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[86.26.100.93]) by smtp.gmail.com with ESMTPSA id x67sm2529314wma.23.2018.04.18.08.36.13 (version=TLS1_2 cipher=ECDHE-RSA-AES128-SHA bits=128/128); Wed, 18 Apr 2018 08:36:14 -0700 (PDT) From: srinivas.kandagatla@linaro.org To: andy.gross@linaro.org, broonie@kernel.org, linux-arm-msm@vger.kernel.org, alsa-devel@alsa-project.org, robh+dt@kernel.org Date: Wed, 18 Apr 2018 16:31:48 +0100 Message-Id: <20180418153157.1960-15-srinivas.kandagatla@linaro.org> X-Mailer: git-send-email 2.16.2 In-Reply-To: <20180418153157.1960-1-srinivas.kandagatla@linaro.org> References: <20180418153157.1960-1-srinivas.kandagatla@linaro.org> Cc: mark.rutland@arm.com, devicetree@vger.kernel.org, bgoswami@codeaurora.org, rohkumar@qti.qualcomm.com, gregkh@linuxfoundation.org, plai@codeaurora.org, lgirdwood@gmail.com, tiwai@suse.com, david.brown@linaro.org, Srinivas Kandagatla , linux-arm-kernel@lists.infradead.org, spatakok@qti.qualcomm.com, linux-kernel@vger.kernel.org Subject: [alsa-devel] [PATCH v5 14/23] ASoC: qdsp6: q6asm: Add support to audio stream apis X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.14 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , MIME-Version: 1.0 Errors-To: alsa-devel-bounces@alsa-project.org Sender: alsa-devel-bounces@alsa-project.org From: Srinivas Kandagatla This patch adds support to open, write and media format commands in the q6asm module. Signed-off-by: Srinivas Kandagatla Reviewed-and-tested-by: Rohit kumar --- sound/soc/qcom/qdsp6/q6asm.c | 744 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.h | 49 +++ 2 files changed, 792 insertions(+), 1 deletion(-) -- 2.16.2 _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 51a5681dd61f..7f10d5f3c58b 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -9,6 +9,8 @@ #include #include #include +#include +#include #include #include #include @@ -17,10 +19,36 @@ #include "q6dsp-errno.h" #include "q6dsp-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 ASM_NULL_POPP_TOPOLOGY 0x00010C68 +#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09 +#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10 +#define ASM_STREAM_POSTPROC_TOPO_ID_NONE 0x00010C68 #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_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3 +#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_DATA_CMD_READ_V2 0x00010DAC +#define ASM_SESSION_CMD_SUSPEND 0x00010DEC +#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3 +#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4 +#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A +#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D + + +#define ASM_LEGACY_STREAM_SESSION 0 +/* Bit shift for the stream_perf_mode subfield. */ +#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ 29 +#define ASM_END_POINT_DEVICE_MATRIX 0 +#define ASM_DEFAULT_APP_TYPE 0 #define ASM_SYNC_IO_MODE 0x0001 #define ASM_ASYNC_IO_MODE 0x0002 #define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */ @@ -46,6 +74,96 @@ 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[PCM_FORMAT_MAX_NUM_CHANNEL]; +} __packed; + +struct asm_stream_cmd_set_encdec_param { + u32 param_id; + u32 param_size; +} __packed; + +struct asm_enc_cfg_blk_param_v2 { + u32 frames_per_buf; + u32 enc_cfg_blk_size; +} __packed; + +struct asm_multi_channel_pcm_enc_cfg_v2 { + struct apr_hdr hdr; + struct asm_stream_cmd_set_encdec_param encdec; + struct asm_enc_cfg_blk_param_v2 encblk; + uint16_t num_channels; + uint16_t bits_per_sample; + uint32_t sample_rate; + uint16_t is_signed; + uint16_t reserved; + uint8_t channel_mapping[8]; +} __packed; + +struct asm_data_cmd_read_v2 { + struct apr_hdr hdr; + u32 buf_addr_lsw; + u32 buf_addr_msw; + u32 mem_map_handle; + u32 buf_size; + u32 seq_id; +} __packed; + +struct asm_data_cmd_read_v2_done { + u32 status; + u32 buf_addr_lsw; + u32 buf_addr_msw; +}; + +struct asm_stream_cmd_open_read_v3 { + struct apr_hdr hdr; + u32 mode_flags; + u32 src_endpointype; + u32 preprocopo_id; + u32 enc_cfg_id; + u16 bits_per_sample; + u16 reserved; +} __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 { phys_addr_t