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Message-ID: <20180102200813.GA8625@builder>
Date: Tue, 2 Jan 2018 12:08:13 -0800
From: Bjorn Andersson <bjorn.andersson@...aro.org>
To: srinivas.kandagatla@...aro.org
Cc: Andy Gross <andy.gross@...aro.org>,
Mark Brown <broonie@...nel.org>, linux-arm-msm@...r.kernel.org,
alsa-devel@...a-project.org, David Brown <david.brown@...aro.org>,
Rob Herring <robh+dt@...nel.org>,
Mark Rutland <mark.rutland@....com>,
Liam Girdwood <lgirdwood@...il.com>,
Patrick Lai <plai@...eaurora.org>,
Banajit Goswami <bgoswami@...eaurora.org>,
Jaroslav Kysela <perex@...ex.cz>,
Takashi Iwai <tiwai@...e.com>, linux-soc@...r.kernel.org,
devicetree@...r.kernel.org, linux-kernel@...r.kernel.org,
linux-arm-kernel@...ts.infradead.org, sboyd@...eaurora.org
Subject: Re: [RESEND PATCH v2 08/15] ASoC: qcom: q6asm: add support to audio
stream apis
On Thu 14 Dec 09:33 PST 2017, srinivas.kandagatla@...aro.org wrote:
> From: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
>
> This patch adds support to open, write and media format commands
> in the q6asm module.
>
> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
> ---
> sound/soc/qcom/qdsp6/q6asm.c | 530 ++++++++++++++++++++++++++++++++++++++++++-
> sound/soc/qcom/qdsp6/q6asm.h | 42 ++++
> 2 files changed, 571 insertions(+), 1 deletion(-)
>
> diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
> index 4be92441f524..dabd6509ef99 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.c
> +++ b/sound/soc/qcom/qdsp6/q6asm.c
> @@ -8,16 +8,34 @@
> #include <linux/soc/qcom/apr.h>
> #include <linux/device.h>
> #include <linux/platform_device.h>
> +#include <uapi/sound/asound.h>
> #include <linux/delay.h>
> #include <linux/slab.h>
> #include <linux/mm.h>
> #include "q6asm.h"
> #include "common.h"
>
> +#define ASM_STREAM_CMD_CLOSE 0x00010BCD
> +#define ASM_STREAM_CMD_FLUSH 0x00010BCE
> +#define ASM_SESSION_CMD_PAUSE 0x00010BD3
> +#define ASM_DATA_CMD_EOS 0x00010BDB
> +#define DEFAULT_POPP_TOPOLOGY 0x00010BE4
> +#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
> #define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92
> #define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93
> #define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94
> -
> +#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98
> +#define ASM_DATA_EVENT_WRITE_DONE_V2 0x00010D99
> +#define ASM_SESSION_CMD_RUN_V2 0x00010DAA
> +#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
> +#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
> +#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
> +#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3
> +
> +#define ASM_LEGACY_STREAM_SESSION 0
> +#define ASM_END_POINT_DEVICE_MATRIX 0
> +#define DEFAULT_APP_TYPE 0
> +#define TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */
> #define TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
> #define SYNC_IO_MODE 0x0001
> #define ASYNC_IO_MODE 0x0002
Probably prettier to reorder these and make them Q6ASM_IO_MODE_xyz
[..]
>
> +static int32_t q6asm_callback(struct apr_device *adev,
This callback is an extracted part of q6asm_srvc_callback(), can it be
given a more descriptive name?
> + struct apr_client_data *data, int session_id)
> +{
> + struct audio_client *ac;// = (struct audio_client *)priv;
> + uint32_t token;
> + uint32_t *payload;
> + uint32_t wakeup_flag = 1;
> + uint32_t client_event = 0;
> + struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
> +
> + if (data == NULL)
> + return -EINVAL;
> +
> + ac = q6asm_get_audio_client(q6asm, session_id);
> + if (!q6asm_is_valid_audio_client(ac))
> + return -EINVAL;
> +
> + payload = data->payload;
> +
> + if (data->opcode == APR_BASIC_RSP_RESULT) {
Move this into the switch.
> + token = data->token;
> + switch (payload[0]) {
This is again that common response struct.
