Received: (majordomo@vger.kernel.org) by vger.kernel.org via listexpand id S965460AbeAMImx (ORCPT + 1 other); Sat, 13 Jan 2018 03:42:53 -0500 Received: from smtp.codeaurora.org ([198.145.29.96]:58388 "EHLO smtp.codeaurora.org" rhost-flags-OK-OK-OK-OK) by vger.kernel.org with ESMTP id S1754647AbeAMImv (ORCPT ); Sat, 13 Jan 2018 03:42:51 -0500 DMARC-Filter: OpenDMARC Filter v1.3.2 smtp.codeaurora.org 66F1F607A4 Authentication-Results: pdx-caf-mail.web.codeaurora.org; dmarc=none (p=none dis=none) header.from=codeaurora.org Authentication-Results: pdx-caf-mail.web.codeaurora.org; spf=none smtp.mailfrom=rohitkr@codeaurora.org Subject: Re: [alsa-devel] [RESEND PATCH v2 08/15] ASoC: qcom: q6asm: add support to audio stream apis To: srinivas.kandagatla@linaro.org, Andy Gross , Mark Brown , linux-arm-msm@vger.kernel.org, alsa-devel@alsa-project.org Cc: Mark Rutland , devicetree@vger.kernel.org, Banajit Goswami , linux-kernel@vger.kernel.org, Patrick Lai , Takashi Iwai , sboyd@codeaurora.org, Liam Girdwood , David Brown , Rob Herring , linux-soc@vger.kernel.org, linux-arm-kernel@lists.infradead.org References: <20171214173402.19074-1-srinivas.kandagatla@linaro.org> <20171214173402.19074-9-srinivas.kandagatla@linaro.org> From: Rohit Kumar Message-ID: <8a895c84-cf2f-b192-ab66-14ea624edcdb@codeaurora.org> Date: Sat, 13 Jan 2018 14:12:33 +0530 User-Agent: Mozilla/5.0 (Windows NT 6.1; WOW64; rv:52.0) Gecko/20100101 Thunderbird/52.5.2 MIME-Version: 1.0 In-Reply-To: <20171214173402.19074-9-srinivas.kandagatla@linaro.org> Content-Type: text/plain; charset=utf-8; format=flowed Content-Transfer-Encoding: 7bit Content-Language: en-US Sender: linux-kernel-owner@vger.kernel.org List-ID: X-Mailing-List: linux-kernel@vger.kernel.org Return-Path: On 12/14/2017 11:03 PM, srinivas.kandagatla@linaro.org wrote: > From: Srinivas Kandagatla > > This patch adds support to open, write and media format commands > in the q6asm module. [..] > +static int32_t q6asm_callback(struct apr_device *adev, > + 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; > + ac could get freed by q6asm_audio_client_free during the execution of q6asm_callback as they are running in different thread. Add synchronization. > + payload = data->payload; > + > + if (data->opcode == APR_BASIC_RSP_RESULT) { > + token = data->token; > + switch (payload[0]) { > + 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; > +} > + [..] > +/** > + * 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, > + uint16_t bits_per_sample) > +{ > + struct asm_multi_channel_pcm_fmt_blk_v2 fmt; asm_multi_channel_pcm_fmt_blk_v4 is now being used in latest adsp. Better to add adsp version based support to handle different struct > + u8 *channel_mapping; > + int rc = 0; > + > + 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) { > + 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) > +{ > + 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) { > + 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]; > + > + 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); > + else > + write.flags = (0x80000000 | flags); > + > + 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; > + 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; > + > + 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; > +} > + > +/** > + * q6asm_cmd() - run cmd on audio client > + * > + * @ac: audio client pointer > + * @cmd: command to run on audio client. > + * > + * Return: Will be an negative value on error or zero on success > + */ > +int q6asm_cmd(struct audio_client *ac, int cmd) > +{ > + return __q6asm_cmd(ac, cmd, true); > +} > +EXPORT_SYMBOL_GPL(q6asm_cmd); > + > +/** > + * q6asm_cmd_nowait() - non blocking, run cmd on audio client > + * > + * @ac: audio client pointer > + * @cmd: command to run on audio client. > + * > + * Return: Will be an negative value on error or zero on success > + */ > +int q6asm_cmd_nowait(struct audio_client *ac, int cmd) > +{ > + return __q6asm_cmd(ac, cmd, false); > +} > +EXPORT_SYMBOL_GPL(q6asm_cmd_nowait); > > static int q6asm_probe(struct apr_device *adev) > { > 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 > +#define ASM_CLIENT_EVENT_CMD_PAUSE_DONE 0x1001 > +#define CMD_FLUSH 0x0002 > +#define ASM_CLIENT_EVENT_CMD_FLUSH_DONE 0x1002 > +#define CMD_EOS 0x0003 > +#define ASM_CLIENT_EVENT_CMD_EOS_DONE 0x1003 > +#define CMD_CLOSE 0x0004 > +#define ASM_CLIENT_EVENT_CMD_CLOSE_DONE 0x1004 > +#define CMD_OUT_FLUSH 0x0005 > +#define ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE 0x1005 > +#define CMD_SUSPEND 0x0006 > +#define ASM_CLIENT_EVENT_CMD_SUSPEND_DONE 0x1006 > +#define ASM_CLIENT_EVENT_CMD_RUN_DONE 0x1008 > +#define ASM_CLIENT_EVENT_DATA_WRITE_DONE 0x1009 > + > +#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 > > typedef void (*app_cb) (uint32_t opcode, uint32_t token, > uint32_t *payload, void *priv); > @@ -10,6 +37,21 @@ struct audio_client; > struct audio_client *q6asm_audio_client_alloc(struct device *dev, > app_cb cb, void *priv); > void q6asm_audio_client_free(struct audio_client *ac); > +int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts, > + uint32_t lsw_ts, uint32_t flags); > +int q6asm_open_write(struct audio_client *ac, uint32_t format, > + uint16_t bits_per_sample); > +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, > + uint16_t bits_per_sample); > +int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, > + uint32_t lsw_ts); > +int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, > + uint32_t lsw_ts); > +int q6asm_cmd(struct audio_client *ac, int cmd); > +int q6asm_cmd_nowait(struct audio_client *ac, int cmd); > int q6asm_get_session_id(struct audio_client *ac); > int q6asm_map_memory_regions(unsigned int dir, > struct audio_client *ac,