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linux-next/sound/soc/codecs/sdw-mockup.c

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ASoC: codecs: add SoundWire mockup device support To test the host controller and bus management code, it is currently required to have a physical SoundWire peripheral attached to the bus. To help with pre-silicon or early hardware enablement, it would be very useful to have a SoundWire 'mockup' device that is exposed in platform firmware but does not drive any signal on the bus. This is different to the existing ASoC 'dummy' codec uses for I2S/TDM, the SoundWire spec makes it clear that a device that is not attached to the bus is not permitted to interact with the bus, be it for command/control or data. This patch exposes a 'mockup' device, with a minimalist driver, with 4 partID values reserved by Intel for such test configurations. The mockup device exposes one full-duplex DAI based on 2 ports (DP1 for playback and DP8 for capture). The capture data port is just virtual, such a mockup device is prevented by the SoundWire specification from presenting any data generated by a Source port without being Attached. All the callbacks exposed by the SoundWire Slave interface are populated, even if they just return immediately. This is intentional to describe what a minimal codec driver should do and implement and help new codec vendors provide support for their devices. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Link: https://lore.kernel.org/r/20210714032209.11284-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
2021-07-14 11:22:00 +08:00
// SPDX-License-Identifier: GPL-2.0-only
//
// sdw-mockup.c -- a mockup SoundWire codec for tests where only the host
// drives the bus.
//
// Copyright(c) 2021 Intel Corporation
//
//
#include <linux/device.h>
#include <linux/mod_devicetable.h>
#include <linux/module.h>
#include <linux/soundwire/sdw.h>
#include <linux/soundwire/sdw_type.h>
#include <linux/soundwire/sdw_registers.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
struct sdw_mockup_priv {
struct sdw_slave *slave;
};
struct sdw_stream_data {
struct sdw_stream_runtime *sdw_stream;
};
static int sdw_mockup_component_probe(struct snd_soc_component *component)
{
return 0;
}
static void sdw_mockup_component_remove(struct snd_soc_component *component)
{
}
static const struct snd_soc_component_driver snd_soc_sdw_mockup_component = {
.probe = sdw_mockup_component_probe,
.remove = sdw_mockup_component_remove,
};
static int sdw_mockup_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
int direction)
{
struct sdw_stream_data *stream;
if (!sdw_stream)
return 0;
stream = kzalloc(sizeof(*stream), GFP_KERNEL);
if (!stream)
return -ENOMEM;
stream->sdw_stream = sdw_stream;
/* Use tx_mask or rx_mask to configure stream tag and set dma_data */
if (direction == SNDRV_PCM_STREAM_PLAYBACK)
dai->playback_dma_data = stream;
else
dai->capture_dma_data = stream;
return 0;
}
static void sdw_mockup_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct sdw_stream_data *stream;
stream = snd_soc_dai_get_dma_data(dai, substream);
snd_soc_dai_set_dma_data(dai, substream, NULL);
kfree(stream);
}
static int sdw_mockup_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct sdw_mockup_priv *sdw_mockup = snd_soc_component_get_drvdata(component);
struct sdw_stream_config stream_config;
struct sdw_port_config port_config;
enum sdw_data_direction direction;
struct sdw_stream_data *stream;
int num_channels;
int port;
int ret;
stream = snd_soc_dai_get_dma_data(dai, substream);
if (!stream)
return -EINVAL;
if (!sdw_mockup->slave)
return -EINVAL;
/* SoundWire specific configuration */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
direction = SDW_DATA_DIR_RX;
port = 1;
} else {
direction = SDW_DATA_DIR_TX;
port = 8;
}
stream_config.frame_rate = params_rate(params);
stream_config.ch_count = params_channels(params);
stream_config.bps = snd_pcm_format_width(params_format(params));
stream_config.direction = direction;
num_channels = params_channels(params);
port_config.ch_mask = (1 << num_channels) - 1;
port_config.num = port;
ret = sdw_stream_add_slave(sdw_mockup->slave, &stream_config,
&port_config, 1, stream->sdw_stream);
if (ret)
dev_err(dai->dev, "Unable to configure port\n");
return ret;
}