> + case ASM_SESSION_CMD_PAUSE:
> + client_event = ASM_CLIENT_EVENT_CMD_PAUSE_DONE;
> + break;
> + case ASM_SESSION_CMD_SUSPEND:
> + client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
> + break;
> + case ASM_DATA_CMD_EOS:
> + client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
> + break;
> + break;
> + case ASM_STREAM_CMD_FLUSH:
> + client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
> + break;
> + case ASM_SESSION_CMD_RUN_V2:
> + client_event = ASM_CLIENT_EVENT_CMD_RUN_DONE;
> + break;
> +
> + case ASM_STREAM_CMD_FLUSH_READBUFS:
> + if (token != ac->session) {
> + dev_err(ac->dev, "session invalid\n");
> + return -EINVAL;
> + }
> + case ASM_STREAM_CMD_CLOSE:
> + client_event = ASM_CLIENT_EVENT_CMD_CLOSE_DONE;
> + break;
> + case ASM_STREAM_CMD_OPEN_WRITE_V3:
> + case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
> + if (payload[1] != 0) {
> + dev_err(ac->dev,
> + "cmd = 0x%x returned error = 0x%x\n",
> + payload[0], payload[1]);
> + if (wakeup_flag) {
> + ac->cmd_state = payload[1];
> + wake_up(&ac->cmd_wait);
> + }
> + return 0;
> + }
> + break;
> + default:
> + dev_err(ac->dev, "command[0x%x] not expecting rsp\n",
> + payload[0]);
> + break;
> + }
> +
> + if (ac->cmd_state && wakeup_flag) {
> + ac->cmd_state = 0;
> + wake_up(&ac->cmd_wait);
> + }
> + if (ac->cb)
> + ac->cb(client_event, data->token,
> + data->payload, ac->priv);
> +
> + return 0;
> + }
> +
> + switch (data->opcode) {
> + case ASM_DATA_EVENT_WRITE_DONE_V2:{
> + struct audio_port_data *port =
> + &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
> +
> + client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE;
> +
> + if (ac->io_mode & SYNC_IO_MODE) {
> + dma_addr_t phys = port->buf[data->token].phys;
> +
> + if (lower_32_bits(phys) != payload[0] ||
> + upper_32_bits(phys) != payload[1]) {
> + dev_err(ac->dev, "Expected addr %pa\n",
> + &port->buf[data->token].phys);
> + return -EINVAL;
> + }
> + token = data->token;
> + port->buf[token].used = 1;
> + }
> + break;
> + }
> + }
> + if (ac->cb)
> + ac->cb(client_event, data->token, data->payload, ac->priv);
> +
> + return 0;
> +}
> +
> static int q6asm_srvc_callback(struct apr_device *adev, struct apr_client_data *data)
> {
> struct q6asm *a, *q6asm = dev_get_drvdata(&adev->dev);
> @@ -415,12 +581,16 @@ static int q6asm_srvc_callback(struct apr_device *adev, struct apr_client_data *
> struct audio_port_data *port;
> uint32_t dir = 0;
> uint32_t sid = 0;
> + int dest_port;
> uint32_t *payload;
>
> if (!data) {
> dev_err(&adev->dev, "%s: Invalid CB\n", __func__);
> return 0;
> }
> + dest_port = (data->dest_port >> 8) & 0xFF;
> + if (dest_port)
> + return q6asm_callback(adev, data, dest_port);
You call dest_port "session_id" above, this seems to be a better name
for this variable.
>
> payload = data->payload;
> sid = (data->token >> 8) & 0x0F;
> @@ -540,6 +710,364 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev,
> }
> EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc);
>
> +static int __q6asm_open_write(struct audio_client *ac, uint32_t format,
> + uint16_t bits_per_sample, uint32_t stream_id,
> + bool is_gapless_mode)
> +{
> + struct asm_stream_cmd_open_write_v3 open;
> + int rc;
> +
> + q6asm_add_hdr(ac, &open.hdr, sizeof(open), true, stream_id);
> + ac->cmd_state = -1;
> +
> + open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
> + open.mode_flags = 0x00;
> + open.mode_flags |= ASM_LEGACY_STREAM_SESSION;
> + if (is_gapless_mode)
This is hard coded as false.
