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[86.26.100.93]) by smtp.gmail.com with ESMTPSA id t138sm1633154wme.16.2017.12.05.14.20.26 (version=TLS1_2 cipher=ECDHE-RSA-AES128-SHA bits=128/128); Tue, 05 Dec 2017 14:20:27 -0800 (PST) From: srinivas.kandagatla@linaro.org To: broonie@kernel.org, bgoswami@codeaurora.org Cc: alsa-devel@alsa-project.org, tiwai@suse.com, plai@codeaurora.org, kwestfie@codeaurora.org, linux-kernel@vger.kernel.org, linux-arm-msm@vger.kernel.org, lkasam@qti.qualcomm.com, Srinivas Kandagatla Subject: [PATCH v2 08/15] ASoC: qcom: q6asm: add support to audio stream apis Date: Tue, 5 Dec 2017 22:18:05 +0000 Message-Id: <20171205221812.25641-9-srinivas.kandagatla@linaro.org> X-Mailer: git-send-email 2.15.0 In-Reply-To: <20171205221812.25641-1-srinivas.kandagatla@linaro.org> References: <20171205221812.25641-1-srinivas.kandagatla@linaro.org> Sender: linux-kernel-owner@vger.kernel.org Precedence: bulk List-ID: X-Mailing-List: linux-kernel@vger.kernel.org From: Srinivas Kandagatla This patch adds support to open, write and media format commands in the q6asm module. Signed-off-by: Srinivas Kandagatla --- sound/soc/qcom/qdsp6/q6asm.c | 530 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.h | 42 ++++ 2 files changed, 571 insertions(+), 1 deletion(-) -- 2.15.0 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 #include #include +#include #include #include #include #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 @@ -42,6 +60,49 @@ struct avs_cmd_shared_mem_unmap_regions { u32 mem_map_handle; } __packed; +struct asm_data_cmd_media_fmt_update_v2 { + u32 fmt_blk_size; +} __packed; + +struct asm_multi_channel_pcm_fmt_blk_v2 { + struct apr_hdr hdr; + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u16 num_channels; + u16 bits_per_sample; + u32 sample_rate; + u16 is_signed; + u16 reserved; + u8 channel_mapping[8]; +} __packed; + +struct asm_data_cmd_write_v2 { + struct apr_hdr hdr; + u32 buf_addr_lsw; + u32 buf_addr_msw; + u32 mem_map_handle; + u32 buf_size; + u32 seq_id; + u32 timestamp_lsw; + u32 timestamp_msw; + u32 flags; +} __packed; + +struct asm_stream_cmd_open_write_v3 { + struct apr_hdr hdr; + uint32_t mode_flags; + uint16_t sink_endpointype; + uint16_t bits_per_sample; + uint32_t postprocopo_id; + uint32_t dec_fmt_id; +} __packed; + +struct asm_session_cmd_run_v2 { + struct apr_hdr hdr; + u32 flags; + u32 time_lsw; + u32 time_msw; +} __packed; + struct audio_buffer { dma_addr_t phys; uint32_t used; @@ -408,6 +469,111 @@ static struct audio_client *q6asm_get_audio_client(struct q6asm *a, return NULL; } +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; + + 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; +} + 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); 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) + 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; + } + + 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, + 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) { + 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, + uint16_t bits_per_sample) +{ + struct asm_multi_channel_pcm_fmt_blk_v2 fmt; + 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,