static int sdw_mockup_pcm_hw_free(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct sdw_mockup_priv *sdw_mockup = snd_soc_component_get_drvdata(component);
struct sdw_stream_data *stream =
snd_soc_dai_get_dma_data(dai, substream);
if (!sdw_mockup->slave)
return -EINVAL;
sdw_stream_remove_slave(sdw_mockup->slave, stream->sdw_stream);
return 0;
}
static const struct snd_soc_dai_ops sdw_mockup_ops = {
.hw_params = sdw_mockup_pcm_hw_params,
.hw_free = sdw_mockup_pcm_hw_free,
ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire The HDAudio ASoC support relies on the set_tdm_slots() helper to store the HDaudio stream tag in the tx_mask. This only works because of the pre-existing order in soc-pcm.c, where the hw_params() is handled for codec_dais *before* cpu_dais. When the order is reversed, the stream_tag is used as a mask in the codec fixup functions: /* fixup params based on TDM slot masks */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && codec_dai->tx_mask) soc_pcm_codec_params_fixup(&codec_params, codec_dai->tx_mask); As a result of this confusion, the codec_params_fixup() ends-up generating bad channel masks, depending on what stream_tag was allocated. We could add a flag to state that the tx_mask is really not a mask, but it would be quite ugly to persist in overloading concepts. Instead, this patch suggests a more generic get/set 'stream' API based on the existing model for SoundWire. We can expand the concept to store 'stream' opaque information that is specific to different DAI types. In the case of HDAudio DAIs, we only need to store a stream tag as an unsigned char pointer. The TDM rx_ and tx_masks should really only be used to store masks. Rename get_sdw_stream/set_sdw_stream callbacks and helpers as get_stream/set_stream. No functionality change beyond the rename. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Acked-By: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20211224021034.26635-5-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
2021-12-24 10:10:31 +08:00
.set_stream = sdw_mockup_set_sdw_stream,
ASoC: codecs: add SoundWire mockup device support To test the host controller and bus management code, it is currently required to have a physical SoundWire peripheral attached to the bus. To help with pre-silicon or early hardware enablement, it would be very useful to have a SoundWire 'mockup' device that is exposed in platform firmware but does not drive any signal on the bus. This is different to the existing ASoC 'dummy' codec uses for I2S/TDM, the SoundWire spec makes it clear that a device that is not attached to the bus is not permitted to interact with the bus, be it for command/control or data. This patch exposes a 'mockup' device, with a minimalist driver, with 4 partID values reserved by Intel for such test configurations. The mockup device exposes one full-duplex DAI based on 2 ports (DP1 for playback and DP8 for capture). The capture data port is just virtual, such a mockup device is prevented by the SoundWire specification from presenting any data generated by a Source port without being Attached. All the callbacks exposed by the SoundWire Slave interface are populated, even if they just return immediately. This is intentional to describe what a minimal codec driver should do and implement and help new codec vendors provide support for their devices. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Link: https://lore.kernel.org/r/20210714032209.11284-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
2021-07-14 11:22:00 +08:00
.shutdown = sdw_mockup_shutdown,
};
static struct snd_soc_dai_driver sdw_mockup_dai[] = {
{
.name = "sdw-mockup-aif1",
.id = 1,
.playback = {
.stream_name = "DP1 Playback",
.channels_min = 1,
.channels_max = 2,
},
.capture = {
.stream_name = "DP8 Capture",
.channels_min = 1,
.channels_max = 2,
},
.ops = &sdw_mockup_ops,
},
};
static int sdw_mockup_update_status(struct sdw_slave *slave,
enum sdw_slave_status status)
{
return 0;
}
static int sdw_mockup_read_prop(struct sdw_slave *slave)
{
struct sdw_slave_prop *prop = &slave->prop;
int nval;
int i, j;
u32 bit;
unsigned long addr;
struct sdw_dpn_prop *dpn;
prop->paging_support = false;
/*
* first we need to allocate memory for set bits in port lists
* the port allocation is completely arbitrary:
* DP0 is not supported
* DP1 is sink
* DP8 is source
*/
prop->source_ports = BIT(8);
prop->sink_ports = BIT(1);
