, "Cannot find Audio Path!"); \ + if (ap->pcm_ops->callback) { \ + return ap->pcm_ops->callback(substream, \ + ##__VA_ARGS__); \ + } \ + } while (0) + +/** + * configure_pcm - set up substream properties from user configuration + * @substream: pointer to ALSA PCM substream + * @return 0 on success or error code otherwise + */ +static int configure_pcm(struct snd_pcm_substream *substream) +{ + struct avirt_alsa_dev_config *config; + struct avirt_audiopath *audiopath; + struct avirt_alsa_dev_group *group; + struct snd_pcm_hardware *hw; + unsigned bytes_per_sample = 0, blocksize = 0; + + audiopath = avirt_get_current_audiopath(); + CHK_NULL_V(audiopath, "Cannot find Audio Path!"); + + 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; + } + + // Get device group (playback/capture) + group = avirt_alsa_get_dev_group(substream->stream); + CHK_NULL(group); + + // Check if substream id is valid + if (substream->pcm->device >= group->devices) + return -1; + + // Setup remaining hw properties + config = &group->config[substream->pcm->device]; + hw->channels_min = config->channels; + hw->channels_max = config->channels; + hw->buffer_bytes_max = blocksize * hw->periods_max * bytes_per_sample * + config->channels; + hw->period_bytes_min = blocksize * bytes_per_sample * config->channels; + hw->period_bytes_max = blocksize * bytes_per_sample * config->channels; + + return 0; +} + +/******************************************************************************* + * 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) +{ + // Setup the pcm device based on the configuration assigned + CHK_ERR_V(configure_pcm(substream), "Failed to setup pcm device"); + + // Do additional Audio Path 'open' callback + DO_AUDIOPATH_CB(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) +{ + // Do additional Audio Path 'close' callback + DO_AUDIOPATH_CB(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 channels, err; + size_t bufsz; + struct avirt_audiopath *audiopath; + struct avirt_alsa_dev_group *group; + + group = avirt_alsa_get_dev_group(substream->stream); + CHK_NULL(group); + + channels = group->config[substream->pcm->device].channels; + + if ((params_channels(hw_params) > channels) || + (params_channels(hw_params) < channels)) { + pr_err("Requested number of channels not supported.\n"); + return -EINVAL; + } + + audiopath = avirt_get_current_audiopath(); + CHK_NULL_V(audiopath, "Cannot find Audio Path!"); + + 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) +{ + 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) +{ + struct avirt_alsa_dev_group *group; + struct snd_pcm_runtime *runtime = substream->runtime; + + group = avirt_alsa_get_dev_group(substream->stream); + CHK_NULL(group); + + // Reset HW buffer index for the device + group->streams[substream->pcm->device].hw_frame_idx = 0; + + group->buffersize = frames_to_bytes(runtime, runtime->buffer_size); + + // Do additional Audio Path 'prepare' callback + DO_AUDIOPATH_CB(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) +{ + struct avirt_alsa_dev_group *group; + + group = avirt_alsa_get_dev_group(substream->stream); + CHK_NULL(group); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + group->streams[substream->pcm->device].substream = substream; + break; + 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(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) +{ + struct avirt_alsa_dev_group *group; + + group = avirt_alsa_get_dev_group(substream->stream); + CHK_NULL(group); + + return group->streams[substream->pcm->device].hw_frame_idx; +} + +/** + * 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) +{ + struct avirt_alsa_dev_group *group; + + group = avirt_alsa_get_dev_group(substream->stream); + CHK_NULL(group); + + DO_AUDIOPATH_CB(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(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(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. + * + */ +int pcm_ack(struct snd_pcm_substream *substream) +{ + DO_AUDIOPATH_CB(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(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, + .mmap = snd_pcm_lib_mmap_vmalloc, + .ack = pcm_ack, + +}; |