| /* Copyright (c) 2012-2017, The Linux Foundation. All rights reserved. |
| * |
| * This program is free software; you can redistribute it and/or modify |
| * it under the terms of the GNU General Public License version 2 and |
| * only version 2 as published by the Free Software Foundation. |
| * |
| * This program is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
| * GNU General Public License for more details. |
| */ |
| |
| |
| #include <linux/init.h> |
| #include <linux/err.h> |
| #include <linux/module.h> |
| #include <linux/moduleparam.h> |
| #include <linux/time.h> |
| #include <linux/math64.h> |
| #include <linux/wait.h> |
| #include <linux/platform_device.h> |
| #include <linux/slab.h> |
| #include <sound/core.h> |
| #include <sound/soc.h> |
| #include <sound/soc-dapm.h> |
| #include <sound/pcm.h> |
| #include <sound/initval.h> |
| #include <sound/control.h> |
| #include <sound/q6asm-v2.h> |
| #include <sound/pcm_params.h> |
| #include <sound/audio_effects.h> |
| #include <asm/dma.h> |
| #include <linux/dma-mapping.h> |
| #include <linux/msm_audio_ion.h> |
| #include <linux/msm_audio.h> |
| |
| #include <sound/timer.h> |
| #include <sound/tlv.h> |
| |
| #include <sound/apr_audio-v2.h> |
| #include <sound/q6asm-v2.h> |
| #include <sound/compress_params.h> |
| #include <sound/compress_offload.h> |
| #include <sound/compress_driver.h> |
| #include <sound/msm-audio-effects-q6-v2.h> |
| #include "msm-pcm-routing-v2.h" |
| #include "msm-qti-pp-config.h" |
| |
| #define DSP_PP_BUFFERING_IN_MSEC 25 |
| #define PARTIAL_DRAIN_ACK_EARLY_BY_MSEC 150 |
| #define MP3_OUTPUT_FRAME_SZ 1152 |
| #define AAC_OUTPUT_FRAME_SZ 1024 |
| #define AC3_OUTPUT_FRAME_SZ 1536 |
| #define EAC3_OUTPUT_FRAME_SZ 1536 |
| #define DSP_NUM_OUTPUT_FRAME_BUFFERED 2 |
| #define FLAC_BLK_SIZE_LIMIT 65535 |
| |
| /* Timestamp mode payload offsets */ |
| #define CAPTURE_META_DATA_TS_OFFSET_LSW 6 |
| #define CAPTURE_META_DATA_TS_OFFSET_MSW 7 |
| |
| /* decoder parameter length */ |
| #define DDP_DEC_MAX_NUM_PARAM 18 |
| |
| /* Default values used if user space does not set */ |
| #define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024) |
| #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024) |
| #define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4) |
| #define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4) |
| |
| #define COMPRESSED_LR_VOL_MAX_STEPS 0x2000 |
| const DECLARE_TLV_DB_LINEAR(msm_compr_vol_gain, 0, |
| COMPRESSED_LR_VOL_MAX_STEPS); |
| |
| /* Stream id switches between 1 and 2 */ |
| #define NEXT_STREAM_ID(stream_id) ((stream_id & 1) + 1) |
| |
| #define STREAM_ARRAY_INDEX(stream_id) (stream_id - 1) |
| |
| #define MAX_NUMBER_OF_STREAMS 2 |
| |
| struct msm_compr_gapless_state { |
| bool set_next_stream_id; |
| int32_t stream_opened[MAX_NUMBER_OF_STREAMS]; |
| uint32_t initial_samples_drop; |
| uint32_t trailing_samples_drop; |
| uint32_t gapless_transition; |
| bool use_dsp_gapless_mode; |
| union snd_codec_options codec_options; |
| }; |
| |
| static unsigned int supported_sample_rates[] = { |
| 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, |
| 88200, 96000, 128000, 144000, 176400, 192000, 352800, 384000, 2822400, |
| 5644800 |
| }; |
| |
| struct msm_compr_pdata { |
| struct snd_compr_stream *cstream[MSM_FRONTEND_DAI_MAX]; |
| uint32_t volume[MSM_FRONTEND_DAI_MAX][2]; /* For both L & R */ |
| struct msm_compr_audio_effects *audio_effects[MSM_FRONTEND_DAI_MAX]; |
| bool use_dsp_gapless_mode; |
| bool use_legacy_api; /* indicates use older asm apis*/ |
| struct msm_compr_dec_params *dec_params[MSM_FRONTEND_DAI_MAX]; |
| struct msm_compr_ch_map *ch_map[MSM_FRONTEND_DAI_MAX]; |
| }; |
| |
| struct msm_compr_audio { |
| struct snd_compr_stream *cstream; |
| struct snd_compr_caps compr_cap; |
| struct snd_compr_codec_caps codec_caps; |
| struct snd_compr_params codec_param; |
| struct audio_client *audio_client; |
| |
| uint32_t codec; |
| uint32_t compr_passthr; |
| void *buffer; /* virtual address */ |
| phys_addr_t buffer_paddr; /* physical address */ |
| uint32_t app_pointer; |
| uint32_t buffer_size; |
| uint32_t byte_offset; |
| uint64_t copied_total; /* bytes consumed by DSP */ |
| uint64_t bytes_received; /* from userspace */ |
| uint64_t bytes_sent; /* to DSP */ |
| |
| uint64_t received_total; /* bytes received from DSP */ |
| uint64_t bytes_copied; /* to userspace */ |
| uint64_t bytes_read; /* from DSP */ |
| uint32_t bytes_read_offset; /* bytes read offset */ |
| |
| uint32_t ts_header_offset; /* holds the timestamp header offset */ |
| |
| int32_t first_buffer; |
| int32_t last_buffer; |
| int32_t partial_drain_delay; |
| |
| uint16_t session_id; |
| |
| uint32_t sample_rate; |
| uint32_t num_channels; |
| |
| /* |
| * convention - commands coming from the same thread |
| * can use the common cmd_ack var. Others (e.g drain/EOS) |
| * must use separate vars to track command status. |
| */ |
| uint32_t cmd_ack; |
| uint32_t cmd_interrupt; |
| uint32_t drain_ready; |
| uint32_t eos_ack; |
| |
| uint32_t stream_available; |
| uint32_t next_stream; |
| |
| uint32_t run_mode; |
| uint32_t start_delay_lsw; |
| uint32_t start_delay_msw; |
| |
| uint64_t marker_timestamp; |
| |
| struct msm_compr_gapless_state gapless_state; |
| |
| atomic_t start; |
| atomic_t eos; |
| atomic_t drain; |
| atomic_t xrun; |
| atomic_t close; |
| atomic_t wait_on_close; |
| atomic_t error; |
| |
| wait_queue_head_t eos_wait; |
| wait_queue_head_t drain_wait; |
| wait_queue_head_t close_wait; |
| wait_queue_head_t wait_for_stream_avail; |
| |
| spinlock_t lock; |
| }; |
| |
| const u32 compr_codecs[] = { |
| SND_AUDIOCODEC_AC3, SND_AUDIOCODEC_EAC3, SND_AUDIOCODEC_DTS, |
| SND_AUDIOCODEC_DSD, SND_AUDIOCODEC_TRUEHD, SND_AUDIOCODEC_IEC61937}; |
| |
| struct query_audio_effect { |
| uint32_t mod_id; |
| uint32_t parm_id; |
| uint32_t size; |
| uint32_t offset; |
| uint32_t device; |
| }; |
| |
| struct msm_compr_audio_effects { |
| struct bass_boost_params bass_boost; |
| struct pbe_params pbe; |
| struct virtualizer_params virtualizer; |
| struct reverb_params reverb; |
| struct eq_params equalizer; |
| struct soft_volume_params volume; |
| struct query_audio_effect query; |
| }; |
| |
| struct msm_compr_dec_params { |
| struct snd_dec_ddp ddp_params; |
| }; |
| |
| struct msm_compr_ch_map { |
| bool set_ch_map; |
| char channel_map[PCM_FORMAT_MAX_NUM_CHANNEL]; |
| }; |
| |
| static int msm_compr_send_dec_params(struct snd_compr_stream *cstream, |
| struct msm_compr_dec_params *dec_params, |
| int stream_id); |
| |
| static int msm_compr_set_render_mode(struct msm_compr_audio *prtd, |
| uint32_t render_mode) { |
| int ret = -EINVAL; |
| struct audio_client *ac = prtd->audio_client; |
| |
| pr_debug("%s, got render mode %u\n", __func__, render_mode); |
| |
| if (render_mode == SNDRV_COMPRESS_RENDER_MODE_AUDIO_MASTER) { |
| render_mode = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT; |
| } else if (render_mode == SNDRV_COMPRESS_RENDER_MODE_STC_MASTER) { |
| render_mode = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC; |
| prtd->run_mode = ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY; |
| } else { |
| pr_err("%s, Invalid render mode %u\n", __func__, |
| render_mode); |
| ret = -EINVAL; |
| goto exit; |
| } |
| |
| ret = q6asm_send_mtmx_strtr_render_mode(ac, render_mode); |
| if (ret) { |
| pr_err("%s, Render mode can't be set error %d\n", __func__, |
| ret); |
| } |
| exit: |
| return ret; |
| } |
| |
| static int msm_compr_set_clk_rec_mode(struct audio_client *ac, |
| uint32_t clk_rec_mode) { |
| int ret = -EINVAL; |
| |
| pr_debug("%s, got clk rec mode %u\n", __func__, clk_rec_mode); |
| |
| if (clk_rec_mode == SNDRV_COMPRESS_CLK_REC_MODE_NONE) { |
| clk_rec_mode = ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_NONE; |
| } else if (clk_rec_mode == SNDRV_COMPRESS_CLK_REC_MODE_AUTO) { |
| clk_rec_mode = ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_AUTO; |
| } else { |
| pr_err("%s, Invalid clk rec_mode mode %u\n", __func__, |
| clk_rec_mode); |
| ret = -EINVAL; |
| goto exit; |
| } |
| |
| ret = q6asm_send_mtmx_strtr_clk_rec_mode(ac, clk_rec_mode); |
| if (ret) { |
| pr_err("%s, clk rec mode can't be set, error %d\n", __func__, |
| ret); |
| } |
| |
| exit: |
| return ret; |
| } |
| |
| static int msm_compr_set_render_window(struct audio_client *ac, |
| uint32_t ws_lsw, uint32_t ws_msw, |
| uint32_t we_lsw, uint32_t we_msw) |
| { |
| int ret = -EINVAL; |
| struct asm_session_mtmx_strtr_param_window_v2_t asm_mtmx_strtr_window; |
| uint32_t param_id; |
| |
| pr_debug("%s, ws_lsw 0x%x ws_msw 0x%x we_lsw 0x%x we_ms 0x%x\n", |
| __func__, ws_lsw, ws_msw, we_lsw, we_msw); |
| |
| memset(&asm_mtmx_strtr_window, 0, |
| sizeof(struct asm_session_mtmx_strtr_param_window_v2_t)); |
| asm_mtmx_strtr_window.window_lsw = ws_lsw; |
| asm_mtmx_strtr_window.window_msw = ws_msw; |
| param_id = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2; |
| ret = q6asm_send_mtmx_strtr_window(ac, &asm_mtmx_strtr_window, |
| param_id); |
| if (ret) { |
| pr_err("%s, start window can't be set error %d\n", __func__, |
| ret); |
| goto exit; |
| } |
| |
| asm_mtmx_strtr_window.window_lsw = we_lsw; |
| asm_mtmx_strtr_window.window_msw = we_msw; |
| param_id = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2; |
| ret = q6asm_send_mtmx_strtr_window(ac, &asm_mtmx_strtr_window, |
| param_id); |
| if (ret) { |
| pr_err("%s, end window can't be set error %d\n", __func__, |
| ret); |
| } |
| |
| exit: |
| return ret; |
| } |
| |
| static int msm_compr_enable_adjust_session_clock(struct audio_client *ac, |
| bool enable) |
| { |
| int ret; |
| |
| pr_debug("%s, enable adjust_session %d\n", __func__, enable); |
| |
| ret = q6asm_send_mtmx_strtr_enable_adjust_session_clock(ac, enable); |
| if (ret) |
| pr_err("%s, adjust session clock can't be set error %d\n", |
| __func__, ret); |
| |
| return ret; |
| } |
| |
| static int msm_compr_adjust_session_clock(struct audio_client *ac, |
| uint32_t adjust_session_lsw, uint32_t adjust_session_msw) |
| { |
| int ret; |
| |
| pr_debug("%s, adjust_session_time_msw 0x%x adjust_session_time_lsw 0x%x\n", |
| __func__, adjust_session_msw, adjust_session_lsw); |
| |
| ret = q6asm_adjust_session_clock(ac, |
| adjust_session_lsw, |
| adjust_session_msw); |
| if (ret) |
| pr_err("%s, adjust session clock can't be set error %d\n", |
| __func__, ret); |
| |
| return ret; |
| } |
| |
| static int msm_compr_set_volume(struct snd_compr_stream *cstream, |
| uint32_t volume_l, uint32_t volume_r) |
| { |
| struct msm_compr_audio *prtd; |
| int rc = 0; |
| uint32_t avg_vol, gain_list[VOLUME_CONTROL_MAX_CHANNELS]; |
| uint32_t num_channels; |
| struct snd_soc_pcm_runtime *rtd; |
| struct msm_compr_pdata *pdata; |
| bool use_default = true; |
| u8 *chmap = NULL; |
| |
| pr_debug("%s: volume_l %d volume_r %d\n", |
| __func__, volume_l, volume_r); |
| if (!cstream || !cstream->runtime) { |
| pr_err("%s: session not active\n", __func__); |
| return -EPERM; |
| } |
| rtd = cstream->private_data; |
| prtd = cstream->runtime->private_data; |
| |
| if (!rtd || !rtd->platform || !prtd || !prtd->audio_client) { |
| pr_err("%s: invalid rtd, prtd or audio client", __func__); |
| return rc; |
| } |
| pdata = snd_soc_platform_get_drvdata(rtd->platform); |
| |
| if (prtd->compr_passthr != LEGACY_PCM) { |
| pr_debug("%s: No volume config for passthrough %d\n", |
| __func__, prtd->compr_passthr); |
| return rc; |
| } |
| |
| use_default = !(pdata->ch_map[rtd->dai_link->id]->set_ch_map); |
| chmap = pdata->ch_map[rtd->dai_link->id]->channel_map; |
| num_channels = prtd->num_channels; |
| |
| if (prtd->num_channels > 2) { |
| /* |
| * Currently the left and right gains are averaged an applied |
| * to all channels. This might not be desirable. But currently, |
| * there exists no API in userspace to send a list of gains for |
| * each channel either. If such an API does become available, |
| * the mixer control must be updated to accept more than 2 |
| * channel gains. |
| * |
| */ |
| avg_vol = (volume_l + volume_r) / 2; |
| rc = q6asm_set_volume(prtd->audio_client, avg_vol); |
| } else { |
| gain_list[0] = volume_l; |
| gain_list[1] = volume_r; |
| /* force sending FR/FL/FC volume for mono */ |
| if (prtd->num_channels == 1) { |
| gain_list[2] = volume_l; |
| num_channels = 3; |
| use_default = true; |
| } |
| rc = q6asm_set_multich_gain(prtd->audio_client, num_channels, |
| gain_list, chmap, use_default); |
| } |
| |
| if (rc < 0) |
| pr_err("%s: Send vol gain command failed rc=%d\n", |
| __func__, rc); |
| |
| return rc; |
| } |
| |
| static int msm_compr_send_ddp_cfg(struct audio_client *ac, |
| struct snd_dec_ddp *ddp, |
| int stream_id) |
| { |
| int i, rc; |
| |
| pr_debug("%s\n", __func__); |
| for (i = 0; i < ddp->params_length; i++) { |
| rc = q6asm_ds1_set_stream_endp_params(ac, ddp->params_id[i], |
| ddp->params_value[i], |
| stream_id); |
| if (rc) { |
| pr_err("sending params_id: %d failed\n", |
| ddp->params_id[i]); |
| return rc; |
| } |
| } |
| return 0; |
| } |
| |
| static int msm_compr_send_buffer(struct msm_compr_audio *prtd) |
| { |
| int buffer_length; |
| uint64_t bytes_available; |
| struct audio_aio_write_param param; |
| struct snd_codec_metadata *buff_addr; |
| |
| if (!atomic_read(&prtd->start)) { |
| pr_err("%s: stream is not in started state\n", __func__); |
| return -EINVAL; |
| } |
| |
| |
| if (atomic_read(&prtd->xrun)) { |
| WARN(1, "%s called while xrun is true", __func__); |
| return -EPERM; |
| } |
| |
| pr_debug("%s: bytes_received = %llu copied_total = %llu\n", |