> + open.mode_flags |= 1 << ASM_SHIFT_GAPLESS_MODE_FLAG;
> +
> + /* source endpoint : matrix */
> + open.sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
> + open.bits_per_sample = bits_per_sample;
> + open.postprocopo_id = DEFAULT_POPP_TOPOLOGY;
> +
> + switch (format) {
> + case FORMAT_LINEAR_PCM:
> + open.dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
> + break;
> + default:
> + dev_err(ac->dev, "Invalid format 0x%x\n", format);
> + return -EINVAL;
> + }
> + rc = apr_send_pkt(ac->adev, (uint32_t *) &open);
> + if (rc < 0)
> + return rc;
> +
> + rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state >= 0), 5 * HZ);
> + if (!rc) {
> + dev_err(ac->dev, "timeout on open write\n");
> + return -ETIMEDOUT;
> + }
Almost every time you apr_send_pkt() you have this wait with timeout,
can this send/wait/return be wrapped in a helper function to reduce the
duplication?
Creating a q6asm_send_sync() and q6asm_send_async() pair with this logic
should help quite a bit.
> +
> + if (ac->cmd_state > 0)
> + return adsp_err_get_lnx_err_code(ac->cmd_state);
> +
> + ac->io_mode |= TUN_WRITE_IO_MODE;
> +
> + return 0;
> +}
> +
> +/**
> + * q6asm_open_write() - Open audio client for writing
> + *
> + * @ac: audio client pointer
> + * @format: audio sample format
> + * @bits_per_sample: bits per sample
> + *
> + * Return: Will be an negative value on error or zero on success
> + */
> +int q6asm_open_write(struct audio_client *ac, uint32_t format,
> + uint16_t bits_per_sample)
> +{
> + return __q6asm_open_write(ac, format, bits_per_sample,
I don't see a particular reason for not inlining this, is there one
coming later in the series?
> + ac->stream_id, false);
> +}
> +EXPORT_SYMBOL_GPL(q6asm_open_write);
> +
> +static int __q6asm_run(struct audio_client *ac, uint32_t flags,
> + uint32_t msw_ts, uint32_t lsw_ts, bool wait)
> +{
> + struct asm_session_cmd_run_v2 run;
> + int rc;
> +
> + q6asm_add_hdr(ac, &run.hdr, sizeof(run), true, ac->stream_id);
> + ac->cmd_state = -1;
> +
> + run.hdr.opcode = ASM_SESSION_CMD_RUN_V2;
> + run.flags = flags;
> + run.time_lsw = lsw_ts;
> + run.time_msw = msw_ts;
> +
> + rc = apr_send_pkt(ac->adev, (uint32_t *) &run);
> + if (rc < 0)
> + return rc;
> +
> + if (wait) {
Rather than having half of the function conditional I would recommend
inlining this function in the two callers.
In particular if you can come up with a helper function for the
send/wait/handle-error case.
> + rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state >= 0),
> + 5 * HZ);
> + if (!rc) {
> + dev_err(ac->dev, "timeout on run cmd\n");
> + return -ETIMEDOUT;
> + }
> + if (ac->cmd_state > 0)
> + return adsp_err_get_lnx_err_code(ac->cmd_state);
> + }
> +
> + return 0;
> +}
> +
> +/**
> + * q6asm_run() - start the audio client
> + *
> + * @ac: audio client pointer
> + * @flags: flags associated with write
> + * @msw_ts: timestamp msw
> + * @lsw_ts: timestamp lsw
> + *
> + * Return: Will be an negative value on error or zero on success
> + */
> +int q6asm_run(struct audio_client *ac, uint32_t flags,
> + uint32_t msw_ts, uint32_t lsw_ts)
> +{
> + return __q6asm_run(ac, flags, msw_ts, lsw_ts, true);
> +}
> +EXPORT_SYMBOL_GPL(q6asm_run);
> +
> +/**
> + * q6asm_run_nowait() - start the audio client withou blocking
> + *
> + * @ac: audio client pointer
> + * @flags: flags associated with write
> + * @msw_ts: timestamp msw
> + * @lsw_ts: timestamp lsw
> + *
> + * Return: Will be an negative value on error or zero on success
> + */
> +int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
> + uint32_t msw_ts, uint32_t lsw_ts)
> +{
> + return __q6asm_run(ac, flags, msw_ts, lsw_ts, false);
> +}
> +EXPORT_SYMBOL_GPL(q6asm_run_nowait);
> +
> +/**
> + * q6asm_media_format_block_multi_ch_pcm() - setup pcm configuration
> + *
> + * @ac: audio client pointer
> + * @rate: audio sample rate
> + * @channels: number of audio channels.
> + * @use_default_chmap: flag to use default ch map.