nval = hweight32(prop->source_ports);
prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval,
sizeof(*prop->src_dpn_prop),
GFP_KERNEL);
if (!prop->src_dpn_prop)
return -ENOMEM;
i = 0;
dpn = prop->src_dpn_prop;
addr = prop->source_ports;
for_each_set_bit(bit, &addr, 32) {
dpn[i].num = bit;
dpn[i].type = SDW_DPN_FULL;
dpn[i].simple_ch_prep_sm = true;
i++;
}
/* do this again for sink now */
nval = hweight32(prop->sink_ports);
prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval,
sizeof(*prop->sink_dpn_prop),
GFP_KERNEL);
if (!prop->sink_dpn_prop)
return -ENOMEM;
j = 0;
dpn = prop->sink_dpn_prop;
addr = prop->sink_ports;
for_each_set_bit(bit, &addr, 32) {
dpn[j].num = bit;
dpn[j].type = SDW_DPN_FULL;
dpn[j].simple_ch_prep_sm = true;
j++;
}
prop->simple_clk_stop_capable = true;
/* wake-up event */
prop->wake_capable = 0;
return 0;
}
static int sdw_mockup_bus_config(struct sdw_slave *slave,
struct sdw_bus_params *params)
{
return 0;
}
static int sdw_mockup_interrupt_callback(struct sdw_slave *slave,
struct sdw_slave_intr_status *status)
{
return 0;
}
static const struct sdw_slave_ops sdw_mockup_slave_ops = {
.read_prop = sdw_mockup_read_prop,
.interrupt_callback = sdw_mockup_interrupt_callback,
.update_status = sdw_mockup_update_status,
.bus_config = sdw_mockup_bus_config,
};
static int sdw_mockup_sdw_probe(struct sdw_slave *slave,
const struct sdw_device_id *id)
{
struct device *dev = &slave->dev;
struct sdw_mockup_priv *sdw_mockup;
int ret;
sdw_mockup = devm_kzalloc(dev, sizeof(*sdw_mockup), GFP_KERNEL);
if (!sdw_mockup)
return -ENOMEM;
dev_set_drvdata(dev, sdw_mockup);
sdw_mockup->slave = slave;
slave->is_mockup_device = true;
ASoC: codecs: add SoundWire mockup device support To test the host controller and bus management code, it is currently required to have a physical SoundWire peripheral attached to the bus. To help with pre-silicon or early hardware enablement, it would be very useful to have a SoundWire 'mockup' device that is exposed in platform firmware but does not drive any signal on the bus. This is different to the existing ASoC 'dummy' codec uses for I2S/TDM, the SoundWire spec makes it clear that a device that is not attached to the bus is not permitted to interact with the bus, be it for command/control or data. This patch exposes a 'mockup' device, with a minimalist driver, with 4 partID values reserved by Intel for such test configurations. The mockup device exposes one full-duplex DAI based on 2 ports (DP1 for playback and DP8 for capture). The capture data port is just virtual, such a mockup device is prevented by the SoundWire specification from presenting any data generated by a Source port without being Attached. All the callbacks exposed by the SoundWire Slave interface are populated, even if they just return immediately. This is intentional to describe what a minimal codec driver should do and implement and help new codec vendors provide support for their devices. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Link: https://lore.kernel.org/r/20210714032209.11284-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
2021-07-14 11:22:00 +08:00
ret = devm_snd_soc_register_component(dev,
&snd_soc_sdw_mockup_component,
sdw_mockup_dai,
ARRAY_SIZE(sdw_mockup_dai));
return ret;
}
static int sdw_mockup_sdw_remove(struct sdw_slave *slave)
{
return 0;
}
/*
* Intel reserved parts ID with the following mapping expected:
* 0xAAAA: generic full-duplex codec
* 0xAA55: headset codec (mock-up of RT711/RT5682) - full-duplex
* 0x55AA: amplifier (mock-up of RT1308/Maxim 98373) - playback only with
* IV feedback
* 0x5555: mic codec (mock-up of RT715) - capture-only
*/
static const struct sdw_device_id sdw_mockup_id[] = {
SDW_SLAVE_ENTRY_EXT(0x0105, 0xAAAA, 0x0, 0, 0),
SDW_SLAVE_ENTRY_EXT(0x0105, 0xAA55, 0x0, 0, 0),
SDW_SLAVE_ENTRY_EXT(0x0105, 0x55AA, 0x0, 0, 0),
SDW_SLAVE_ENTRY_EXT(0x0105, 0x5555, 0x0, 0, 0),
{},
};
MODULE_DEVICE_TABLE(sdw, sdw_mockup_id);
static struct sdw_driver sdw_mockup_sdw_driver = {
.driver = {
.name = "sdw-mockup",
.owner = THIS_MODULE,
},
.probe = sdw_mockup_sdw_probe,
.remove = sdw_mockup_sdw_remove,
.ops = &sdw_mockup_slave_ops,
.id_table = sdw_mockup_id,
};
module_sdw_driver(sdw_mockup_sdw_driver);
MODULE_DESCRIPTION("ASoC SDW mockup codec driver");
MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>");
MODULE_LICENSE("GPL");