| __func__, prtd->bytes_received, prtd->copied_total); |
| if (prtd->first_buffer && prtd->gapless_state.use_dsp_gapless_mode && |
| prtd->compr_passthr == LEGACY_PCM) |
| q6asm_stream_send_meta_data(prtd->audio_client, |
| prtd->audio_client->stream_id, |
| prtd->gapless_state.initial_samples_drop, |
| prtd->gapless_state.trailing_samples_drop); |
| |
| buffer_length = prtd->codec_param.buffer.fragment_size; |
| bytes_available = prtd->bytes_received - prtd->copied_total; |
| if (bytes_available < prtd->codec_param.buffer.fragment_size) |
| buffer_length = bytes_available; |
| |
| if (prtd->byte_offset + buffer_length > prtd->buffer_size) { |
| buffer_length = (prtd->buffer_size - prtd->byte_offset); |
| pr_debug("%s: wrap around situation, send partial data %d now", |
| __func__, buffer_length); |
| } |
| |
| if (buffer_length) { |
| param.paddr = prtd->buffer_paddr + prtd->byte_offset; |
| WARN(prtd->byte_offset % 32 != 0, "offset %x not multiple of 32\n", |
| prtd->byte_offset); |
| } else { |
| param.paddr = prtd->buffer_paddr; |
| } |
| param.len = buffer_length; |
| if (prtd->ts_header_offset) { |
| buff_addr = (struct snd_codec_metadata *) |
| (prtd->buffer + prtd->byte_offset); |
| param.len = buff_addr->length; |
| param.msw_ts = (uint32_t) |
| ((buff_addr->timestamp & 0xFFFFFFFF00000000LL) >> 32); |
| param.lsw_ts = (uint32_t) (buff_addr->timestamp & 0xFFFFFFFFLL); |
| param.paddr += prtd->ts_header_offset; |
| param.flags = SET_TIMESTAMP; |
| param.metadata_len = prtd->ts_header_offset; |
| } else { |
| param.msw_ts = 0; |
| param.lsw_ts = 0; |
| param.flags = NO_TIMESTAMP; |
| param.metadata_len = 0; |
| } |
| param.uid = buffer_length; |
| param.last_buffer = prtd->last_buffer; |
| |
| pr_debug("%s: sending %d bytes to DSP byte_offset = %d\n", |
| __func__, param.len, prtd->byte_offset); |
| if (q6asm_async_write(prtd->audio_client, ¶m) < 0) { |
| pr_err("%s:q6asm_async_write failed\n", __func__); |
| } else { |
| prtd->bytes_sent += buffer_length; |
| if (prtd->first_buffer) |
| prtd->first_buffer = 0; |
| } |
| |
| return 0; |
| } |
| |
| static int msm_compr_read_buffer(struct msm_compr_audio *prtd) |
| { |
| int buffer_length; |
| uint64_t bytes_available; |
| uint64_t buffer_sent; |
| struct audio_aio_read_param param; |
| int ret; |
| |
| if (!atomic_read(&prtd->start)) { |
| pr_err("%s: stream is not in started state\n", __func__); |
| return -EINVAL; |
| } |
| |
| buffer_length = prtd->codec_param.buffer.fragment_size - |
| prtd->ts_header_offset; |
| bytes_available = prtd->received_total - prtd->bytes_copied; |
| buffer_sent = prtd->bytes_read - prtd->bytes_copied; |
| if (buffer_sent + buffer_length + prtd->ts_header_offset |
| > prtd->buffer_size) { |
| pr_debug(" %s : Buffer is Full bytes_available: %llu\n", |
| __func__, bytes_available); |
| return 0; |
| } |
| |
| memset(¶m, 0x0, sizeof(struct audio_aio_read_param)); |
| param.paddr = prtd->buffer_paddr + prtd->bytes_read_offset + |
| prtd->ts_header_offset; |
| param.len = buffer_length; |
| param.uid = buffer_length; |
| param.flags = prtd->codec_param.codec.flags; |
| |
| pr_debug("%s: reading %d bytes from DSP byte_offset = %llu\n", |
| __func__, buffer_length, prtd->bytes_read); |
| ret = q6asm_async_read(prtd->audio_client, ¶m); |
| if (ret < 0) { |
| pr_err("%s: q6asm_async_read failed - %d\n", |
| __func__, ret); |
| return ret; |
| } |
| prtd->bytes_read += buffer_length; |
| prtd->bytes_read_offset += buffer_length; |
| if (prtd->bytes_read_offset >= prtd->buffer_size) |
| prtd->bytes_read_offset -= prtd->buffer_size; |
| |
| return 0; |
| } |
| |
| static void compr_event_handler(uint32_t opcode, |
| uint32_t token, uint32_t *payload, void *priv) |
| { |
| struct msm_compr_audio *prtd = priv; |
| struct snd_compr_stream *cstream; |
| struct audio_client *ac; |
| uint32_t chan_mode = 0; |
| uint32_t sample_rate = 0; |
| uint64_t bytes_available; |
| int stream_id; |
| uint32_t stream_index; |
| unsigned long flags; |
| uint64_t read_size; |
| uint32_t *buff_addr; |
| struct snd_soc_pcm_runtime *rtd; |
| int ret = 0; |
| |
| if (!prtd) { |
| pr_err("%s: prtd is NULL\n", __func__); |
| return; |
| } |
| cstream = prtd->cstream; |
| if (!cstream) { |
| pr_err("%s: cstream is NULL\n", __func__); |
| return; |
| } |
| |
| ac = prtd->audio_client; |
| |
| /* |
| * Token for rest of the compressed commands use to set |
| * session id, stream id, dir etc. |
| */ |
| stream_id = q6asm_get_stream_id_from_token(token); |
| |
| pr_debug("%s opcode =%08x\n", __func__, opcode); |
| switch (opcode) { |
| case ASM_DATA_EVENT_WRITE_DONE_V2: |
| spin_lock_irqsave(&prtd->lock, flags); |
| |
| if (payload[3]) { |
| pr_err("%s: WRITE FAILED w/ err 0x%x !, paddr 0x%x, byte_offset=%d,copied_total=%llu,token=%d\n", |
| __func__, |
| payload[3], |
| payload[0], |
| prtd->byte_offset, |
| prtd->copied_total, token); |
| |
| if (atomic_cmpxchg(&prtd->drain, 1, 0) && |
| prtd->last_buffer) { |
| pr_debug("%s: wake up on drain\n", __func__); |
| prtd->drain_ready = 1; |
| wake_up(&prtd->drain_wait); |
| prtd->last_buffer = 0; |
| } else { |
| atomic_set(&prtd->start, 0); |
| } |
| } else { |
| pr_debug("ASM_DATA_EVENT_WRITE_DONE_V2 offset %d, length %d\n", |
| prtd->byte_offset, token); |
| } |
| |
| /* |
| * Token for WRITE command represents the amount of data |
| * written to ADSP in the last write, update offset and |
| * total copied data accordingly. |
| */ |
| if (prtd->ts_header_offset) { |
| /* Always assume that the data will be sent to DSP on |
| * frame boundary. |
| * i.e, one frame of userspace write will result in |
| * one kernel write to DSP. This is needed as |
| * timestamp will be sent per frame. |
| */ |
| prtd->byte_offset += |
| prtd->codec_param.buffer.fragment_size; |
| prtd->copied_total += |
| prtd->codec_param.buffer.fragment_size; |
| } else { |
| prtd->byte_offset += token; |
| prtd->copied_total += token; |
| } |
| if (prtd->byte_offset >= prtd->buffer_size) |
| prtd->byte_offset -= prtd->buffer_size; |
| |
| snd_compr_fragment_elapsed(cstream); |
| |
| if (!atomic_read(&prtd->start)) { |
| /* Writes must be restarted from _copy() */ |
| pr_debug("write_done received while not started, treat as xrun"); |
| atomic_set(&prtd->xrun, 1); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| } |
| |
| bytes_available = prtd->bytes_received - prtd->copied_total; |
| if (bytes_available < cstream->runtime->fragment_size) { |
| pr_debug("WRITE_DONE Insufficient data to send. break out\n"); |
| atomic_set(&prtd->xrun, 1); |
| |
| if (prtd->last_buffer) |
| prtd->last_buffer = 0; |
| if (atomic_read(&prtd->drain)) { |
| pr_debug("wake up on drain\n"); |
| prtd->drain_ready = 1; |
| wake_up(&prtd->drain_wait); |
| atomic_set(&prtd->drain, 0); |
| } |
| } else if ((bytes_available == cstream->runtime->fragment_size) |
| && atomic_read(&prtd->drain)) { |
| prtd->last_buffer = 1; |
| msm_compr_send_buffer(prtd); |
| prtd->last_buffer = 0; |
| } else |
| msm_compr_send_buffer(prtd); |
| |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| |
| case ASM_DATA_EVENT_READ_DONE_V2: |
| spin_lock_irqsave(&prtd->lock, flags); |
| |
| pr_debug("ASM_DATA_EVENT_READ_DONE_V2 offset %d, length %d\n", |
| prtd->byte_offset, payload[4]); |
| |
| if (prtd->ts_header_offset) { |
| /* Update the header for received buffer */ |
| buff_addr = prtd->buffer + prtd->byte_offset; |
| /* Write the length of the buffer */ |
| *buff_addr = prtd->codec_param.buffer.fragment_size |
| - prtd->ts_header_offset; |
| buff_addr++; |
| /* Write the offset */ |
| *buff_addr = prtd->ts_header_offset; |
| buff_addr++; |
| /* Write the TS LSW */ |
| *buff_addr = payload[CAPTURE_META_DATA_TS_OFFSET_LSW]; |
| buff_addr++; |
| /* Write the TS MSW */ |
| *buff_addr = payload[CAPTURE_META_DATA_TS_OFFSET_MSW]; |
| } |
| /* Always assume read_size is same as fragment_size */ |
| read_size = prtd->codec_param.buffer.fragment_size; |
| prtd->byte_offset += read_size; |
| prtd->received_total += read_size; |
| if (prtd->byte_offset >= prtd->buffer_size) |
| prtd->byte_offset -= prtd->buffer_size; |
| |
| snd_compr_fragment_elapsed(cstream); |
| |
| if (!atomic_read(&prtd->start)) { |
| pr_debug("read_done received while not started, treat as xrun"); |
| atomic_set(&prtd->xrun, 1); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| } |
| msm_compr_read_buffer(prtd); |
| |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| |
| case ASM_DATA_EVENT_RENDERED_EOS: |
| spin_lock_irqsave(&prtd->lock, flags); |
| pr_debug("%s: ASM_DATA_CMDRSP_EOS token 0x%x,stream id %d\n", |
| __func__, token, stream_id); |
| if (atomic_read(&prtd->eos) && |
| !prtd->gapless_state.set_next_stream_id) { |
| pr_debug("ASM_DATA_CMDRSP_EOS wake up\n"); |
| prtd->eos_ack = 1; |
| wake_up(&prtd->eos_wait); |
| } |
| atomic_set(&prtd->eos, 0); |
| stream_index = STREAM_ARRAY_INDEX(stream_id); |
| if (stream_index >= MAX_NUMBER_OF_STREAMS || |
| stream_index < 0) { |
| pr_err("%s: Invalid stream index %d", __func__, |
| stream_index); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| } |
| |
| if (prtd->gapless_state.set_next_stream_id && |
| prtd->gapless_state.stream_opened[stream_index]) { |
| pr_debug("%s: CMD_CLOSE stream_id %d\n", |
| __func__, stream_id); |
| q6asm_stream_cmd_nowait(ac, CMD_CLOSE, stream_id); |
| atomic_set(&prtd->close, 1); |
| prtd->gapless_state.stream_opened[stream_index] = 0; |
| prtd->gapless_state.set_next_stream_id = false; |
| } |
| if (prtd->gapless_state.gapless_transition) |
| prtd->gapless_state.gapless_transition = 0; |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| case ASM_STREAM_PP_EVENT: |
| case ASM_STREAM_CMD_ENCDEC_EVENTS: |
| pr_debug("%s: ASM_STREAM_EVENT(0x%x)\n", __func__, opcode); |
| rtd = cstream->private_data; |
| if (!rtd) { |
| pr_err("%s: rtd is NULL\n", __func__); |
| return; |
| } |
| |
| ret = msm_adsp_inform_mixer_ctl(rtd, payload); |
| if (ret) { |
| pr_err("%s: failed to inform mixer ctrl. err = %d\n", |
| __func__, ret); |
| return; |
| } |
| break; |
| case ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY: |
| case ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY: { |
| pr_debug("ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY\n"); |
| chan_mode = payload[1] >> 16; |
| sample_rate = payload[2] >> 16; |
| if (prtd && (chan_mode != prtd->num_channels || |
| sample_rate != prtd->sample_rate)) { |
| prtd->num_channels = chan_mode; |
| prtd->sample_rate = sample_rate; |
| } |
| } |
| /* Fallthrough here */ |
| case APR_BASIC_RSP_RESULT: { |
| switch (payload[0]) { |
| case ASM_SESSION_CMD_RUN_V2: |
| /* check if the first buffer need to be sent to DSP */ |
| pr_debug("ASM_SESSION_CMD_RUN_V2\n"); |
| |
| /* FIXME: A state is a better way, dealing with this */ |
| spin_lock_irqsave(&prtd->lock, flags); |
| |
| if (cstream->direction == SND_COMPRESS_CAPTURE) { |
| atomic_set(&prtd->start, 1); |
| msm_compr_read_buffer(prtd); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| } |
| |
| if (!prtd->bytes_sent) { |
| bytes_available = prtd->bytes_received - |
| prtd->copied_total; |
| if (bytes_available < |
| cstream->runtime->fragment_size) { |
| pr_debug("CMD_RUN_V2 Insufficient data to send. break out\n"); |
| atomic_set(&prtd->xrun, 1); |
| } else { |
| msm_compr_send_buffer(prtd); |
| } |
| } |
| |
| /* |
| * The condition below ensures playback finishes in the |
| * follow cornercase |
| * WRITE(last buffer) |
| * WAIT_FOR_DRAIN |
| * PAUSE |
| * WRITE_DONE(X) |
| * RESUME |
| */ |
| if ((prtd->copied_total == prtd->bytes_sent) && |
| atomic_read(&prtd->drain)) { |
| pr_debug("RUN ack, wake up & continue pending drain\n"); |
| |
| if (prtd->last_buffer) |
| prtd->last_buffer = 0; |
| |
| prtd->drain_ready = 1; |
| wake_up(&prtd->drain_wait); |
| atomic_set(&prtd->drain, 0); |
| } |
| |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| case ASM_STREAM_CMD_FLUSH: |
| pr_debug("%s: ASM_STREAM_CMD_FLUSH:", __func__); |
| pr_debug("token 0x%x, stream id %d\n", token, |
| stream_id); |
| prtd->cmd_ack = 1; |
| break; |
| case ASM_DATA_CMD_REMOVE_INITIAL_SILENCE: |
| pr_debug("%s: ASM_DATA_CMD_REMOVE_INITIAL_SILENCE:", |
| __func__); |
| pr_debug("token 0x%x, stream id = %d\n", token, |
| stream_id); |
| break; |
| case ASM_DATA_CMD_REMOVE_TRAILING_SILENCE: |
| pr_debug("%s: ASM_DATA_CMD_REMOVE_TRAILING_SILENCE:", |
| __func__); |
| pr_debug("token = 0x%x, stream id = %d\n", token, |
| stream_id); |
| break; |
| case ASM_STREAM_CMD_CLOSE: |
| pr_debug("%s: ASM_DATA_CMD_CLOSE:", __func__); |
| pr_debug("token 0x%x, stream id %d\n", token, |
| stream_id); |
| /* |
| * wakeup wait for stream avail on stream 3 |
| * after stream 1 ends. |
| */ |
| if (prtd->next_stream) { |
| pr_debug("%s:CLOSE:wakeup wait for stream\n", |
| __func__); |
| prtd->stream_available = 1; |
| wake_up(&prtd->wait_for_stream_avail); |
| prtd->next_stream = 0; |
| } |
| if (atomic_read(&prtd->close) && |
| atomic_read(&prtd->wait_on_close)) { |
| prtd->cmd_ack = 1; |
| wake_up(&prtd->close_wait); |
| } |
| atomic_set(&prtd->close, 0); |
| break; |
| case ASM_STREAM_CMD_REGISTER_PP_EVENTS: |
| pr_debug("%s: ASM_STREAM_CMD_REGISTER_PP_EVENTS:", |
| __func__); |
| break; |
| default: |
| break; |
| } |
| break; |
| } |
| case ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3: |
| pr_debug("%s: ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3\n", |
| __func__); |
| break; |
| case RESET_EVENTS: |
| pr_err("%s: Received reset events CB, move to error state", |
| __func__); |
| spin_lock_irqsave(&prtd->lock, flags); |
| /* |