> + * @channel_map: channel map pointer
> + * @bits_per_sample: bits per sample
> + *
> + * Return: Will be an negative value on error or zero on success
> + */
> +int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
> + uint32_t rate, uint32_t channels,
> + bool use_default_chmap,
> + char *channel_map,
This should be u8 channel_map[PCM_FORMAT_MAX_NUM_CHANNEL], possibly
char. Unless you, as I suggest below, want to be able to represent
use_default_chmap = false, by setting this to NULL.
> + uint16_t bits_per_sample)
> +{
> + struct asm_multi_channel_pcm_fmt_blk_v2 fmt;
> + u8 *channel_mapping;
> + int rc = 0;
Unnecessary initialization.
> +
> + q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), true, ac->stream_id);
> + ac->cmd_state = -1;
> +
> + fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
> + fmt.fmt_blk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
> + sizeof(fmt.fmt_blk);
> + fmt.num_channels = channels;
> + fmt.bits_per_sample = bits_per_sample;
> + fmt.sample_rate = rate;
> + fmt.is_signed = 1;
> +
> + channel_mapping = fmt.channel_mapping;
> +
> + if (use_default_chmap) {
Passing NULL as channel_map would probably be a nicer way to say this,
instead of having a separate bool.
> + if (q6dsp_map_channels(channel_mapping, channels)) {
> + dev_err(ac->dev, " map channels failed %d\n", channels);
> + return -EINVAL;
> + }
> + } else {
> + memcpy(channel_mapping, channel_map,
> + PCM_FORMAT_MAX_NUM_CHANNEL);
> + }
> +
> + rc = apr_send_pkt(ac->adev, (uint32_t *) &fmt);
> + if (rc < 0)
> + goto fail_cmd;
> +
> + rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state >= 0), 5 * HZ);
> + if (!rc) {
> + dev_err(ac->dev, "timeout on format update\n");
> + return -ETIMEDOUT;
> + }
> + if (ac->cmd_state > 0)
> + return adsp_err_get_lnx_err_code(ac->cmd_state);
> +
> + return 0;
> +fail_cmd:
> + return rc;
> +}
> +EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
> +
> +/**
> + * q6asm_write_nolock() - non blocking write
> + *
> + * @ac: audio client pointer
> + * @len: lenght in bytes
> + * @msw_ts: timestamp msw
> + * @lsw_ts: timestamp lsw
> + * @flags: flags associated with write
> + *
> + * Return: Will be an negative value on error or zero on success
> + */
> +int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
> + uint32_t lsw_ts, uint32_t flags)
q6asm_write_async() is probably a better name, nolock indicates some
relationship to mutual exclusions...
> +{
> + struct asm_data_cmd_write_v2 write;
> + struct audio_port_data *port;
> + struct audio_buffer *ab;
> + int dsp_buf = 0;
> + int rc = 0;
> +
> + if (ac->io_mode & SYNC_IO_MODE) {
Bail early if this isn't true, to save you the indentation level.
> + port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
> + q6asm_add_hdr(ac, &write.hdr, sizeof(write), false,
> + ac->stream_id);
> +
> + dsp_buf = port->dsp_buf;
> + ab = &port->buf[dsp_buf];
So we're just unconditionally telling the remote side about the next buf
in our ring buffer. Do we need to ensure that this is available/ready?
> +
> + write.hdr.token = port->dsp_buf;
> + write.hdr.opcode = ASM_DATA_CMD_WRITE_V2;
> + write.buf_addr_lsw = lower_32_bits(ab->phys);
> + write.buf_addr_msw = upper_32_bits(ab->phys);
> + write.buf_size = len;
> + write.seq_id = port->dsp_buf;
> + write.timestamp_lsw = lsw_ts;
> + write.timestamp_msw = msw_ts;
> + write.mem_map_handle =
> + ac->port[SNDRV_PCM_STREAM_PLAYBACK].mem_map_handle;
> +
> + if (flags == NO_TIMESTAMP)
> + write.flags = (flags & 0x800000FF);
Fill in the constant and this becomes
if flags == 0xff00:
write.flags = 0xff00 & 0x800000ff;
Or in other words:
if flags == 0xff00:
write.flags = 0;
> + else
> + write.flags = (0x80000000 | flags);
Drop the parenthesis and flip the |. It would be nice to have a define
or a comment indicating what BIT(31) is...