phys; uint32_t used; @@ -394,6 +512,153 @@ static struct audio_client *q6asm_get_audio_client(struct q6asm *a, return a->session[session_id]; } +static int32_t q6asm_stream_callback(struct apr_device *adev, + struct apr_client_message *data, + int session_id) +{ + struct q6asm *q6asm = dev_get_drvdata(&adev->dev); + struct aprv2_ibasic_rsp_result_t *result; + struct audio_port_data *port; + struct audio_client *ac; + uint32_t token; + uint32_t client_event = 0; + + ac = q6asm_get_audio_client(q6asm, session_id); + if (!ac)/* Audio client might already be freed by now */ + return 0; + + if (!q6asm_is_valid_audio_client(ac)) + return -EINVAL; + + result = data->payload; + + switch (data->opcode) { + case APR_BASIC_RSP_RESULT: + token = data->token; + switch (result->opcode) { + 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_STREAM_CMD_OPEN_READ_V3: + case ASM_STREAM_CMD_OPEN_READWRITE_V2: + case ASM_STREAM_CMD_SET_ENCDEC_PARAM: + case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2: + if (result->status != 0) { + dev_err(ac->dev, + "cmd = 0x%x returned error = 0x%x\n", + result->opcode, result->status); + ac->result = *result; + wake_up(&ac->cmd_wait); + return 0; + } + break; + default: + dev_err(ac->dev, "command[0x%x] not expecting rsp\n", + result->opcode); + break; + } + + ac->result = *result; + wake_up(&ac->cmd_wait); + + if (ac->cb) + ac->cb(client_event, data->token, + data->payload, ac->priv); + + return 0; + + case ASM_DATA_EVENT_WRITE_DONE_V2: + port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK]; + + client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE; + + if (ac->io_mode & ASM_SYNC_IO_MODE) { + phys_addr_t phys; + unsigned long flags; + + spin_lock_irqsave(&ac->buf_lock, flags); + if (!port->buf) { + spin_unlock_irqrestore(&ac->buf_lock, flags); + return 0; + } + + phys = port->buf[data->token].phys; + + if (lower_32_bits(phys) != result->opcode || + upper_32_bits(phys) != result->status) { + dev_err(ac->dev, "Expected addr %pa\n", + &port->buf[data->token].phys); + spin_unlock_irqrestore(&ac->buf_lock, flags); + return -EINVAL; + } + token = data->token; + port->buf[token].used = 1; + spin_unlock_irqrestore(&ac->buf_lock, flags); + } + break; + case ASM_DATA_EVENT_READ_DONE_V2: + port = &ac->port[SNDRV_PCM_STREAM_CAPTURE]; + client_event = ASM_CLIENT_EVENT_DATA_READ_DONE; + + if (ac->io_mode & ASM_SYNC_IO_MODE) { + struct asm_data_cmd_read_v2_done *done = data->payload; + unsigned long flags; + phys_addr_t phys; + + spin_lock_irqsave(&ac->buf_lock, flags); + if (!port->buf) { + spin_unlock_irqrestore(&ac->buf_lock, flags); + return 0; + } + + phys = port->buf[data->token].phys; + token = data->token; + port->buf[token].used = 0; + + if (upper_32_bits(phys) != done->buf_addr_msw || + lower_32_bits(phys) != done->buf_addr_lsw) { + dev_err(ac->dev, "Expected addr %pa %08x-%08x\n", + &port->buf[data->token].phys, + done->buf_addr_lsw, + done->buf_addr_msw); + spin_unlock_irqrestore(&ac->buf_lock, flags); + return -EINVAL; + } + spin_unlock_irqrestore(&ac->buf_lock, flags); + } + + 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_message *data) { @@ -403,6 +668,11 @@ static int q6asm_srvc_callback(struct apr_device *adev, struct audio_port_data *port; uint32_t dir = 0; uint32_t sid = 0; + int session_id; + + session_id = (data->dest_port >> 8) & 0xFF; + if (session_id) + return q6asm_stream_callback(adev, data, session_id); result = data->payload; sid = (data->token >> 8) & 0x0F; @@ -510,6 +780,478 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb, } EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc); +static int q6asm_ac_send_cmd_sync(struct audio_client *ac, void *cmd) +{ + struct apr_hdr *hdr = cmd; + int rc; + + mutex_lock(&ac->lock); + ac->result.opcode = 0; + ac->result.status = 0; + + rc = apr_send_pkt(ac->adev, cmd); + if (rc < 0) + goto err; + + rc = wait_event_timeout(ac->cmd_wait, + (ac->result.opcode == hdr->opcode), 5 * HZ); + if (!rc) { + dev_err(ac->dev, "CMD timeout\n"); + rc = -ETIMEDOUT; + goto err; + } + + if (ac->result.status > 0) { + dev_err(ac->dev, "DSP returned error[%x]\n", + ac->result.status); + rc = -EINVAL; + } + + +err: + mutex_unlock(&ac->lock); + return rc; +} + +/** + * 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) +{ + struct asm_stream_cmd_open_write_v3 open; + int rc; + + q6asm_add_hdr(ac, &open.hdr, sizeof(open), true, ac->stream_id); + + open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3; + open.mode_flags = 0x00; + open.mode_flags |= ASM_LEGACY_STREAM_SESSION; + + /* source endpoint : matrix */ + open.sink_endpointype = ASM_END_POINT_DEVICE_MATRIX; + open.bits_per_sample = bits_per_sample; + open.postprocopo_id = ASM_NULL_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 = q6asm_ac_send_cmd_sync(ac, &open); + if (rc < 0) + return rc; + + ac->io_mode |= ASM_TUN_WRITE_IO_MODE; + + return 0; +} +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; + + q6asm_add_hdr(ac, &run.hdr, sizeof(run), true, ac->stream_id); + + run.hdr.opcode = ASM_SESSION_CMD_RUN_V2; + run.flags = flags; + run.time_lsw = lsw_ts; + run.time_msw = msw_ts; + if (wait) + return q6asm_ac_send_cmd_sync(ac, &run); + else + return apr_send_pkt(ac->adev, &run); + +} + +/** + * 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, + u8 channel_map[PCM_FORMAT_MAX_NUM_CHANNEL], + uint16_t bits_per_sample) +{ + struct asm_multi_channel_pcm_fmt_blk_v2 fmt; + u8 *channel_mapping; + int rc; + + q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), true, ac->stream_id); + + 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 (channel_map) { + memcpy(channel_mapping, channel_map, + PCM_FORMAT_MAX_NUM_CHANNEL); + } else { + if (q6dsp_map_channels(channel_mapping, channels)) { + dev_err(ac->dev, " map channels failed %d\n", channels); + return -EINVAL; + } + } + + rc = q6asm_ac_send_cmd_sync(ac, &fmt); + if (rc < 0) + goto fail_cmd; + + return 0; +fail_cmd: + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm); + +/** + * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture + * + * @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_enc_cfg_blk_pcm_format_support(struct audio_client *ac, + uint32_t rate, uint32_t channels, uint16_t bits_per_sample) +{ + struct asm_multi_channel_pcm_enc_cfg_v2 enc_cfg; + u8 *channel_mapping; + u32 frames_per_buf = 0; + + q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), true, ac->stream_id); + enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM; + enc_cfg.encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2; + enc_cfg.encdec.param_size = sizeof(enc_cfg) - sizeof(enc_cfg.hdr) - + sizeof(enc_cfg.encdec); + enc_cfg.encblk.frames_per_buf = frames_per_buf; + enc_cfg.encblk.enc_cfg_blk_size = enc_cfg.encdec.param_size - + sizeof(struct asm_enc_cfg_blk_param_v2); + + enc_cfg.num_channels = channels; + enc_cfg.bits_per_sample = bits_per_sample; + enc_cfg.sample_rate = rate; + enc_cfg.is_signed = 1; + channel_mapping = enc_cfg.channel_mapping; + + memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL); + + if (q6dsp_map_channels(channel_mapping, channels)) + return -EINVAL; + + + return q6asm_ac_send_cmd_sync(ac, &enc_cfg); +} +EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support); + +/** + * q6asm_read() - read data of period size from audio client + * + * @ac: audio client pointer + * + * Return: Will be an negative value on error or zero on success + */ +int q6asm_read(struct audio_client *ac) +{ + struct asm_data_cmd_read_v2 read; + struct audio_port_data *port; + struct audio_buffer *ab; + int rc; + + if (!(ac->io_mode & ASM_SYNC_IO_MODE)) + return 0; + + port = &ac->port[SNDRV_PCM_STREAM_CAPTURE]; + q6asm_add_hdr(ac, &read.hdr, sizeof(read), false, ac->stream_id); + ab = &port->buf[port->dsp_buf]; + read.hdr.opcode = ASM_DATA_CMD_READ_V2; + read.buf_addr_lsw = lower_32_bits(ab->phys); + read.buf_addr_msw = upper_32_bits(ab->phys); + read.mem_map_handle = ac->port[SNDRV_PCM_STREAM_CAPTURE].mem_map_handle; + + read.buf_size = ab->size; + read.seq_id = port->dsp_buf; + read.hdr.token = port->dsp_buf; + + port->dsp_buf++; + + if (port->dsp_buf >= port->num_periods) + port->dsp_buf = 0; + + rc = apr_send_pkt(ac->adev, &read); + if (rc < 0) { + pr_err("read op[0x%x]rc[%d]\n", read.hdr.opcode, rc); + return