| * Since ADSP is down, let this driver pretend that it copied |
| * all the bytes received, so that next write will be triggered |
| */ |
| prtd->copied_total = prtd->bytes_received; |
| snd_compr_fragment_elapsed(cstream); |
| atomic_set(&prtd->error, 1); |
| wake_up(&prtd->drain_wait); |
| if (atomic_cmpxchg(&prtd->eos, 1, 0)) { |
| pr_debug("%s:unblock eos wait queues", __func__); |
| wake_up(&prtd->eos_wait); |
| } |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| default: |
| pr_debug("%s: Not Supported Event opcode[0x%x]\n", |
| __func__, opcode); |
| break; |
| } |
| } |
| |
| static int msm_compr_get_partial_drain_delay(int frame_sz, int sample_rate) |
| { |
| int delay_time_ms = 0; |
| |
| delay_time_ms = ((DSP_NUM_OUTPUT_FRAME_BUFFERED * frame_sz * 1000) / |
| sample_rate) + DSP_PP_BUFFERING_IN_MSEC; |
| delay_time_ms = delay_time_ms > PARTIAL_DRAIN_ACK_EARLY_BY_MSEC ? |
| delay_time_ms - PARTIAL_DRAIN_ACK_EARLY_BY_MSEC : 0; |
| |
| pr_debug("%s: frame_sz %d, sample_rate %d, partial drain delay %d\n", |
| __func__, frame_sz, sample_rate, delay_time_ms); |
| return delay_time_ms; |
| } |
| |
| static void populate_codec_list(struct msm_compr_audio *prtd) |
| { |
| pr_debug("%s\n", __func__); |
| prtd->compr_cap.direction = SND_COMPRESS_PLAYBACK; |
| prtd->compr_cap.min_fragment_size = |
| COMPR_PLAYBACK_MIN_FRAGMENT_SIZE; |
| prtd->compr_cap.max_fragment_size = |
| COMPR_PLAYBACK_MAX_FRAGMENT_SIZE; |
| prtd->compr_cap.min_fragments = |
| COMPR_PLAYBACK_MIN_NUM_FRAGMENTS; |
| prtd->compr_cap.max_fragments = |
| COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; |
| prtd->compr_cap.num_codecs = 17; |
| prtd->compr_cap.codecs[0] = SND_AUDIOCODEC_MP3; |
| prtd->compr_cap.codecs[1] = SND_AUDIOCODEC_AAC; |
| prtd->compr_cap.codecs[2] = SND_AUDIOCODEC_AC3; |
| prtd->compr_cap.codecs[3] = SND_AUDIOCODEC_EAC3; |
| prtd->compr_cap.codecs[4] = SND_AUDIOCODEC_MP2; |
| prtd->compr_cap.codecs[5] = SND_AUDIOCODEC_PCM; |
| prtd->compr_cap.codecs[6] = SND_AUDIOCODEC_WMA; |
| prtd->compr_cap.codecs[7] = SND_AUDIOCODEC_WMA_PRO; |
| prtd->compr_cap.codecs[8] = SND_AUDIOCODEC_FLAC; |
| prtd->compr_cap.codecs[9] = SND_AUDIOCODEC_VORBIS; |
| prtd->compr_cap.codecs[10] = SND_AUDIOCODEC_ALAC; |
| prtd->compr_cap.codecs[11] = SND_AUDIOCODEC_APE; |
| prtd->compr_cap.codecs[12] = SND_AUDIOCODEC_DTS; |
| prtd->compr_cap.codecs[13] = SND_AUDIOCODEC_DSD; |
| prtd->compr_cap.codecs[14] = SND_AUDIOCODEC_APTX; |
| prtd->compr_cap.codecs[15] = SND_AUDIOCODEC_TRUEHD; |
| prtd->compr_cap.codecs[16] = SND_AUDIOCODEC_IEC61937; |
| } |
| |
| static int msm_compr_send_media_format_block(struct snd_compr_stream *cstream, |
| int stream_id, |
| bool use_gapless_codec_options) |
| { |
| struct snd_compr_runtime *runtime = cstream->runtime; |
| struct msm_compr_audio *prtd = runtime->private_data; |
| struct snd_soc_pcm_runtime *rtd = cstream->private_data; |
| struct msm_compr_pdata *pdata = |
| snd_soc_platform_get_drvdata(rtd->platform); |
| struct asm_aac_cfg aac_cfg; |
| struct asm_wma_cfg wma_cfg; |
| struct asm_wmapro_cfg wma_pro_cfg; |
| struct asm_flac_cfg flac_cfg; |
| struct asm_vorbis_cfg vorbis_cfg; |
| struct asm_alac_cfg alac_cfg; |
| struct asm_ape_cfg ape_cfg; |
| struct asm_dsd_cfg dsd_cfg; |
| struct aptx_dec_bt_addr_cfg aptx_cfg; |
| union snd_codec_options *codec_options; |
| |
| int ret = 0; |
| uint16_t bit_width; |
| bool use_default_chmap = true; |
| char *chmap = NULL; |
| uint16_t sample_word_size; |
| |
| pr_debug("%s: use_gapless_codec_options %d\n", |
| __func__, use_gapless_codec_options); |
| |
| if (use_gapless_codec_options) |
| codec_options = &(prtd->gapless_state.codec_options); |
| else |
| codec_options = &(prtd->codec_param.codec.options); |
| |
| if (!codec_options) { |
| pr_err("%s: codec_options is NULL\n", __func__); |
| return -EINVAL; |
| } |
| |
| switch (prtd->codec) { |
| case FORMAT_LINEAR_PCM: |
| pr_debug("SND_AUDIOCODEC_PCM\n"); |
| if (pdata->ch_map[rtd->dai_link->id]) { |
| use_default_chmap = |
| !(pdata->ch_map[rtd->dai_link->id]->set_ch_map); |
| chmap = |
| pdata->ch_map[rtd->dai_link->id]->channel_map; |
| } |
| |
| switch (prtd->codec_param.codec.format) { |
| case SNDRV_PCM_FORMAT_S32_LE: |
| bit_width = 32; |
| sample_word_size = 32; |
| break; |
| case SNDRV_PCM_FORMAT_S24_LE: |
| bit_width = 24; |
| sample_word_size = 32; |
| break; |
| case SNDRV_PCM_FORMAT_S24_3LE: |
| bit_width = 24; |
| sample_word_size = 24; |
| break; |
| case SNDRV_PCM_FORMAT_S16_LE: |
| default: |
| bit_width = 16; |
| sample_word_size = 16; |
| break; |
| } |
| ret = q6asm_media_format_block_pcm_format_support_v4( |
| prtd->audio_client, |
| prtd->sample_rate, |
| prtd->num_channels, |
| bit_width, stream_id, |
| use_default_chmap, |
| chmap, |
| sample_word_size, |
| ASM_LITTLE_ENDIAN, |
| DEFAULT_QF); |
| if (ret < 0) |
| pr_err("%s: CMD Format block failed\n", __func__); |
| |
| break; |
| case FORMAT_MP3: |
| pr_debug("SND_AUDIOCODEC_MP3\n"); |
| /* no media format block needed */ |
| break; |
| case FORMAT_MPEG4_AAC: |
| pr_debug("SND_AUDIOCODEC_AAC\n"); |
| memset(&aac_cfg, 0x0, sizeof(struct asm_aac_cfg)); |
| aac_cfg.aot = AAC_ENC_MODE_EAAC_P; |
| if (prtd->codec_param.codec.format == |
| SND_AUDIOSTREAMFORMAT_MP4ADTS) |
| aac_cfg.format = 0x0; |
| else if (prtd->codec_param.codec.format == |
| SND_AUDIOSTREAMFORMAT_MP4LATM) |
| aac_cfg.format = 0x04; |
| else |
| aac_cfg.format = 0x03; |
| aac_cfg.ch_cfg = prtd->num_channels; |
| aac_cfg.sample_rate = prtd->sample_rate; |
| ret = q6asm_stream_media_format_block_aac(prtd->audio_client, |
| &aac_cfg, stream_id); |
| if (ret < 0) |
| pr_err("%s: CMD Format block failed\n", __func__); |
| break; |
| case FORMAT_AC3: |
| pr_debug("SND_AUDIOCODEC_AC3\n"); |
| break; |
| case FORMAT_EAC3: |
| pr_debug("SND_AUDIOCODEC_EAC3\n"); |
| break; |
| case FORMAT_WMA_V9: |
| pr_debug("SND_AUDIOCODEC_WMA\n"); |
| memset(&wma_cfg, 0x0, sizeof(struct asm_wma_cfg)); |
| wma_cfg.format_tag = prtd->codec_param.codec.format; |
| wma_cfg.ch_cfg = prtd->codec_param.codec.ch_in; |
| wma_cfg.sample_rate = prtd->sample_rate; |
| wma_cfg.avg_bytes_per_sec = codec_options->wma.avg_bit_rate/8; |
| wma_cfg.block_align = codec_options->wma.super_block_align; |
| wma_cfg.valid_bits_per_sample = |
| codec_options->wma.bits_per_sample; |
| wma_cfg.ch_mask = codec_options->wma.channelmask; |
| wma_cfg.encode_opt = codec_options->wma.encodeopt; |
| ret = q6asm_media_format_block_wma(prtd->audio_client, |
| &wma_cfg, stream_id); |
| if (ret < 0) |
| pr_err("%s: CMD Format block failed\n", __func__); |
| break; |
| case FORMAT_WMA_V10PRO: |
| pr_debug("SND_AUDIOCODEC_WMA_PRO\n"); |
| memset(&wma_pro_cfg, 0x0, sizeof(struct asm_wmapro_cfg)); |
| wma_pro_cfg.format_tag = prtd->codec_param.codec.format; |
| wma_pro_cfg.ch_cfg = prtd->codec_param.codec.ch_in; |
| wma_pro_cfg.sample_rate = prtd->sample_rate; |
| wma_cfg.avg_bytes_per_sec = codec_options->wma.avg_bit_rate/8; |
| wma_pro_cfg.block_align = codec_options->wma.super_block_align; |
| wma_pro_cfg.valid_bits_per_sample = |
| codec_options->wma.bits_per_sample; |
| wma_pro_cfg.ch_mask = codec_options->wma.channelmask; |
| wma_pro_cfg.encode_opt = codec_options->wma.encodeopt; |
| wma_pro_cfg.adv_encode_opt = codec_options->wma.encodeopt1; |
| wma_pro_cfg.adv_encode_opt2 = codec_options->wma.encodeopt2; |
| ret = q6asm_media_format_block_wmapro(prtd->audio_client, |
| &wma_pro_cfg, stream_id); |
| if (ret < 0) |
| pr_err("%s: CMD Format block failed\n", __func__); |
| break; |
| case FORMAT_MP2: |
| pr_debug("%s: SND_AUDIOCODEC_MP2\n", __func__); |
| break; |
| case FORMAT_FLAC: |
| pr_debug("%s: SND_AUDIOCODEC_FLAC\n", __func__); |
| memset(&flac_cfg, 0x0, sizeof(struct asm_flac_cfg)); |
| flac_cfg.ch_cfg = prtd->num_channels; |
| flac_cfg.sample_rate = prtd->sample_rate; |
| flac_cfg.stream_info_present = 1; |
| flac_cfg.sample_size = codec_options->flac_dec.sample_size; |
| flac_cfg.min_blk_size = codec_options->flac_dec.min_blk_size; |
| flac_cfg.max_blk_size = codec_options->flac_dec.max_blk_size; |
| flac_cfg.max_frame_size = |
| codec_options->flac_dec.max_frame_size; |
| flac_cfg.min_frame_size = |
| codec_options->flac_dec.min_frame_size; |
| |
| ret = q6asm_stream_media_format_block_flac(prtd->audio_client, |
| &flac_cfg, stream_id); |
| if (ret < 0) |
| pr_err("%s: CMD Format block failed ret %d\n", |
| __func__, ret); |
| |
| break; |
| case FORMAT_VORBIS: |
| pr_debug("%s: SND_AUDIOCODEC_VORBIS\n", __func__); |
| memset(&vorbis_cfg, 0x0, sizeof(struct asm_vorbis_cfg)); |
| vorbis_cfg.bit_stream_fmt = |
| codec_options->vorbis_dec.bit_stream_fmt; |
| |
| ret = q6asm_stream_media_format_block_vorbis( |
| prtd->audio_client, &vorbis_cfg, |
| stream_id); |
| if (ret < 0) |
| pr_err("%s: CMD Format block failed ret %d\n", |
| __func__, ret); |
| |
| break; |
| case FORMAT_ALAC: |
| pr_debug("%s: SND_AUDIOCODEC_ALAC\n", __func__); |
| memset(&alac_cfg, 0x0, sizeof(struct asm_alac_cfg)); |
| alac_cfg.num_channels = prtd->num_channels; |
| alac_cfg.sample_rate = prtd->sample_rate; |
| alac_cfg.frame_length = codec_options->alac.frame_length; |
| alac_cfg.compatible_version = |
| codec_options->alac.compatible_version; |
| alac_cfg.bit_depth = codec_options->alac.bit_depth; |
| alac_cfg.pb = codec_options->alac.pb; |
| alac_cfg.mb = codec_options->alac.mb; |
| alac_cfg.kb = codec_options->alac.kb; |
| alac_cfg.max_run = codec_options->alac.max_run; |
| alac_cfg.max_frame_bytes = codec_options->alac.max_frame_bytes; |
| alac_cfg.avg_bit_rate = codec_options->alac.avg_bit_rate; |
| alac_cfg.channel_layout_tag = |
| codec_options->alac.channel_layout_tag; |
| |
| ret = q6asm_media_format_block_alac(prtd->audio_client, |
| &alac_cfg, stream_id); |
| if (ret < 0) |
| pr_err("%s: CMD Format block failed ret %d\n", |
| __func__, ret); |
| break; |
| case FORMAT_APE: |
| pr_debug("%s: SND_AUDIOCODEC_APE\n", __func__); |
| memset(&ape_cfg, 0x0, sizeof(struct asm_ape_cfg)); |
| ape_cfg.num_channels = prtd->num_channels; |
| ape_cfg.sample_rate = prtd->sample_rate; |
| ape_cfg.compatible_version = |
| codec_options->ape.compatible_version; |
| ape_cfg.compression_level = |
| codec_options->ape.compression_level; |
| ape_cfg.format_flags = codec_options->ape.format_flags; |
| ape_cfg.blocks_per_frame = codec_options->ape.blocks_per_frame; |
| ape_cfg.final_frame_blocks = |
| codec_options->ape.final_frame_blocks; |
| ape_cfg.total_frames = codec_options->ape.total_frames; |
| ape_cfg.bits_per_sample = codec_options->ape.bits_per_sample; |
| ape_cfg.seek_table_present = |
| codec_options->ape.seek_table_present; |
| |
| ret = q6asm_media_format_block_ape(prtd->audio_client, |
| &ape_cfg, stream_id); |
| |
| if (ret < 0) |
| pr_err("%s: CMD Format block failed ret %d\n", |
| __func__, ret); |
| break; |
| case FORMAT_DTS: |
| pr_debug("SND_AUDIOCODEC_DTS\n"); |
| /* no media format block needed */ |
| break; |
| case FORMAT_DSD: |
| pr_debug("%s: SND_AUDIOCODEC_DSD\n", __func__); |
| memset(&dsd_cfg, 0x0, sizeof(struct asm_dsd_cfg)); |
| dsd_cfg.num_channels = prtd->num_channels; |
| dsd_cfg.dsd_data_rate = prtd->sample_rate; |
| dsd_cfg.num_version = 0; |
| dsd_cfg.is_bitwise_big_endian = 1; |
| dsd_cfg.dsd_channel_block_size = 1; |
| ret = q6asm_media_format_block_dsd(prtd->audio_client, |
| &dsd_cfg, stream_id); |
| if (ret < 0) |
| pr_err("%s: CMD DSD Format block failed ret %d\n", |
| __func__, ret); |
| break; |
| case FORMAT_TRUEHD: |
| pr_debug("SND_AUDIOCODEC_TRUEHD\n"); |
| /* no media format block needed */ |
| break; |
| case FORMAT_IEC61937: |
| pr_debug("SND_AUDIOCODEC_IEC61937\n"); |
| ret = q6asm_media_format_block_iec(prtd->audio_client, |
| prtd->sample_rate, |
| prtd->num_channels); |
| if (ret < 0) |
| pr_err("%s: CMD IEC61937 Format block failed ret %d\n", |
| __func__, ret); |
| break; |
| case FORMAT_APTX: |
| pr_debug("SND_AUDIOCODEC_APTX\n"); |
| memset(&aptx_cfg, 0x0, sizeof(struct aptx_dec_bt_addr_cfg)); |
| ret = q6asm_stream_media_format_block_aptx_dec( |
| prtd->audio_client, |
| prtd->sample_rate, |
| stream_id); |
| if (ret >= 0) { |
| aptx_cfg.nap = codec_options->aptx_dec.nap; |
| aptx_cfg.uap = codec_options->aptx_dec.uap; |
| aptx_cfg.lap = codec_options->aptx_dec.lap; |
| q6asm_set_aptx_dec_bt_addr(prtd->audio_client, |
| &aptx_cfg); |
| } else { |
| pr_err("%s: CMD Format block failed ret %d\n", |
| __func__, ret); |
| } |
| break; |
| default: |
| pr_debug("%s, unsupported format, skip", __func__); |
| break; |
| } |
| return ret; |
| } |
| |
| static int msm_compr_init_pp_params(struct snd_compr_stream *cstream, |
| struct audio_client *ac) |
| { |
| int ret = 0; |
| struct asm_softvolume_params softvol = { |
| .period = SOFT_VOLUME_PERIOD, |
| .step = SOFT_VOLUME_STEP, |
| .rampingcurve = SOFT_VOLUME_CURVE_LINEAR, |
| }; |
| |
| switch (ac->topology) { |
| default: |
| ret = q6asm_set_softvolume_v2(ac, &softvol, |
| SOFT_VOLUME_INSTANCE_1); |
| if (ret < 0) |
| pr_err("%s: Send SoftVolume Param failed ret=%d\n", |
| __func__, ret); |
| |
| break; |
| } |
| return ret; |
| } |
| |
| static int msm_compr_configure_dsp_for_playback |
| (struct snd_compr_stream *cstream) |
| { |
| struct snd_compr_runtime *runtime = cstream->runtime; |
| struct msm_compr_audio *prtd = runtime->private_data; |
| struct snd_soc_pcm_runtime *soc_prtd = cstream->private_data; |
| uint16_t bits_per_sample = 16; |
| int dir = IN, ret = 0; |
| struct audio_client *ac = prtd->audio_client; |
| uint32_t stream_index; |
| struct asm_softpause_params softpause = { |
| .enable = SOFT_PAUSE_ENABLE, |
| .period = SOFT_PAUSE_PERIOD, |