> +
> + port->dsp_buf++;
> +
> + if (port->dsp_buf >= port->max_buf_cnt)
> + port->dsp_buf = 0;
> +
> + rc = apr_send_pkt(ac->adev, (uint32_t *) &write);
> + if (rc < 0)
> + return rc;
> + }
> +
> + return 0;
> +}
> +EXPORT_SYMBOL_GPL(q6asm_write_nolock);
> +
> +static void q6asm_reset_buf_state(struct audio_client *ac)
> +{
> + int cnt = 0;
> + int loopcnt = 0;
> + int used;
> + struct audio_port_data *port = NULL;
> +
> + if (ac->io_mode & SYNC_IO_MODE) {
> + used = (ac->io_mode & TUN_WRITE_IO_MODE ? 1 : 0);
> + mutex_lock(&ac->cmd_lock);
> + for (loopcnt = 0; loopcnt <= SNDRV_PCM_STREAM_CAPTURE;
> + loopcnt++) {
> + port = &ac->port[loopcnt];
> + cnt = port->max_buf_cnt - 1;
> + port->dsp_buf = 0;
> + while (cnt >= 0) {
> + if (!port->buf)
> + continue;
> + port->buf[cnt].used = used;
> + cnt--;
> + }
> + }
> + mutex_unlock(&ac->cmd_lock);
> + }
> +}
> +
> +static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
> +{
> + int stream_id = ac->stream_id;
> + struct apr_hdr hdr;
> + int rc;
> +
> + q6asm_add_hdr(ac, &hdr, sizeof(hdr), true, stream_id);
> + ac->cmd_state = -1;
Resetting cmd_state relates to the send, don't mix it with building the
packet.
> + switch (cmd) {
> + case CMD_PAUSE:
> + hdr.opcode = ASM_SESSION_CMD_PAUSE;
> + break;
> + case CMD_SUSPEND:
> + hdr.opcode = ASM_SESSION_CMD_SUSPEND;
> + break;
> + case CMD_FLUSH:
> + hdr.opcode = ASM_STREAM_CMD_FLUSH;
> + break;
> + case CMD_OUT_FLUSH:
> + hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
> + break;
> + case CMD_EOS:
> + hdr.opcode = ASM_DATA_CMD_EOS;
> + ac->cmd_state = 0;
> + break;
> + case CMD_CLOSE:
> + hdr.opcode = ASM_STREAM_CMD_CLOSE;
> + break;
> + default:
> + return -EINVAL;
> + }
> +
> + rc = apr_send_pkt(ac->adev, (uint32_t *) &hdr);
> + if (rc < 0)
> + return rc;
> +
> + if (!wait)
> + return 0;
I've asked you to split the others into _sync() vs _async() operations.
One particular concern I have is that I don't see any mutual exclusion
protecting the cmd_state and a call with !wait will overwrite the
existing value, which might be unexpected.
> +
> + rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state >= 0), 5 * HZ);
> + if (!rc) {
> + dev_err(ac->dev, "timeout response for opcode[0x%x]\n",
> + hdr.opcode);
> + return -ETIMEDOUT;
> + }
> + if (ac->cmd_state > 0)
> + return adsp_err_get_lnx_err_code(ac->cmd_state);
> +
> + if (cmd == CMD_FLUSH)
> + q6asm_reset_buf_state(ac);
> +
> + return 0;
> +}
[..]
> diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
> index e1409c368600..b4896059da79 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.h
> +++ b/sound/soc/qcom/qdsp6/q6asm.h
> @@ -2,7 +2,34 @@
> #ifndef __Q6_ASM_H__
> #define __Q6_ASM_H__
>
> +/* ASM client callback events */
> +#define CMD_PAUSE 0x0001
These defines has rather generic names...
[..]
> +
> +#define MSM_FRONTEND_DAI_MULTIMEDIA1 0
> +#define MSM_FRONTEND_DAI_MULTIMEDIA2 1
> +#define MSM_FRONTEND_DAI_MULTIMEDIA3 2
> +#define MSM_FRONTEND_DAI_MULTIMEDIA4 3
> +#define MSM_FRONTEND_DAI_MULTIMEDIA5 4
> +#define MSM_FRONTEND_DAI_MULTIMEDIA6 5
> +#define MSM_FRONTEND_DAI_MULTIMEDIA7 6
> +#define MSM_FRONTEND_DAI_MULTIMEDIA8 7
> +
> #define MAX_SESSIONS 16
> +#define NO_TIMESTAMP 0xFF00
> +#define FORMAT_LINEAR_PCM 0x0000
Ditto.
Regards,
Bjorn
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