rc; + } + + return 0; +} +EXPORT_SYMBOL_GPL(q6asm_read); + +static int __q6asm_open_read(struct audio_client *ac, + uint32_t format, uint16_t bits_per_sample) +{ + struct asm_stream_cmd_open_read_v3 open; + + q6asm_add_hdr(ac, &open.hdr, sizeof(open), true, ac->stream_id); + open.hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3; + /* Stream prio : High, provide meta info with encoded frames */ + open.src_endpointype = ASM_END_POINT_DEVICE_MATRIX; + + open.preprocopo_id = ASM_STREAM_POSTPROC_TOPO_ID_NONE; + open.bits_per_sample = bits_per_sample; + open.mode_flags = 0x0; + + open.mode_flags |= ASM_LEGACY_STREAM_SESSION << + ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ; + + switch (format) { + case FORMAT_LINEAR_PCM: + open.mode_flags |= 0x00; + open.enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2; + break; + default: + pr_err("Invalid format[%d]\n", format); + } + + return q6asm_ac_send_cmd_sync(ac, &open); +} + +/** + * q6asm_open_read() - Open audio client for reading + * + * @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_read(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample) +{ + return __q6asm_open_read(ac, format, bits_per_sample); +} +EXPORT_SYMBOL_GPL(q6asm_open_read); + +/** + * q6asm_write_async() - 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_async(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 rc = 0; + + if (!(ac->io_mode & ASM_SYNC_IO_MODE)) + return 0; + + port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK]; + q6asm_add_hdr(ac, &write.hdr, sizeof(write), false, + ac->stream_id); + + ab = &port->buf[port->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->num_periods) + port->dsp_buf = 0; + + rc = apr_send_pkt(ac->adev, &write); + if (rc < 0) + return rc; + + return 0; +} +EXPORT_SYMBOL_GPL(q6asm_write_async); + +static void q6asm_reset_buf_state(struct audio_client *ac) +{ + struct audio_port_data *port = NULL; + unsigned long flags; + int loopcnt = 0; + int cnt = 0; + int used; + + if (!(ac->io_mode & ASM_SYNC_IO_MODE)) + return; + + used = (ac->io_mode & ASM_TUN_WRITE_IO_MODE ? 1 : 0); + spin_lock_irqsave(&ac->buf_lock, flags); + for (loopcnt = 0; loopcnt <= SNDRV_PCM_STREAM_CAPTURE; loopcnt++) { + port = &ac->port[loopcnt]; + cnt = port->num_periods - 1; + port->dsp_buf = 0; + while (cnt >= 0) { + if (!port->buf) + continue; + port->buf[cnt].used = used; + cnt--; + } + } + spin_unlock_irqrestore(&ac->buf_lock, flags); +} + +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); + + 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; + break; + case CMD_CLOSE: + hdr.opcode = ASM_STREAM_CMD_CLOSE; + break; + default: + return -EINVAL; + } + + if (wait) + rc = q6asm_ac_send_cmd_sync(ac, &hdr); + else + return apr_send_pkt(ac->adev, &hdr); + + if (rc < 0) + return rc; + + 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 faa49ca1cde6..9ed15e0edaf0 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -1,8 +1,36 @@ // SPDX-License-Identifier: GPL-2.0 #ifndef __Q6_ASM_H__ #define __Q6_ASM_H__ +#include "q6dsp-common.h" +#include + +/* 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 ASM_CLIENT_EVENT_DATA_READ_DONE 0x100a + +enum { + LEGACY_PCM_MODE = 0, + LOW_LATENCY_PCM_MODE, + ULTRA_LOW_LATENCY_PCM_MODE, + ULL_POST_PROCESSING_PCM_MODE, +}; #define MAX_SESSIONS 16 +#define NO_TIMESTAMP 0xFF00 +#define FORMAT_LINEAR_PCM 0x0000 typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token, void *payload, void *priv); @@ -11,6 +39,27 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb, void *priv, int session_id, int perf_mode); void q6asm_audio_client_free(struct audio_client *ac); +int q6asm_write_async(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_open_read(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample); +int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac, + uint32_t rate, uint32_t channels, uint16_t bits_per_sample); +int q6asm_read(struct audio_client *ac); + +int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, + uint32_t rate, uint32_t channels, + u8 channel_map[PCM_FORMAT_MAX_NUM_CHANNEL], + 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,