| .step = SOFT_PAUSE_STEP, |
| .rampingcurve = SOFT_PAUSE_CURVE_LINEAR, |
| }; |
| struct asm_softvolume_params softvol = { |
| .period = SOFT_VOLUME_PERIOD, |
| .step = SOFT_VOLUME_STEP, |
| .rampingcurve = SOFT_VOLUME_CURVE_LINEAR, |
| }; |
| struct snd_kcontrol *kctl; |
| struct snd_ctl_elem_value kctl_elem_value; |
| uint16_t target_asm_bit_width = 0; |
| |
| pr_debug("%s: stream_id %d\n", __func__, ac->stream_id); |
| stream_index = STREAM_ARRAY_INDEX(ac->stream_id); |
| if (stream_index >= MAX_NUMBER_OF_STREAMS || stream_index < 0) { |
| pr_err("%s: Invalid stream index:%d", __func__, stream_index); |
| return -EINVAL; |
| } |
| |
| kctl = snd_soc_card_get_kcontrol(soc_prtd->card, |
| DSP_BIT_WIDTH_MIXER_CTL); |
| if (kctl) { |
| kctl->get(kctl, &kctl_elem_value); |
| target_asm_bit_width = kctl_elem_value.value.integer.value[0]; |
| if (target_asm_bit_width > 0) { |
| pr_debug("%s enforce ASM bitwidth to %d from %d\n", |
| __func__, |
| target_asm_bit_width, |
| bits_per_sample); |
| bits_per_sample = target_asm_bit_width; |
| } |
| } else { |
| pr_info("%s: failed to get mixer ctl for %s.\n", |
| __func__, DSP_BIT_WIDTH_MIXER_CTL); |
| } |
| |
| if ((prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S24_LE) || |
| (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S24_3LE)) |
| bits_per_sample = 24; |
| else if (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S32_LE) |
| bits_per_sample = 32; |
| |
| if (prtd->compr_passthr != LEGACY_PCM) { |
| ret = q6asm_open_write_compressed(ac, prtd->codec, |
| prtd->compr_passthr); |
| if (ret < 0) { |
| pr_err("%s:ASM open write err[%d] for compr_type[%d]\n", |
| __func__, ret, prtd->compr_passthr); |
| return ret; |
| } |
| prtd->gapless_state.stream_opened[stream_index] = 1; |
| |
| ret = msm_pcm_routing_reg_phy_compr_stream( |
| soc_prtd->dai_link->id, |
| ac->perf_mode, |
| prtd->session_id, |
| SNDRV_PCM_STREAM_PLAYBACK, |
| prtd->compr_passthr); |
| if (ret) { |
| pr_err("%s: compr stream reg failed:%d\n", __func__, |
| ret); |
| return ret; |
| } |
| } else { |
| pr_debug("%s: stream_id %d bits_per_sample %d\n", |
| __func__, ac->stream_id, bits_per_sample); |
| ret = q6asm_stream_open_write_v4(ac, |
| prtd->codec, bits_per_sample, |
| ac->stream_id, |
| prtd->gapless_state.use_dsp_gapless_mode); |
| if (ret < 0) { |
| pr_err("%s:ASM open write err[%d] for compr type[%d]\n", |
| __func__, ret, prtd->compr_passthr); |
| return -ENOMEM; |
| } |
| prtd->gapless_state.stream_opened[stream_index] = 1; |
| |
| pr_debug("%s: BE id %d\n", __func__, soc_prtd->dai_link->id); |
| ret = msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->id, |
| ac->perf_mode, |
| prtd->session_id, |
| SNDRV_PCM_STREAM_PLAYBACK); |
| if (ret) { |
| pr_err("%s: stream reg failed:%d\n", __func__, ret); |
| return ret; |
| } |
| } |
| |
| ret = msm_compr_set_volume(cstream, 0, 0); |
| if (ret < 0) |
| pr_err("%s : Set Volume failed : %d", __func__, ret); |
| |
| if (prtd->compr_passthr != LEGACY_PCM) { |
| pr_debug("%s : Don't send cal and PP params for compress path", |
| __func__); |
| } else { |
| ret = q6asm_send_cal(ac); |
| if (ret < 0) |
| pr_debug("%s : Send cal failed : %d", __func__, ret); |
| |
| ret = q6asm_set_softpause(ac, &softpause); |
| if (ret < 0) |
| pr_err("%s: Send SoftPause Param failed ret=%d\n", |
| __func__, ret); |
| |
| ret = q6asm_set_softvolume(ac, &softvol); |
| if (ret < 0) |
| pr_err("%s: Send SoftVolume Param failed ret=%d\n", |
| __func__, ret); |
| } |
| ret = q6asm_set_io_mode(ac, (COMPRESSED_STREAM_IO | ASYNC_IO_MODE)); |
| if (ret < 0) { |
| pr_err("%s: Set IO mode failed\n", __func__); |
| return -EINVAL; |
| } |
| |
| runtime->fragments = prtd->codec_param.buffer.fragments; |
| runtime->fragment_size = prtd->codec_param.buffer.fragment_size; |
| pr_debug("allocate %d buffers each of size %d\n", |
| runtime->fragments, |
| runtime->fragment_size); |
| ret = q6asm_audio_client_buf_alloc_contiguous(dir, ac, |
| runtime->fragment_size, |
| runtime->fragments); |
| if (ret < 0) { |
| pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret); |
| return -ENOMEM; |
| } |
| |
| prtd->byte_offset = 0; |
| prtd->copied_total = 0; |
| prtd->app_pointer = 0; |
| prtd->bytes_received = 0; |
| prtd->bytes_sent = 0; |
| prtd->buffer = ac->port[dir].buf[0].data; |
| prtd->buffer_paddr = ac->port[dir].buf[0].phys; |
| prtd->buffer_size = runtime->fragments * runtime->fragment_size; |
| |
| /* Bit-0 of flags represent timestamp mode */ |
| if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG) |
| prtd->ts_header_offset = sizeof(struct snd_codec_metadata); |
| else |
| prtd->ts_header_offset = 0; |
| |
| ret = msm_compr_send_media_format_block(cstream, ac->stream_id, false); |
| if (ret < 0) |
| pr_err("%s, failed to send media format block\n", __func__); |
| |
| return ret; |
| } |
| |
| static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream) |
| { |
| struct snd_compr_runtime *runtime = cstream->runtime; |
| struct msm_compr_audio *prtd = runtime->private_data; |
| struct snd_soc_pcm_runtime *soc_prtd = cstream->private_data; |
| uint16_t bits_per_sample; |
| uint16_t sample_word_size; |
| int dir = OUT, ret = 0; |
| struct audio_client *ac = prtd->audio_client; |
| uint32_t stream_index; |
| |
| switch (prtd->codec_param.codec.format) { |
| case SNDRV_PCM_FORMAT_S24_LE: |
| bits_per_sample = 24; |
| sample_word_size = 32; |
| break; |
| case SNDRV_PCM_FORMAT_S24_3LE: |
| bits_per_sample = 24; |
| sample_word_size = 24; |
| break; |
| case SNDRV_PCM_FORMAT_S32_LE: |
| bits_per_sample = 32; |
| sample_word_size = 32; |
| break; |
| case SNDRV_PCM_FORMAT_S16_LE: |
| default: |
| bits_per_sample = 16; |
| sample_word_size = 16; |
| break; |
| } |
| |
| pr_debug("%s: stream_id %d bits_per_sample %d\n", |
| __func__, ac->stream_id, bits_per_sample); |
| |
| if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG) { |
| ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM, |
| bits_per_sample, true); |
| } else { |
| ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM, |
| bits_per_sample, false); |
| } |
| if (ret < 0) { |
| pr_err("%s: q6asm_open_read failed:%d\n", __func__, ret); |
| return ret; |
| } |
| |
| ret = msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->id, |
| ac->perf_mode, |
| prtd->session_id, |
| SNDRV_PCM_STREAM_CAPTURE); |
| if (ret) { |
| pr_err("%s: stream reg failed:%d\n", __func__, ret); |
| return ret; |
| } |
| |
| ret = q6asm_set_io_mode(ac, (COMPRESSED_STREAM_IO | ASYNC_IO_MODE)); |
| if (ret < 0) { |
| pr_err("%s: Set IO mode failed\n", __func__); |
| return -EINVAL; |
| } |
| |
| stream_index = STREAM_ARRAY_INDEX(ac->stream_id); |
| if (stream_index >= MAX_NUMBER_OF_STREAMS || stream_index < 0) { |
| pr_err("%s: Invalid stream index:%d", __func__, stream_index); |
| return -EINVAL; |
| } |
| |
| runtime->fragments = prtd->codec_param.buffer.fragments; |
| runtime->fragment_size = prtd->codec_param.buffer.fragment_size; |
| pr_debug("%s: allocate %d buffers each of size %d\n", |
| __func__, runtime->fragments, |
| runtime->fragment_size); |
| ret = q6asm_audio_client_buf_alloc_contiguous(dir, ac, |
| runtime->fragment_size, |
| runtime->fragments); |
| if (ret < 0) { |
| pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret); |
| return -ENOMEM; |
| } |
| |
| prtd->byte_offset = 0; |
| prtd->received_total = 0; |
| prtd->app_pointer = 0; |
| prtd->bytes_copied = 0; |
| prtd->bytes_read = 0; |
| prtd->bytes_read_offset = 0; |
| prtd->buffer = ac->port[dir].buf[0].data; |
| prtd->buffer_paddr = ac->port[dir].buf[0].phys; |
| prtd->buffer_size = runtime->fragments * runtime->fragment_size; |
| |
| /* Bit-0 of flags represent timestamp mode */ |
| if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG) |
| prtd->ts_header_offset = sizeof(struct snd_codec_metadata); |
| else |
| prtd->ts_header_offset = 0; |
| |
| pr_debug("%s: sample_rate = %d channels = %d bps = %d sample_word_size = %d\n", |
| __func__, prtd->sample_rate, prtd->num_channels, |
| bits_per_sample, sample_word_size); |
| ret = q6asm_enc_cfg_blk_pcm_format_support_v3(prtd->audio_client, |
| prtd->sample_rate, prtd->num_channels, |
| bits_per_sample, sample_word_size); |
| |
| return ret; |
| } |
| |
| static int msm_compr_playback_open(struct snd_compr_stream *cstream) |
| { |
| struct snd_compr_runtime *runtime = cstream->runtime; |
| struct snd_soc_pcm_runtime *rtd = cstream->private_data; |
| struct msm_compr_audio *prtd; |
| struct msm_compr_pdata *pdata = |
| snd_soc_platform_get_drvdata(rtd->platform); |
| |
| pr_debug("%s\n", __func__); |
| prtd = kzalloc(sizeof(struct msm_compr_audio), GFP_KERNEL); |
| if (prtd == NULL) { |
| pr_err("Failed to allocate memory for msm_compr_audio\n"); |
| return -ENOMEM; |
| } |
| |
| runtime->private_data = NULL; |
| prtd->cstream = cstream; |
| pdata->cstream[rtd->dai_link->id] = cstream; |
| pdata->audio_effects[rtd->dai_link->id] = |
| kzalloc(sizeof(struct msm_compr_audio_effects), GFP_KERNEL); |
| if (!pdata->audio_effects[rtd->dai_link->id]) { |
| pr_err("%s: Could not allocate memory for effects\n", __func__); |
| pdata->cstream[rtd->dai_link->id] = NULL; |
| kfree(prtd); |
| return -ENOMEM; |
| } |
| pdata->dec_params[rtd->dai_link->id] = |
| kzalloc(sizeof(struct msm_compr_dec_params), GFP_KERNEL); |
| if (!pdata->dec_params[rtd->dai_link->id]) { |
| pr_err("%s: Could not allocate memory for dec params\n", |
| __func__); |
| kfree(pdata->audio_effects[rtd->dai_link->id]); |
| pdata->cstream[rtd->dai_link->id] = NULL; |
| kfree(prtd); |
| return -ENOMEM; |
| } |
| prtd->codec = FORMAT_MP3; |
| prtd->bytes_received = 0; |
| prtd->bytes_sent = 0; |
| prtd->copied_total = 0; |
| prtd->byte_offset = 0; |
| prtd->sample_rate = 44100; |
| prtd->num_channels = 2; |
| prtd->drain_ready = 0; |
| prtd->last_buffer = 0; |
| prtd->first_buffer = 1; |
| prtd->partial_drain_delay = 0; |
| prtd->next_stream = 0; |
| memset(&prtd->gapless_state, 0, sizeof(struct msm_compr_gapless_state)); |
| /* |
| * Update the use_dsp_gapless_mode from gapless struture with the value |
| * part of platform data. |
| */ |
| prtd->gapless_state.use_dsp_gapless_mode = pdata->use_dsp_gapless_mode; |
| |
| pr_debug("%s: gapless mode %d", __func__, pdata->use_dsp_gapless_mode); |
| |
| spin_lock_init(&prtd->lock); |
| |
| atomic_set(&prtd->eos, 0); |
| atomic_set(&prtd->start, 0); |
| atomic_set(&prtd->drain, 0); |
| atomic_set(&prtd->xrun, 0); |
| atomic_set(&prtd->close, 0); |
| atomic_set(&prtd->wait_on_close, 0); |
| atomic_set(&prtd->error, 0); |
| |
| init_waitqueue_head(&prtd->eos_wait); |
| init_waitqueue_head(&prtd->drain_wait); |
| init_waitqueue_head(&prtd->close_wait); |
| init_waitqueue_head(&prtd->wait_for_stream_avail); |
| |
| runtime->private_data = prtd; |
| populate_codec_list(prtd); |
| prtd->audio_client = q6asm_audio_client_alloc( |
| (app_cb)compr_event_handler, prtd); |
| if (!prtd->audio_client) { |
| pr_err("%s: Could not allocate memory for client\n", __func__); |
| kfree(pdata->audio_effects[rtd->dai_link->id]); |
| kfree(pdata->dec_params[rtd->dai_link->id]); |
| pdata->cstream[rtd->dai_link->id] = NULL; |
| runtime->private_data = NULL; |
| kfree(prtd); |
| return -ENOMEM; |
| } |
| pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session); |
| prtd->audio_client->perf_mode = false; |
| prtd->session_id = prtd->audio_client->session; |
| msm_adsp_init_mixer_ctl_pp_event_queue(rtd); |
| |
| return 0; |
| } |
| |
| static int msm_compr_capture_open(struct snd_compr_stream *cstream) |
| { |
| struct snd_compr_runtime *runtime = cstream->runtime; |
| struct snd_soc_pcm_runtime *rtd = cstream->private_data; |
| struct msm_compr_audio *prtd; |
| struct msm_compr_pdata *pdata = |
| snd_soc_platform_get_drvdata(rtd->platform); |
| |
| pr_debug("%s\n", __func__); |
| prtd = kzalloc(sizeof(struct msm_compr_audio), GFP_KERNEL); |
| if (prtd == NULL) { |
| pr_err("Failed to allocate memory for msm_compr_audio\n"); |
| return -ENOMEM; |
| } |
| |
| runtime->private_data = NULL; |
| prtd->cstream = cstream; |
| pdata->cstream[rtd->dai_link->id] = cstream; |
| |
| prtd->audio_client = q6asm_audio_client_alloc( |
| (app_cb)compr_event_handler, prtd); |
| if (!prtd->audio_client) { |
| pr_err("%s: Could not allocate memory for client\n", __func__); |
| pdata->cstream[rtd->dai_link->id] = NULL; |
| kfree(prtd); |
| return -ENOMEM; |
| } |
| pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session); |
| prtd->audio_client->perf_mode = false; |
| prtd->session_id = prtd->audio_client->session; |
| prtd->codec = FORMAT_LINEAR_PCM; |
| prtd->bytes_copied = 0; |
| prtd->bytes_read = 0; |
| prtd->bytes_read_offset = 0; |
| prtd->received_total = 0; |
| prtd->byte_offset = 0; |
| prtd->sample_rate = 48000; |
| prtd->num_channels = 2; |
| prtd->first_buffer = 0; |
| |
| spin_lock_init(&prtd->lock); |
| |
| atomic_set(&prtd->eos, 0); |
| atomic_set(&prtd->start, 0); |
| atomic_set(&prtd->drain, 0); |
| atomic_set(&prtd->xrun, 0); |
| atomic_set(&prtd->close, 0); |
| atomic_set(&prtd->wait_on_close, 0); |
| atomic_set(&prtd->error, 0); |
| |
| runtime->private_data = prtd; |
| |
| return 0; |
| } |
| |
| static int msm_compr_open(struct snd_compr_stream *cstream) |
| { |
| int ret = 0; |
| |
| if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| ret = msm_compr_playback_open(cstream); |
| else if (cstream->direction == SND_COMPRESS_CAPTURE) |
| ret = msm_compr_capture_open(cstream); |
| return ret; |
| } |
| |
| static int msm_compr_playback_free(struct snd_compr_stream *cstream) |
| { |
| struct snd_compr_runtime *runtime; |
| struct msm_compr_audio *prtd; |
| struct snd_soc_pcm_runtime *soc_prtd; |
| struct msm_compr_pdata *pdata; |
| struct audio_client *ac; |
| int dir = IN, ret = 0, stream_id; |
| unsigned long flags; |
| uint32_t stream_index; |
| |
| pr_debug("%s\n", __func__); |
| |
| if (!cstream) { |
| pr_err("%s cstream is null\n", __func__); |
| return 0; |
| } |
| runtime = cstream->runtime; |
| soc_prtd = cstream->private_data; |
| if (!runtime || !soc_prtd || !(soc_prtd->platform)) { |
| pr_err("%s runtime or soc_prtd or platform is null\n", |
| __func__); |
| return 0; |
| } |
| prtd = runtime->private_data; |
| if (!prtd) { |
| pr_err("%s prtd is null\n", __func__); |
| return 0; |
| } |
| prtd->cmd_interrupt = 1; |
| wake_up(&prtd->drain_wait); |
| pdata = snd_soc_platform_get_drvdata(soc_prtd->platform); |
| ac = prtd->audio_client; |
| if (!pdata || !ac) { |
| pr_err("%s pdata or ac is null\n", __func__); |
| return 0; |
| } |
| if (atomic_read(&prtd->eos)) { |
| ret = wait_event_timeout(prtd->eos_wait, |
| prtd->eos_ack, 5 * HZ); |
| if (!ret) |
| pr_err("%s: CMD_EOS failed\n", __func__); |
| } |
| if (atomic_read(&prtd->close)) { |
| prtd->cmd_ack = 0; |
| atomic_set(&prtd->wait_on_close, 1); |
| ret = wait_event_timeout(prtd->close_wait, |
| prtd->cmd_ack, 5 * HZ); |
| if (!ret) |
| pr_err("%s: CMD_CLOSE failed\n", __func__); |
| } |
| |
| spin_lock_irqsave(&prtd->lock, flags); |
| stream_id = ac->stream_id; |
| stream_index = STREAM_ARRAY_INDEX(NEXT_STREAM_ID(stream_id)); |
| |
| if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0) && |
| (prtd->gapless_state.stream_opened[stream_index])) { |
| prtd->gapless_state.stream_opened[stream_index] = 0; |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| pr_debug(" close stream %d", NEXT_STREAM_ID(stream_id)); |
| q6asm_stream_cmd(ac, CMD_CLOSE, NEXT_STREAM_ID(stream_id)); |
| spin_lock_irqsave(&prtd->lock, flags); |
| } |
| |
| stream_index = STREAM_ARRAY_INDEX(stream_id); |
| if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0) && |
| (prtd->gapless_state.stream_opened[stream_index])) { |
| prtd->gapless_state.stream_opened[stream_index] = 0; |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| pr_debug("close stream %d", stream_id); |
| q6asm_stream_cmd(ac, CMD_CLOSE, stream_id); |
| spin_lock_irqsave(&prtd->lock, flags); |
| } |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| |
| pdata->cstream[soc_prtd->dai_link->id] = NULL; |
| if (cstream->direction == SND_COMPRESS_PLAYBACK) { |
| msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->id, |
| SNDRV_PCM_STREAM_PLAYBACK); |
| } |
| |
| q6asm_audio_client_buf_free_contiguous(dir, ac); |
| |
| q6asm_audio_client_free(ac); |
| msm_adsp_clean_mixer_ctl_pp_event_queue(soc_prtd); |
| kfree(pdata->audio_effects[soc_prtd->dai_link->id]); |
| pdata->audio_effects[soc_prtd->dai_link->id] = NULL; |
| kfree(pdata->dec_params[soc_prtd->dai_link->id]); |
| pdata->dec_params[soc_prtd->dai_link->id] = NULL; |
| kfree(prtd); |
| runtime->private_data = NULL; |
| |
| return 0; |
| } |
| |
| static int msm_compr_capture_free(struct snd_compr_stream *cstream) |
| { |
| struct snd_compr_runtime *runtime; |
| struct msm_compr_audio *prtd; |
| struct snd_soc_pcm_runtime *soc_prtd; |
| struct msm_compr_pdata *pdata; |
| struct audio_client *ac; |
| int dir = OUT, stream_id; |
| unsigned long flags; |
| uint32_t stream_index; |
| |
| if (!cstream) { |
| pr_err("%s cstream is null\n", __func__); |
| return 0; |
| } |
| runtime = cstream->runtime; |
| soc_prtd = cstream->private_data; |
| if (!runtime || !soc_prtd || !(soc_prtd->platform)) { |
| pr_err("%s runtime or soc_prtd or platform is null\n", |
| __func__); |
| return 0; |
| } |
| prtd = runtime->private_data; |
| if (!prtd) { |
| pr_err("%s prtd is null\n", __func__); |
| return 0; |
| } |
| pdata = snd_soc_platform_get_drvdata(soc_prtd->platform); |
| ac = prtd->audio_client; |
| if (!pdata || !ac) { |
| pr_err("%s pdata or ac is null\n", __func__); |
| return 0; |
| } |
| |
| spin_lock_irqsave(&prtd->lock, flags); |
| stream_id = ac->stream_id; |
| |
| stream_index = STREAM_ARRAY_INDEX(stream_id); |
| if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0)) { |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| pr_debug("close stream %d", stream_id); |
| q6asm_stream_cmd(ac, CMD_CLOSE, stream_id); |
| spin_lock_irqsave(&prtd->lock, flags); |
| } |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| |
| pdata->cstream[soc_prtd->dai_link->id] = NULL; |
| msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->id, |
| SNDRV_PCM_STREAM_CAPTURE); |
| |
| q6asm_audio_client_buf_free_contiguous(dir, ac); |
| |
| q6asm_audio_client_free(ac); |
| |
| kfree(prtd); |
| runtime->private_data = NULL; |
| |
| return 0; |
| } |
| |
| static int msm_compr_free(struct snd_compr_stream *cstream) |
| { |
| int ret = 0; |
| |
| if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| ret = msm_compr_playback_free(cstream); |
| else if (cstream->direction == SND_COMPRESS_CAPTURE) |
| ret = msm_compr_capture_free(cstream); |
| return ret; |
| } |
| |
| static bool msm_compr_validate_codec_compr(__u32 codec_id) |
| { |
| int32_t i; |
| |
| for (i = 0; i < ARRAY_SIZE(compr_codecs); i++) { |
| if (compr_codecs[i] == codec_id) |
| return true; |
| } |
| return false; |
| } |
| |
| /* compress stream operations */ |
| static int msm_compr_set_params(struct snd_compr_stream *cstream, |
| struct snd_compr_params *params) |
| { |
| struct snd_compr_runtime *runtime = cstream->runtime; |
| struct msm_compr_audio *prtd = runtime->private_data; |
| int ret = 0, frame_sz = 0; |
| int i, num_rates; |
| bool is_format_gapless = false; |
| |
| pr_debug("%s\n", __func__); |
| |
| num_rates = sizeof(supported_sample_rates)/sizeof(unsigned int); |
| for (i = 0; i < num_rates; i++) |
| if (params->codec.sample_rate == supported_sample_rates[i]) |
| break; |
| if (i == num_rates) |
| return -EINVAL; |
| |
| memcpy(&prtd->codec_param, params, sizeof(struct snd_compr_params)); |
| /* ToDo: remove duplicates */ |
| prtd->num_channels = prtd->codec_param.codec.ch_in; |
| prtd->sample_rate = prtd->codec_param.codec.sample_rate; |
| pr_debug("%s: sample_rate %d\n", __func__, prtd->sample_rate); |
| |
| if ((prtd->codec_param.codec.compr_passthr >= LEGACY_PCM && |
| prtd->codec_param. |
| codec.compr_passthr <= COMPRESSED_PASSTHROUGH_DSD) || |
| (prtd->codec_param. |
| codec.compr_passthr == COMPRESSED_PASSTHROUGH_IEC61937)) |
| prtd->compr_passthr = prtd->codec_param.codec.compr_passthr; |
| else |
| prtd->compr_passthr = LEGACY_PCM; |
| pr_debug("%s: compr_passthr = %d", __func__, prtd->compr_passthr); |
| if (prtd->compr_passthr != LEGACY_PCM) { |
| pr_debug("%s: Reset gapless mode playback for compr_type[%d]\n", |
| __func__, prtd->compr_passthr); |
| prtd->gapless_state.use_dsp_gapless_mode = 0; |
| if (!msm_compr_validate_codec_compr(params->codec.id)) { |
| pr_err("%s codec not supported in passthrough,id =%d\n", |
| __func__, params->codec.id); |
| return -EINVAL; |
| } |
| } |
| |
| switch (params->codec.id) { |
| case SND_AUDIOCODEC_PCM: { |
| pr_debug("SND_AUDIOCODEC_PCM\n"); |
| prtd->codec = FORMAT_LINEAR_PCM; |
| is_format_gapless = true; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_MP3: { |
| pr_debug("SND_AUDIOCODEC_MP3\n"); |
| prtd->codec = FORMAT_MP3; |
| frame_sz = MP3_OUTPUT_FRAME_SZ; |
| is_format_gapless = true; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_AAC: { |
| pr_debug("SND_AUDIOCODEC_AAC\n"); |
| prtd->codec = FORMAT_MPEG4_AAC; |
| frame_sz = AAC_OUTPUT_FRAME_SZ; |
| is_format_gapless = true; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_AC3: { |
| pr_debug("SND_AUDIOCODEC_AC3\n"); |
| prtd->codec = FORMAT_AC3; |
| frame_sz = AC3_OUTPUT_FRAME_SZ; |
| is_format_gapless = true; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_EAC3: { |
| pr_debug("SND_AUDIOCODEC_EAC3\n"); |
| prtd->codec = FORMAT_EAC3; |
| frame_sz = EAC3_OUTPUT_FRAME_SZ; |
| is_format_gapless = true; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_MP2: { |
| pr_debug("SND_AUDIOCODEC_MP2\n"); |
| prtd->codec = FORMAT_MP2; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_WMA: { |
| pr_debug("SND_AUDIOCODEC_WMA\n"); |
| prtd->codec = FORMAT_WMA_V9; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_WMA_PRO: { |
| pr_debug("SND_AUDIOCODEC_WMA_PRO\n"); |
| prtd->codec = FORMAT_WMA_V10PRO; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_FLAC: { |
| pr_debug("%s: SND_AUDIOCODEC_FLAC\n", __func__); |
| prtd->codec = FORMAT_FLAC; |
| /* |
| * DSP bufferring is based on blk size, |
| * consider mininum buffering to rule out any false wait |
| */ |
| frame_sz = |
| prtd->codec_param.codec.options.flac_dec.min_blk_size; |
| is_format_gapless = true; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_VORBIS: { |
| pr_debug("%s: SND_AUDIOCODEC_VORBIS\n", __func__); |
| prtd->codec = FORMAT_VORBIS; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_ALAC: { |
| pr_debug("%s: SND_AUDIOCODEC_ALAC\n", __func__); |
| prtd->codec = FORMAT_ALAC; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_APE: { |
| pr_debug("%s: SND_AUDIOCODEC_APE\n", __func__); |
| prtd->codec = FORMAT_APE; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_DTS: { |
| pr_debug("%s: SND_AUDIOCODEC_DTS\n", __func__); |
| prtd->codec = FORMAT_DTS; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_DSD: { |
| pr_debug("%s: SND_AUDIOCODEC_DSD\n", __func__); |
| prtd->codec = FORMAT_DSD; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_TRUEHD: { |
| pr_debug("%s: SND_AUDIOCODEC_TRUEHD\n", __func__); |
| prtd->codec = FORMAT_TRUEHD; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_IEC61937: { |
| pr_debug("%s: SND_AUDIOCODEC_IEC61937\n", __func__); |
| prtd->codec = FORMAT_IEC61937; |
| break; |
| } |
| |
| case SND_AUDIOCODEC_APTX: { |
| pr_debug("%s: SND_AUDIOCODEC_APTX\n", __func__); |
| prtd->codec = FORMAT_APTX; |
| break; |
| } |
| |
| default: |
| pr_err("codec not supported, id =%d\n", params->codec.id); |
| return -EINVAL; |
| } |
| |
| if (!is_format_gapless) |
| prtd->gapless_state.use_dsp_gapless_mode = false; |
| |
| prtd->partial_drain_delay = |
| msm_compr_get_partial_drain_delay(frame_sz, prtd->sample_rate); |
| |
| if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| ret = msm_compr_configure_dsp_for_playback(cstream); |
| else if (cstream->direction == SND_COMPRESS_CAPTURE) |
| ret = msm_compr_configure_dsp_for_capture(cstream); |
| |
| return ret; |
| } |
| |
| static int msm_compr_drain_buffer(struct msm_compr_audio *prtd, |
| unsigned long *flags) |
| { |
| int rc = 0; |
| |
| atomic_set(&prtd->drain, 1); |
| prtd->drain_ready = 0; |
| spin_unlock_irqrestore(&prtd->lock, *flags); |
| pr_debug("%s: wait for buffer to be drained\n", __func__); |
| rc = wait_event_interruptible(prtd->drain_wait, |
| prtd->drain_ready || |
| prtd->cmd_interrupt || |
| atomic_read(&prtd->xrun) || |
| atomic_read(&prtd->error)); |
| pr_debug("%s: out of buffer drain wait with ret %d\n", __func__, rc); |
| spin_lock_irqsave(&prtd->lock, *flags); |
| if (prtd->cmd_interrupt) { |
| pr_debug("%s: buffer drain interrupted by flush)\n", __func__); |
| rc = -EINTR; |
| prtd->cmd_interrupt = 0; |
| } |
| if (atomic_read(&prtd->error)) { |
| pr_err("%s: Got RESET EVENTS notification, return\n", |
| __func__); |
| rc = -ENETRESET; |
| } |
| return rc; |
| } |
| |
| static int msm_compr_wait_for_stream_avail(struct msm_compr_audio *prtd, |
| unsigned long *flags) |
| { |
| int rc = 0; |
| |
| pr_debug("next session is already in opened state\n"); |
| prtd->next_stream = 1; |
| prtd->cmd_interrupt = 0; |
| spin_unlock_irqrestore(&prtd->lock, *flags); |
| /* |
| * Wait for stream to be available, or the wait to be interrupted by |
| * commands like flush or till a timeout of one second. |
| */ |
| rc = wait_event_timeout(prtd->wait_for_stream_avail, |
| prtd->stream_available || prtd->cmd_interrupt, 1 * HZ); |
| pr_err("%s:prtd->stream_available %d, prtd->cmd_interrupt %d rc %d\n", |
| __func__, prtd->stream_available, prtd->cmd_interrupt, rc); |
| |
| spin_lock_irqsave(&prtd->lock, *flags); |
| if (rc == 0) { |
| pr_err("%s: wait_for_stream_avail timed out\n", |
| __func__); |
| rc = -ETIMEDOUT; |
| } else if (prtd->cmd_interrupt == 1) { |
| /* |
| * This scenario might not happen as we do not allow |
| * flush in transition state. |
| */ |
| pr_debug("%s: wait_for_stream_avail interrupted\n", __func__); |
| prtd->cmd_interrupt = 0; |
| prtd->stream_available = 0; |
| rc = -EINTR; |
| } else { |
| prtd->stream_available = 0; |
| rc = 0; |
| } |
| pr_debug("%s : rc = %d", __func__, rc); |
| return rc; |
| } |
| |
| static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd) |
| { |
| struct snd_compr_runtime *runtime = cstream->runtime; |
| struct msm_compr_audio *prtd = runtime->private_data; |
| struct snd_soc_pcm_runtime *rtd = cstream->private_data; |
| struct msm_compr_pdata *pdata = |
| snd_soc_platform_get_drvdata(rtd->platform); |
| uint32_t *volume = pdata->volume[rtd->dai_link->id]; |
| struct audio_client *ac = prtd->audio_client; |
| unsigned long fe_id = rtd->dai_link->id; |
| int rc = 0; |
| int bytes_to_write; |
| unsigned long flags; |
| int stream_id; |
| uint32_t stream_index; |
| uint16_t bits_per_sample = 16; |
| |
| spin_lock_irqsave(&prtd->lock, flags); |
| if (atomic_read(&prtd->error)) { |
| pr_err("%s Got RESET EVENTS notification, return immediately", |
| __func__); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| return 0; |
| } |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| |
| switch (cmd) { |
| case SNDRV_PCM_TRIGGER_START: |
| pr_debug("%s: SNDRV_PCM_TRIGGER_START\n", __func__); |
| atomic_set(&prtd->start, 1); |
| |
| /* |
| * compr_set_volume and compr_init_pp_params |
| * are used to configure ASM volume hence not |
| * needed for compress passthrough playback. |
| * |
| * compress passthrough volume is controlled in |
| * ADM by adm_send_compressed_device_mute() |
| */ |
| if (prtd->compr_passthr == LEGACY_PCM && |
| cstream->direction == SND_COMPRESS_PLAYBACK) { |
| /* set volume for the stream before RUN */ |
| rc = msm_compr_set_volume(cstream, |
| volume[0], volume[1]); |
| if (rc) |
| pr_err("%s : Set Volume failed : %d\n", |
| __func__, rc); |
| |
| rc = msm_compr_init_pp_params(cstream, ac); |
| if (rc) |
| pr_err("%s : init PP params failed : %d\n", |
| __func__, rc); |
| } else { |
| msm_compr_read_buffer(prtd); |
| } |
| /* issue RUN command for the stream */ |
| q6asm_run_nowait(prtd->audio_client, prtd->run_mode, |
| prtd->start_delay_msw, prtd->start_delay_lsw); |
| break; |
| case SNDRV_PCM_TRIGGER_STOP: |
| spin_lock_irqsave(&prtd->lock, flags); |
| pr_debug("%s: SNDRV_PCM_TRIGGER_STOP transition %d\n", __func__, |
| prtd->gapless_state.gapless_transition); |
| stream_id = ac->stream_id; |
| atomic_set(&prtd->start, 0); |
| if (cstream->direction == SND_COMPRESS_CAPTURE) { |
| q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE); |
| atomic_set(&prtd->xrun, 0); |
| prtd->received_total = 0; |
| prtd->bytes_copied = 0; |
| prtd->bytes_read = 0; |
| prtd->bytes_read_offset = 0; |
| prtd->byte_offset = 0; |
| prtd->app_pointer = 0; |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| } |
| if (prtd->next_stream) { |
| pr_debug("%s: interrupt next track wait queues\n", |
| __func__); |
| prtd->cmd_interrupt = 1; |
| wake_up(&prtd->wait_for_stream_avail); |
| prtd->next_stream = 0; |
| } |
| if (atomic_read(&prtd->eos)) { |
| pr_debug("%s: interrupt eos wait queues", __func__); |
| /* |
| * Gapless playback does not wait for eos, do not set |
| * cmd_int and do not wake up eos_wait during gapless |
| * transition |
| */ |
| if (!prtd->gapless_state.gapless_transition) { |
| prtd->cmd_interrupt = 1; |
| wake_up(&prtd->eos_wait); |
| } |
| atomic_set(&prtd->eos, 0); |
| } |
| if (atomic_read(&prtd->drain)) { |
| pr_debug("%s: interrupt drain wait queues", __func__); |
| prtd->cmd_interrupt = 1; |
| prtd->drain_ready = 1; |
| wake_up(&prtd->drain_wait); |
| atomic_set(&prtd->drain, 0); |
| } |
| prtd->last_buffer = 0; |
| prtd->cmd_ack = 0; |
| if (!prtd->gapless_state.gapless_transition) { |
| pr_debug("issue CMD_FLUSH stream_id %d\n", stream_id); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| q6asm_stream_cmd( |
| prtd->audio_client, CMD_FLUSH, stream_id); |
| spin_lock_irqsave(&prtd->lock, flags); |
| } else { |
| prtd->first_buffer = 0; |
| } |
| /* FIXME. only reset if flush was successful */ |
| prtd->byte_offset = 0; |
| prtd->copied_total = 0; |
| prtd->app_pointer = 0; |
| prtd->bytes_received = 0; |
| prtd->bytes_sent = 0; |
| prtd->marker_timestamp = 0; |
| |
| atomic_set(&prtd->xrun, 0); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| case SNDRV_PCM_TRIGGER_PAUSE_PUSH: |
| pr_debug("SNDRV_PCM_TRIGGER_PAUSE_PUSH transition %d\n", |
| prtd->gapless_state.gapless_transition); |
| if (!prtd->gapless_state.gapless_transition) { |
| pr_debug("issue CMD_PAUSE stream_id %d\n", |
| ac->stream_id); |
| q6asm_stream_cmd_nowait(ac, CMD_PAUSE, ac->stream_id); |
| atomic_set(&prtd->start, 0); |
| } |
| break; |
| case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: |
| pr_debug("SNDRV_PCM_TRIGGER_PAUSE_RELEASE transition %d\n", |
| prtd->gapless_state.gapless_transition); |
| if (!prtd->gapless_state.gapless_transition) { |
| atomic_set(&prtd->start, 1); |
| q6asm_run_nowait(prtd->audio_client, prtd->run_mode, |
| 0, 0); |
| } |
| break; |
| case SND_COMPR_TRIGGER_PARTIAL_DRAIN: |
| pr_debug("%s: SND_COMPR_TRIGGER_PARTIAL_DRAIN\n", __func__); |
| if (!prtd->gapless_state.use_dsp_gapless_mode) { |
| pr_debug("%s: set partial drain as drain\n", __func__); |
| cmd = SND_COMPR_TRIGGER_DRAIN; |
| } |
| case SND_COMPR_TRIGGER_DRAIN: |
| pr_debug("%s: SNDRV_COMPRESS_DRAIN\n", __func__); |
| /* Make sure all the data is sent to DSP before sending EOS */ |
| spin_lock_irqsave(&prtd->lock, flags); |
| |
| if (!atomic_read(&prtd->start)) { |
| pr_err("%s: stream is not in started state\n", |
| __func__); |
| rc = -EPERM; |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| } |
| if (prtd->bytes_received > prtd->copied_total) { |
| pr_debug("%s: wait till all the data is sent to dsp\n", |
| __func__); |
| rc = msm_compr_drain_buffer(prtd, &flags); |
| if (rc || !atomic_read(&prtd->start)) { |
| if (rc != -ENETRESET) |
| rc = -EINTR; |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| } |
| /* |
| * FIXME: Bug. |
| * Write(32767) |
| * Start |
| * Drain <- Indefinite wait |
| * sol1 : if (prtd->copied_total) then wait? |
| * sol2 : (prtd->cmd_interrupt || prtd->drain_ready || |
| * atomic_read(xrun) |
| */ |
| bytes_to_write = prtd->bytes_received |
| - prtd->copied_total; |
| WARN(bytes_to_write > runtime->fragment_size, |
| "last write %d cannot be > than fragment_size", |
| bytes_to_write); |
| |
| if (bytes_to_write > 0) { |
| pr_debug("%s: send %d partial bytes at the end", |
| __func__, bytes_to_write); |
| atomic_set(&prtd->xrun, 0); |
| prtd->last_buffer = 1; |
| msm_compr_send_buffer(prtd); |
| } |
| } |
| |
| if ((cmd == SND_COMPR_TRIGGER_PARTIAL_DRAIN) && |
| (prtd->gapless_state.set_next_stream_id)) { |
| /* wait for the last buffer to be returned */ |
| |
| if (prtd->last_buffer) { |
| pr_debug("%s: last buffer drain\n", __func__); |
| rc = msm_compr_drain_buffer(prtd, &flags); |
| if (rc || !atomic_read(&prtd->start)) { |
| spin_unlock_irqrestore(&prtd->lock, |
| flags); |
| break; |
| } |
| } |
| /* send EOS */ |
| prtd->eos_ack = 0; |
| atomic_set(&prtd->eos, 1); |
| pr_debug("issue CMD_EOS stream_id %d\n", ac->stream_id); |
| q6asm_stream_cmd_nowait(ac, CMD_EOS, ac->stream_id); |
| pr_info("PARTIAL DRAIN, do not wait for EOS ack\n"); |
| |
| /* send a zero length buffer */ |
| atomic_set(&prtd->xrun, 0); |
| msm_compr_send_buffer(prtd); |
| |
| /* wait for the zero length buffer to be returned */ |
| pr_debug("%s: zero length buffer drain\n", __func__); |
| rc = msm_compr_drain_buffer(prtd, &flags); |
| if (rc || !atomic_read(&prtd->start)) { |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| } |
| |
| /* sleep for additional duration partial drain */ |
| atomic_set(&prtd->drain, 1); |
| prtd->drain_ready = 0; |
| pr_debug("%s, additional sleep: %d\n", __func__, |
| prtd->partial_drain_delay); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| rc = wait_event_timeout(prtd->drain_wait, |
| prtd->drain_ready || prtd->cmd_interrupt, |
| msecs_to_jiffies(prtd->partial_drain_delay)); |
| pr_debug("%s: out of additional wait for low sample rate\n", |
| __func__); |
| spin_lock_irqsave(&prtd->lock, flags); |
| if (prtd->cmd_interrupt) { |
| pr_debug("%s: additional wait interrupted by flush)\n", |
| __func__); |
| rc = -EINTR; |
| prtd->cmd_interrupt = 0; |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| } |
| |
| /* move to next stream and reset vars */ |
| pr_debug("%s: Moving to next stream in gapless\n", |
| __func__); |
| ac->stream_id = NEXT_STREAM_ID(ac->stream_id); |
| prtd->byte_offset = 0; |
| prtd->app_pointer = 0; |
| prtd->first_buffer = 1; |
| prtd->last_buffer = 0; |
| /* |
| * Set gapless transition flag only if EOS hasn't been |
| * acknowledged already. |
| */ |
| if (atomic_read(&prtd->eos)) |
| prtd->gapless_state.gapless_transition = 1; |
| prtd->marker_timestamp = 0; |
| |
| /* |
| * Don't reset these as these vars map to |
| * total_bytes_transferred and total_bytes_available |
| * directly, only total_bytes_transferred will be |
| * updated in the next avail() ioctl |
| * prtd->copied_total = 0; |
| * prtd->bytes_received = 0; |
| */ |
| atomic_set(&prtd->drain, 0); |
| atomic_set(&prtd->xrun, 1); |
| pr_debug("%s: issue CMD_RUN", __func__); |
| q6asm_run_nowait(prtd->audio_client, 0, 0, 0); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| } |
| /* |
| * moving to next stream failed, so reset the gapless state |
| * set next stream id for the same session so that the same |
| * stream can be used for gapless playback |
| */ |
| prtd->gapless_state.set_next_stream_id = false; |
| prtd->gapless_state.gapless_transition = 0; |
| pr_debug("%s:CMD_EOS stream_id %d\n", __func__, ac->stream_id); |
| |
| prtd->eos_ack = 0; |
| atomic_set(&prtd->eos, 1); |
| q6asm_stream_cmd_nowait(ac, CMD_EOS, ac->stream_id); |
| |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| |
| |
| /* Wait indefinitely for DRAIN. Flush can also signal this*/ |
| rc = wait_event_interruptible(prtd->eos_wait, |
| (prtd->eos_ack || |
| prtd->cmd_interrupt || |
| atomic_read(&prtd->error))); |
| |
| if (rc < 0) |
| pr_err("%s: EOS wait failed\n", __func__); |
| |
| pr_debug("%s: SNDRV_COMPRESS_DRAIN out of wait for EOS\n", |
| __func__); |
| |
| if (prtd->cmd_interrupt) |
| rc = -EINTR; |
| |
| if (atomic_read(&prtd->error)) { |
| pr_err("%s: Got RESET EVENTS notification, return\n", |
| __func__); |
| rc = -ENETRESET; |
| } |
| |
| /*FIXME : what if a flush comes while PC is here */ |
| if (rc == 0) { |
| /* |
| * Failed to open second stream in DSP for gapless |
| * so prepare the current stream in session |
| * for gapless playback |
| */ |
| spin_lock_irqsave(&prtd->lock, flags); |
| pr_debug("%s:issue CMD_PAUSE stream_id %d", |
| __func__, ac->stream_id); |
| q6asm_stream_cmd_nowait(ac, CMD_PAUSE, ac->stream_id); |
| prtd->cmd_ack = 0; |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| |
| /* |
| * Cache this time as last known time |
| */ |
| if (pdata->use_legacy_api) |
| q6asm_get_session_time_legacy( |
| prtd->audio_client, |
| &prtd->marker_timestamp); |
| else |
| q6asm_get_session_time(prtd->audio_client, |
| &prtd->marker_timestamp); |
| |
| spin_lock_irqsave(&prtd->lock, flags); |
| /* |
| * Don't reset these as these vars map to |
| * total_bytes_transferred and total_bytes_available. |
| * Just total_bytes_transferred will be updated |
| * in the next avail() ioctl. |
| * prtd->copied_total = 0; |
| * prtd->bytes_received = 0; |
| * do not reset prtd->bytes_sent as well as the same |
| * session is used for gapless playback |
| */ |
| prtd->byte_offset = 0; |
| |
| prtd->app_pointer = 0; |
| prtd->first_buffer = 1; |
| prtd->last_buffer = 0; |
| atomic_set(&prtd->drain, 0); |
| atomic_set(&prtd->xrun, 1); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| |
| pr_debug("%s:issue CMD_FLUSH ac->stream_id %d", |
| __func__, ac->stream_id); |
| q6asm_stream_cmd(ac, CMD_FLUSH, ac->stream_id); |
| |
| q6asm_run_nowait(prtd->audio_client, 0, 0, 0); |
| } |
| prtd->cmd_interrupt = 0; |
| break; |
| case SND_COMPR_TRIGGER_NEXT_TRACK: |
| if (!prtd->gapless_state.use_dsp_gapless_mode) { |
| pr_debug("%s: ignore trigger next track\n", __func__); |
| rc = 0; |
| break; |
| } |
| pr_debug("%s: SND_COMPR_TRIGGER_NEXT_TRACK\n", __func__); |
| spin_lock_irqsave(&prtd->lock, flags); |
| rc = 0; |
| /* next stream in gapless */ |
| stream_id = NEXT_STREAM_ID(ac->stream_id); |
| /* |
| * Wait if stream 1 has not completed before honoring next |
| * track for stream 3. Scenario happens if second clip is |
| * small and fills in one buffer so next track will be |
| * called immediately. |
| */ |
| stream_index = STREAM_ARRAY_INDEX(stream_id); |
| if (stream_index >= MAX_NUMBER_OF_STREAMS || |
| stream_index < 0) { |
| pr_err("%s: Invalid stream index: %d", __func__, |
| stream_index); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| rc = -EINVAL; |
| break; |
| } |
| |
| if (prtd->gapless_state.stream_opened[stream_index]) { |
| if (prtd->gapless_state.gapless_transition) { |
| rc = msm_compr_wait_for_stream_avail(prtd, |
| &flags); |
| } else { |
| /* |
| * If session is already opened break out if |
| * the state is not gapless transition. This |
| * is when seek happens after the last buffer |
| * is sent to the driver. Next track would be |
| * called again after last buffer is sent. |
| */ |
| pr_debug("next session is in opened state\n"); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| break; |
| } |
| } |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| if (rc < 0) { |
| /* |
| * if return type EINTR then reset to zero. Tiny |
| * compress treats EINTR as error and prevents PARTIAL |
| * DRAIN. EINTR is not an error. wait for stream avail |
| * is interrupted by some other command like FLUSH. |
| */ |
| if (rc == -EINTR) { |
| pr_debug("%s: EINTR reset rc to 0\n", __func__); |
| rc = 0; |
| } |
| break; |
| } |
| |
| if (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S24_LE) |
| bits_per_sample = 24; |
| else if (prtd->codec_param.codec.format == |
| SNDRV_PCM_FORMAT_S32_LE) |
| bits_per_sample = 32; |
| |
| pr_debug("%s: open_write stream_id %d bits_per_sample %d", |
| __func__, stream_id, bits_per_sample); |
| rc = q6asm_stream_open_write_v4(prtd->audio_client, |
| prtd->codec, bits_per_sample, |
| stream_id, |
| prtd->gapless_state.use_dsp_gapless_mode); |
| if (rc < 0) { |
| pr_err("%s: Session out open failed for gapless\n", |
| __func__); |
| break; |
| } |
| |
| spin_lock_irqsave(&prtd->lock, flags); |
| prtd->gapless_state.stream_opened[stream_index] = 1; |
| prtd->gapless_state.set_next_stream_id = true; |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| |
| rc = msm_compr_send_media_format_block(cstream, |
| stream_id, false); |
| if (rc < 0) { |
| pr_err("%s, failed to send media format block\n", |
| __func__); |
| break; |
| } |
| msm_compr_send_dec_params(cstream, pdata->dec_params[fe_id], |
| stream_id); |
| break; |
| } |
| |
| return rc; |
| } |
| |
| static int msm_compr_pointer(struct snd_compr_stream *cstream, |
| struct snd_compr_tstamp *arg) |
| { |
| struct snd_compr_runtime *runtime = cstream->runtime; |
| struct snd_soc_pcm_runtime *rtd = cstream->private_data; |
| struct msm_compr_audio *prtd = runtime->private_data; |
| struct msm_compr_pdata *pdata = NULL; |
| struct snd_compr_tstamp tstamp; |
| uint64_t timestamp = 0; |
| int rc = 0, first_buffer; |
| unsigned long flags; |
| uint32_t gapless_transition; |
| |
| pdata = snd_soc_platform_get_drvdata(rtd->platform); |
| pr_debug("%s\n", __func__); |
| memset(&tstamp, 0x0, sizeof(struct snd_compr_tstamp)); |
| |
| spin_lock_irqsave(&prtd->lock, flags); |
| tstamp.sampling_rate = prtd->sample_rate; |
| tstamp.byte_offset = prtd->byte_offset; |
| if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| tstamp.copied_total = prtd->copied_total; |
| else if (cstream->direction == SND_COMPRESS_CAPTURE) |
| tstamp.copied_total = prtd->received_total; |
| first_buffer = prtd->first_buffer; |
| if (atomic_read(&prtd->error)) { |
| pr_err("%s Got RESET EVENTS notification, return error\n", |
| __func__); |
| if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| runtime->total_bytes_transferred = tstamp.copied_total; |
| else |
| runtime->total_bytes_available = tstamp.copied_total; |
| tstamp.pcm_io_frames = 0; |
| memcpy(arg, &tstamp, sizeof(struct snd_compr_tstamp)); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| return -ENETRESET; |
| } |
| if (cstream->direction == SND_COMPRESS_PLAYBACK) { |
| |
| gapless_transition = prtd->gapless_state.gapless_transition; |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| if (gapless_transition) |
| pr_debug("%s session time in gapless transition", |
| __func__); |
| /* |
| *- Do not query if no buffer has been given. |
| *- Do not query on a gapless transition. |
| * Playback for the 2nd stream can start (thus returning time |
| * starting from 0) before the driver knows about EOS of first |
| * stream. |
| */ |
| if (!first_buffer || gapless_transition) { |
| |
| if (pdata->use_legacy_api) |
| rc = q6asm_get_session_time_legacy( |
| prtd->audio_client, &prtd->marker_timestamp); |
| else |
| rc = q6asm_get_session_time( |
| prtd->audio_client, &prtd->marker_timestamp); |
| if (rc < 0) { |
| pr_err("%s: Get Session Time return =%lld\n", |
| __func__, timestamp); |
| if (atomic_read(&prtd->error)) |
| return -ENETRESET; |
| else |
| return -EAGAIN; |
| } |
| } |
| } else { |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| } |
| timestamp = prtd->marker_timestamp; |
| |
| /* DSP returns timestamp in usec */ |
| pr_debug("%s: timestamp = %lld usec\n", __func__, timestamp); |
| timestamp *= prtd->sample_rate; |
| tstamp.pcm_io_frames = (snd_pcm_uframes_t)div64_u64(timestamp, 1000000); |
| memcpy(arg, &tstamp, sizeof(struct snd_compr_tstamp)); |
| |
| return 0; |
| } |
| |
| static int msm_compr_ack(struct snd_compr_stream *cstream, |
| size_t count) |
| { |
| struct snd_compr_runtime *runtime = cstream->runtime; |
| struct msm_compr_audio *prtd = runtime->private_data; |
| void *src, *dstn; |
| size_t copy; |
| unsigned long flags; |
| |
| WARN(1, "This path is untested"); |
| return -EINVAL; |
| |
| pr_debug("%s: count = %zd\n", __func__, count); |
| if (!prtd->buffer) { |
| pr_err("%s: Buffer is not allocated yet ??\n", __func__); |
| return -EINVAL; |
| } |
| src = runtime->buffer + prtd->app_pointer; |
| dstn = prtd->buffer + prtd->app_pointer; |
| if (count < prtd->buffer_size - prtd->app_pointer) { |
| memcpy(dstn, src, count); |
| prtd->app_pointer += count; |
| } else { |
| copy = prtd->buffer_size - prtd->app_pointer; |
| memcpy(dstn, src, copy); |
| memcpy(prtd->buffer, runtime->buffer, count - copy); |
| prtd->app_pointer = count - copy; |
| } |
| |
| /* |
| * If the stream is started and all the bytes received were |
| * copied to DSP, the newly received bytes should be |
| * sent right away |
| */ |
| spin_lock_irqsave(&prtd->lock, flags); |
| |
| if (atomic_read(&prtd->start) && |
| prtd->bytes_received == prtd->copied_total) { |
| prtd->bytes_received += count; |
| msm_compr_send_buffer(prtd); |
| } else |
| prtd->bytes_received += count; |
| |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| |
| return 0; |
| } |
| |
| static int msm_compr_playback_copy(struct snd_compr_stream *cstream, |
| char __user *buf, size_t count) |
| { |
| struct snd_compr_runtime *runtime = cstream->runtime; |
| struct msm_compr_audio *prtd = runtime->private_data; |
| void *dstn; |
| size_t copy; |
| uint64_t bytes_available = 0; |
| unsigned long flags; |
| |
| pr_debug("%s: count = %zd\n", __func__, count); |
| if (!prtd->buffer) { |
| pr_err("%s: Buffer is not allocated yet ??", __func__); |
| return 0; |
| } |
| |
| spin_lock_irqsave(&prtd->lock, flags); |
| if (atomic_read(&prtd->error)) { |
| pr_err("%s Got RESET EVENTS notification", __func__); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| return -ENETRESET; |
| } |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| |
| dstn = prtd->buffer + prtd->app_pointer; |
| if (count < prtd->buffer_size - prtd->app_pointer) { |
| if (copy_from_user(dstn, buf, count)) |
| return -EFAULT; |
| prtd->app_pointer += count; |
| } else { |
| copy = prtd->buffer_size - prtd->app_pointer; |
| if (copy_from_user(dstn, buf, copy)) |
| return -EFAULT; |
| if (copy_from_user(prtd->buffer, buf + copy, count - copy)) |
| return -EFAULT; |
| prtd->app_pointer = count - copy; |
| } |
| |
| /* |
| * If stream is started and there has been an xrun, |
| * since the available bytes fits fragment_size, copy the data |
| * right away. |
| */ |
| spin_lock_irqsave(&prtd->lock, flags); |
| prtd->bytes_received += count; |
| if (atomic_read(&prtd->start)) { |
| if (atomic_read(&prtd->xrun)) { |
| pr_debug("%s: in xrun, count = %zd\n", __func__, count); |
| bytes_available = prtd->bytes_received - |
| prtd->copied_total; |
| if (bytes_available >= runtime->fragment_size) { |
| pr_debug("%s: handle xrun, bytes_to_write = %llu\n", |
| __func__, bytes_available); |
| atomic_set(&prtd->xrun, 0); |
| msm_compr_send_buffer(prtd); |
| } /* else not sufficient data */ |
| } /* writes will continue on the next write_done */ |
| } |
| |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| |
| return count; |
| } |
| |
| static int msm_compr_capture_copy(struct snd_compr_stream *cstream, |
| char __user *buf, size_t count) |
| { |
| struct snd_compr_runtime *runtime = cstream->runtime; |
| struct msm_compr_audio *prtd = runtime->private_data; |
| void *source; |
| unsigned long flags; |
| |
| pr_debug("%s: count = %zd\n", __func__, count); |
| if (!prtd->buffer) { |
| pr_err("%s: Buffer is not allocated yet ??", __func__); |
| return 0; |
| } |
| |
| spin_lock_irqsave(&prtd->lock, flags); |
| if (atomic_read(&prtd->error)) { |
| pr_err("%s Got RESET EVENTS notification", __func__); |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| return -ENETRESET; |
| } |
| |
| source = prtd->buffer + prtd->app_pointer; |
| /* check if we have requested amount of data to copy to user*/ |
| if (count <= prtd->received_total - prtd->bytes_copied) { |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| if (copy_to_user(buf, source, count)) { |
| pr_err("copy_to_user failed"); |
| return -EFAULT; |
| } |
| spin_lock_irqsave(&prtd->lock, flags); |
| prtd->app_pointer += count; |
| if (prtd->app_pointer >= prtd->buffer_size) |
| prtd->app_pointer -= prtd->buffer_size; |
| prtd->bytes_copied += count; |
| } |
| msm_compr_read_buffer(prtd); |
| |
| spin_unlock_irqrestore(&prtd->lock, flags); |
| return count; |
| } |
| |
| static int msm_compr_copy(struct snd_compr_stream *cstream, |
| char __user *buf, size_t count) |
| { |
| int ret = 0; |
| |
| pr_debug(" In %s\n", __func__); |
| if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| ret = msm_compr_playback_copy(cstream, buf, count); |
| else if (cstream->direction == SND_COMPRESS_CAPTURE) |
| ret = msm_compr_capture_copy(cstream, buf, count); |
| return ret; |
| } |
| |
| static int msm_compr_get_caps(struct snd_compr_stream *cstream, |
| struct snd_compr_caps *arg) |
| { |
| struct snd_compr_runtime *runtime = cstream->runtime; |
| struct msm_compr_audio *prtd = runtime->private_data; |
| int ret = 0; |
| |
| pr_debug("%s\n", __func__); |
| if ((arg != NULL) && (prtd != NULL)) { |
| memcpy(arg, &prtd->compr_cap, sizeof(struct snd_compr_caps)); |
| } else { |
| ret = -EINVAL; |
| pr_err("%s: arg (0x%pK), prtd (0x%pK)\n", __func__, arg, prtd); |
| } |
| |
| return ret; |
| } |
| |
| static int msm_compr_get_codec_caps(struct snd_compr_stream *cstream, |
| struct snd_compr_codec_caps *codec) |
| { |
| pr_debug("%s\n", __func__); |
| |
| switch (codec->codec) { |
| case SND_AUDIOCODEC_MP3: |
| codec->num_descriptors = 2; |
| codec->descriptor[0].max_ch = 2; |
| memcpy(codec->descriptor[0].sample_rates, |
| supported_sample_rates, |
| sizeof(supported_sample_rates)); |
| codec->descriptor[0].num_sample_rates = |
| sizeof(supported_sample_rates)/sizeof(unsigned int); |
| codec->descriptor[0].bit_rate[0] = 320; /* 320kbps */ |
| codec->descriptor[0].bit_rate[1] = 128; |
| codec->descriptor[0].num_bitrates = 2; |
| codec->descriptor[0].profiles = 0; |
| codec->descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO; |
| codec->descriptor[0].formats = 0; |
| break; |
| case SND_AUDIOCODEC_AAC: |
| codec->num_descriptors = 2; |
| codec->descriptor[1].max_ch = 2; |
| memcpy(codec->descriptor[1].sample_rates, |
| supported_sample_rates, |
| sizeof(supported_sample_rates)); |
| codec->descriptor[1].num_sample_rates = |
| sizeof(supported_sample_rates)/sizeof(unsigned int); |
| codec->descriptor[1].bit_rate[0] = 320; /* 320kbps */ |
| codec->descriptor[1].bit_rate[1] = 128; |
| codec->descriptor[1].num_bitrates = 2; |
| codec->descriptor[1].profiles = 0; |
| codec->descriptor[1].modes = 0; |
| codec->descriptor[1].formats = |
| (SND_AUDIOSTREAMFORMAT_MP4ADTS | |
| SND_AUDIOSTREAMFORMAT_RAW); |
| break; |
| case SND_AUDIOCODEC_AC3: |
| case SND_AUDIOCODEC_EAC3: |
| case SND_AUDIOCODEC_FLAC: |
| case SND_AUDIOCODEC_VORBIS: |
| case SND_AUDIOCODEC_ALAC: |
| case SND_AUDIOCODEC_APE: |
| case SND_AUDIOCODEC_DTS: |
| case SND_AUDIOCODEC_DSD: |
| case SND_AUDIOCODEC_TRUEHD: |
| case SND_AUDIOCODEC_IEC61937: |
| case SND_AUDIOCODEC_APTX: |
| break; |
| default: |
| pr_err("%s: Unsupported audio codec %d\n", |
| __func__, codec->codec); |
| return -EINVAL; |
| } |
| |
| return 0; |
| } |
| |
| static int msm_compr_set_metadata(struct snd_compr_stream *cstream, |
| struct snd_compr_metadata *metadata) |
| { |
| struct msm_compr_audio *prtd; |
| struct audio_client *ac; |
| pr_debug("%s\n", __func__); |
| |
| if (!metadata || !cstream) |
| return -EINVAL; |
| |
| prtd = cstream->runtime->private_data; |
| if (!prtd || !prtd->audio_client) { |
| pr_err("%s: prtd or audio client is NULL\n", __func__); |
| return -EINVAL; |
| } |
| |
| if (((metadata->key == SNDRV_COMPRESS_ENCODER_PADDING) || |
| (metadata->key == SNDRV_COMPRESS_ENCODER_DELAY)) && |
| (prtd->compr_passthr != LEGACY_PCM)) { |
| pr_debug("%s: No trailing silence for compress_type[%d]\n", |
| __func__, prtd->compr_passthr); |
| return 0; |
| } |
| |
| ac = prtd->audio_client; |
| if (metadata->key == SNDRV_COMPRESS_ENCODER_PADDING) { |
| pr_debug("%s, got encoder padding %u", |
| __func__, metadata->value[0]); |
| prtd->gapless_state.trailing_samples_drop = metadata->value[0]; |
| } else if (metadata->key == SNDRV_COMPRESS_ENCODER_DELAY) { |
| pr_debug("%s, got encoder delay %u", |
| __func__, metadata->value[0]); |
| prtd->gapless_state.initial_samples_drop = metadata->value[0]; |
| } else if (metadata->key == SNDRV_COMPRESS_RENDER_MODE) { |
| return msm_compr_set_render_mode(prtd, metadata->value[0]); |
| } else if (metadata->key == SNDRV_COMPRESS_CLK_REC_MODE) { |
| return msm_compr_set_clk_rec_mode(ac, metadata->value[0]); |
| } else if (metadata->key == SNDRV_COMPRESS_RENDER_WINDOW) { |
| return msm_compr_set_render_window( |
| ac, |
| metadata->value[0], |
| metadata->value[1], |
| metadata->value[2], |
| metadata->value[3]); |
| } else if (metadata->key == SNDRV_COMPRESS_START_DELAY) { |
| prtd->start_delay_lsw = metadata->value[0]; |
| prtd->start_delay_msw = metadata->value[1]; |
| } else if (metadata->key == |
| SNDRV_COMPRESS_ENABLE_ADJUST_SESSION_CLOCK) { |
| return msm_compr_enable_adjust_session_clock(ac, |
| metadata->value[0]); |
| } else if (metadata->key == SNDRV_COMPRESS_ADJUST_SESSION_CLOCK) { |
| return msm_compr_adjust_session_clock(ac, |
| metadata->value[0], |
| metadata->value[1]); |
| } |
| |
| return 0; |
| } |
| |
| static int msm_compr_get_metadata(struct snd_compr_stream *cstream, |
| struct snd_compr_metadata *metadata) |
| { |
| struct msm_compr_audio *prtd; |
| struct audio_client *ac; |
| int ret = -EINVAL; |
| |
| pr_debug("%s\n", __func__); |
| |
| if (!metadata || !cstream || !cstream->runtime) |
| return ret; |
| |
| if (metadata->key != SNDRV_COMPRESS_PATH_DELAY) { |
| pr_err("%s, unsupported key %d\n", __func__, metadata->key); |
| return ret; |
| } |
| |
| prtd = cstream->runtime->private_data; |
| if (!prtd || !prtd->audio_client) { |
| pr_err("%s: prtd or audio client is NULL\n", __func__); |
| return ret; |
| } |
| |
| ac = prtd->audio_client; |
| ret = q6asm_get_path_delay(prtd->audio_client); |
| if (ret) { |
| pr_err("%s: get_path_delay failed, ret=%d\n", __func__, ret); |
| return ret; |
| } |
| |
| pr_debug("%s, path delay(in us) %u\n", __func__, ac->path_delay); |
| |
| metadata->value[0] = ac->path_delay; |
| |
| return ret; |
| } |
| |
| |
| static int msm_compr_set_next_track_param(struct snd_compr_stream *cstream, |
| union snd_codec_options *codec_options) |
| { |
| struct msm_compr_audio *prtd; |
| struct audio_client *ac; |
| int ret = 0; |
| |
| if (!codec_options || !cstream) |
| return -EINVAL; |
| |
| prtd = cstream->runtime->private_data; |
| if (!prtd || !prtd->audio_client) { |
| pr_err("%s: prtd or audio client is NULL\n", __func__); |
| return -EINVAL; |
| } |
| |
| ac = prtd->audio_client; |
| |
| pr_debug("%s: got codec options for codec type %u", |
| __func__, prtd->codec); |
| switch (prtd->codec) { |
| case FORMAT_WMA_V9: |
| case FORMAT_WMA_V10PRO: |
| case FORMAT_FLAC: |
| case FORMAT_VORBIS: |
| case FORMAT_ALAC: |
| case FORMAT_APE: |
| memcpy(&(prtd->gapless_state.codec_options), |
| codec_options, |
| sizeof(union snd_codec_options)); |
| ret = msm_compr_send_media_format_block(cstream, |
| ac->stream_id, true); |
| if (ret < 0) { |
| pr_err("%s: failed to send media format block\n", |
| __func__); |
| } |
| break; |
| |
| default: |
| pr_debug("%s: Ignore sending CMD Format block\n", |
| __func__); |
| break; |
| } |
| |
| return ret; |
| } |
| |
| static int msm_compr_volume_put(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| unsigned long fe_id = kcontrol->private_value; |
| struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| snd_soc_component_get_drvdata(comp); |
| struct snd_compr_stream *cstream = NULL; |
| uint32_t *volume = NULL; |
| |
| if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| pr_err("%s Received out of bounds fe_id %lu\n", |
| __func__, fe_id); |
| return -EINVAL; |
| } |
| |
| cstream = pdata->cstream[fe_id]; |
| volume = pdata->volume[fe_id]; |
| |
| volume[0] = ucontrol->value.integer.value[0]; |
| volume[1] = ucontrol->value.integer.value[1]; |
| pr_debug("%s: fe_id %lu left_vol %d right_vol %d\n", |
| __func__, fe_id, volume[0], volume[1]); |
| if (cstream) |
| msm_compr_set_volume(cstream, volume[0], volume[1]); |
| return 0; |
| } |
| |
| static int msm_compr_volume_get(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| unsigned long fe_id = kcontrol->private_value; |
| |
| struct msm_compr_pdata *pdata = |
| snd_soc_component_get_drvdata(comp); |
| uint32_t *volume = NULL; |
| |
| if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| pr_err("%s Received out of bound fe_id %lu\n", __func__, fe_id); |
| return -EINVAL; |
| } |
| |
| volume = pdata->volume[fe_id]; |
| pr_debug("%s: fe_id %lu\n", __func__, fe_id); |
| ucontrol->value.integer.value[0] = volume[0]; |
| ucontrol->value.integer.value[1] = volume[1]; |
| |
| return 0; |
| } |
| |
| static int msm_compr_audio_effects_config_put(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| unsigned long fe_id = kcontrol->private_value; |
| struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| snd_soc_component_get_drvdata(comp); |
| struct msm_compr_audio_effects *audio_effects = NULL; |
| struct snd_compr_stream *cstream = NULL; |
| struct msm_compr_audio *prtd = NULL; |
| long *values = &(ucontrol->value.integer.value[0]); |
| int effects_module; |
| |
| pr_debug("%s\n", __func__); |
| if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| pr_err("%s Received out of bounds fe_id %lu\n", |
| __func__, fe_id); |
| return -EINVAL; |
| } |
| cstream = pdata->cstream[fe_id]; |
| audio_effects = pdata->audio_effects[fe_id]; |
| if (!cstream || !audio_effects) { |
| pr_err("%s: stream or effects inactive\n", __func__); |
| return -EINVAL; |
| } |
| prtd = cstream->runtime->private_data; |
| if (!prtd) { |
| pr_err("%s: cannot set audio effects\n", __func__); |
| return -EINVAL; |
| } |
| if (prtd->compr_passthr != LEGACY_PCM) { |
| pr_debug("%s: No effects for compr_type[%d]\n", |
| __func__, prtd->compr_passthr); |
| return 0; |
| } |
| pr_debug("%s: Effects supported for compr_type[%d]\n", |
| __func__, prtd->compr_passthr); |
| |
| effects_module = *values++; |
| switch (effects_module) { |
| case VIRTUALIZER_MODULE: |
| pr_debug("%s: VIRTUALIZER_MODULE\n", __func__); |
| if (msm_audio_effects_is_effmodule_supp_in_top(effects_module, |
| prtd->audio_client->topology)) |
| msm_audio_effects_virtualizer_handler( |
| prtd->audio_client, |
| &(audio_effects->virtualizer), |
| values); |
| break; |
| case REVERB_MODULE: |
| pr_debug("%s: REVERB_MODULE\n", __func__); |
| if (msm_audio_effects_is_effmodule_supp_in_top(effects_module, |
| prtd->audio_client->topology)) |
| msm_audio_effects_reverb_handler(prtd->audio_client, |
| &(audio_effects->reverb), |
| values); |
| break; |
| case BASS_BOOST_MODULE: |
| pr_debug("%s: BASS_BOOST_MODULE\n", __func__); |
| if (msm_audio_effects_is_effmodule_supp_in_top(effects_module, |
| prtd->audio_client->topology)) |
| msm_audio_effects_bass_boost_handler(prtd->audio_client, |
| &(audio_effects->bass_boost), |
| values); |
| break; |
| case PBE_MODULE: |
| pr_debug("%s: PBE_MODULE\n", __func__); |
| if (msm_audio_effects_is_effmodule_supp_in_top(effects_module, |
| prtd->audio_client->topology)) |
| msm_audio_effects_pbe_handler(prtd->audio_client, |
| &(audio_effects->pbe), |
| values); |
| break; |
| case EQ_MODULE: |
| pr_debug("%s: EQ_MODULE\n", __func__); |
| if (msm_audio_effects_is_effmodule_supp_in_top(effects_module, |
| prtd->audio_client->topology)) |
| msm_audio_effects_popless_eq_handler(prtd->audio_client, |
| &(audio_effects->equalizer), |
| values); |
| break; |
| case SOFT_VOLUME_MODULE: |
| pr_debug("%s: SOFT_VOLUME_MODULE\n", __func__); |
| break; |
| case SOFT_VOLUME2_MODULE: |
| pr_debug("%s: SOFT_VOLUME2_MODULE\n", __func__); |
| if (msm_audio_effects_is_effmodule_supp_in_top(effects_module, |
| prtd->audio_client->topology)) |
| msm_audio_effects_volume_handler_v2(prtd->audio_client, |
| &(audio_effects->volume), |
| values, SOFT_VOLUME_INSTANCE_2); |
| break; |
| default: |
| pr_err("%s Invalid effects config module\n", __func__); |
| return -EINVAL; |
| } |
| return 0; |
| } |
| |
| static int msm_compr_audio_effects_config_get(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| unsigned long fe_id = kcontrol->private_value; |
| struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| snd_soc_component_get_drvdata(comp); |
| struct msm_compr_audio_effects *audio_effects = NULL; |
| struct snd_compr_stream *cstream = NULL; |
| struct msm_compr_audio *prtd = NULL; |
| |
| pr_debug("%s\n", __func__); |
| if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| pr_err("%s Received out of bounds fe_id %lu\n", |
| __func__, fe_id); |
| return -EINVAL; |
| } |
| cstream = pdata->cstream[fe_id]; |
| audio_effects = pdata->audio_effects[fe_id]; |
| if (!cstream || !audio_effects) { |
| pr_err("%s: stream or effects inactive\n", __func__); |
| return -EINVAL; |
| } |
| prtd = cstream->runtime->private_data; |
| if (!prtd) { |
| pr_err("%s: cannot set audio effects\n", __func__); |
| return -EINVAL; |
| } |
| |
| return 0; |
| } |
| |
| static int msm_compr_query_audio_effect_put(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| unsigned long fe_id = kcontrol->private_value; |
| struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| snd_soc_component_get_drvdata(comp); |
| struct msm_compr_audio_effects *audio_effects = NULL; |
| struct snd_compr_stream *cstream = NULL; |
| struct msm_compr_audio *prtd = NULL; |
| long *values = &(ucontrol->value.integer.value[0]); |
| |
| if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| pr_err("%s Received out of bounds fe_id %lu\n", |
| __func__, fe_id); |
| return -EINVAL; |
| } |
| cstream = pdata->cstream[fe_id]; |
| audio_effects = pdata->audio_effects[fe_id]; |
| if (!cstream || !audio_effects) { |
| pr_err("%s: stream or effects inactive\n", __func__); |
| return -EINVAL; |
| } |
| prtd = cstream->runtime->private_data; |
| if (!prtd) { |
| pr_err("%s: cannot set audio effects\n", __func__); |
| return -EINVAL; |
| } |
| if (prtd->compr_passthr != LEGACY_PCM) { |
| pr_err("%s: No effects for compr_type[%d]\n", |
| __func__, prtd->compr_passthr); |
| return -EPERM; |
| } |
| audio_effects->query.mod_id = (u32)*values++; |
| audio_effects->query.parm_id = (u32)*values++; |
| audio_effects->query.size = (u32)*values++; |
| audio_effects->query.offset = (u32)*values++; |
| audio_effects->query.device = (u32)*values++; |
| return 0; |
| } |
| |
| static int msm_compr_query_audio_effect_get(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| unsigned long fe_id = kcontrol->private_value; |
| struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| snd_soc_component_get_drvdata(comp); |
| struct msm_compr_audio_effects *audio_effects = NULL; |
| struct snd_compr_stream *cstream = NULL; |
| struct msm_compr_audio *prtd = NULL; |
| long *values = &(ucontrol->value.integer.value[0]); |
| |
| if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| pr_err("%s Received out of bounds fe_id %lu\n", |
| __func__, fe_id); |
| return -EINVAL; |
| } |
| cstream = pdata->cstream[fe_id]; |
| audio_effects = pdata->audio_effects[fe_id]; |
| if (!cstream || !audio_effects) { |
| pr_debug("%s: stream or effects inactive\n", __func__); |
| return -EINVAL; |
| } |
| prtd = cstream->runtime->private_data; |
| if (!prtd) { |
| pr_err("%s: cannot set audio effects\n", __func__); |
| return -EINVAL; |
| } |
| values[0] = (long)audio_effects->query.mod_id; |
| values[1] = (long)audio_effects->query.parm_id; |
| values[2] = (long)audio_effects->query.size; |
| values[3] = (long)audio_effects->query.offset; |
| values[4] = (long)audio_effects->query.device; |
| return 0; |
| } |
| |
| static int msm_compr_send_dec_params(struct snd_compr_stream *cstream, |
| struct msm_compr_dec_params *dec_params, |
| int stream_id) |
| { |
| |
| int rc = 0; |
| struct msm_compr_audio *prtd = NULL; |
| struct snd_dec_ddp *ddp = &dec_params->ddp_params; |
| |
| if (!cstream || !dec_params) { |
| pr_err("%s: stream or dec_params inactive\n", __func__); |
| rc = -EINVAL; |
| goto end; |
| } |
| prtd = cstream->runtime->private_data; |
| if (!prtd) { |
| pr_err("%s: cannot set dec_params\n", __func__); |
| rc = -EINVAL; |
| goto end; |
| } |
| switch (prtd->codec) { |
| case FORMAT_MP3: |
| case FORMAT_MPEG4_AAC: |
| case FORMAT_TRUEHD: |
| case FORMAT_IEC61937: |
| case FORMAT_APTX: |
| pr_debug("%s: no runtime parameters for codec: %d\n", __func__, |
| prtd->codec); |
| break; |
| case FORMAT_AC3: |
| case FORMAT_EAC3: |
| if (prtd->compr_passthr != LEGACY_PCM) { |
| pr_debug("%s: No DDP param for compr_type[%d]\n", |
| __func__, prtd->compr_passthr); |
| break; |
| } |
| rc = msm_compr_send_ddp_cfg(prtd->audio_client, ddp, stream_id); |
| if (rc < 0) |
| pr_err("%s: DDP CMD CFG failed %d\n", __func__, rc); |
| break; |
| default: |
| break; |
| } |
| end: |
| return rc; |
| |
| } |
| static int msm_compr_dec_params_put(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| unsigned long fe_id = kcontrol->private_value; |
| struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| snd_soc_component_get_drvdata(comp); |
| struct msm_compr_dec_params *dec_params = NULL; |
| struct snd_compr_stream *cstream = NULL; |
| struct msm_compr_audio *prtd = NULL; |
| long *values = &(ucontrol->value.integer.value[0]); |
| int rc = 0; |
| |
| pr_debug("%s\n", __func__); |
| if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| pr_err("%s Received out of bounds fe_id %lu\n", |
| __func__, fe_id); |
| rc = -EINVAL; |
| goto end; |
| } |
| |
| cstream = pdata->cstream[fe_id]; |
| dec_params = pdata->dec_params[fe_id]; |
| |
| if (!cstream || !dec_params) { |
| pr_err("%s: stream or dec_params inactive\n", __func__); |
| rc = -EINVAL; |
| goto end; |
| } |
| prtd = cstream->runtime->private_data; |
| if (!prtd) { |
| pr_err("%s: cannot set dec_params\n", __func__); |
| rc = -EINVAL; |
| goto end; |
| } |
| |
| switch (prtd->codec) { |
| case FORMAT_MP3: |
| case FORMAT_MPEG4_AAC: |
| case FORMAT_FLAC: |
| case FORMAT_VORBIS: |
| case FORMAT_ALAC: |
| case FORMAT_APE: |
| case FORMAT_DTS: |
| case FORMAT_DSD: |
| case FORMAT_TRUEHD: |
| case FORMAT_IEC61937: |
| case FORMAT_APTX: |
| pr_debug("%s: no runtime parameters for codec: %d\n", __func__, |
| prtd->codec); |
| break; |
| case FORMAT_AC3: |
| case FORMAT_EAC3: { |
| struct snd_dec_ddp *ddp = &dec_params->ddp_params; |
| int cnt; |
| |
| if (prtd->compr_passthr != LEGACY_PCM) { |
| pr_debug("%s: No DDP param for compr_type[%d]\n", |
| __func__, prtd->compr_passthr); |
| break; |
| } |
| |
| ddp->params_length = (*values++); |
| if (ddp->params_length > DDP_DEC_MAX_NUM_PARAM) { |
| pr_err("%s: invalid num of params:: %d\n", __func__, |
| ddp->params_length); |
| rc = -EINVAL; |
| goto end; |
| } |
| for (cnt = 0; cnt < ddp->params_length; cnt++) { |
| ddp->params_id[cnt] = *values++; |
| ddp->params_value[cnt] = *values++; |
| } |
| prtd = cstream->runtime->private_data; |
| if (prtd && prtd->audio_client) |
| rc = msm_compr_send_dec_params(cstream, dec_params, |
| prtd->audio_client->stream_id); |
| break; |
| } |
| default: |
| break; |
| } |
| end: |
| pr_debug("%s: ret %d\n", __func__, rc); |
| return rc; |
| } |
| |
| static int msm_compr_dec_params_get(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| /* dummy function */ |
| return 0; |
| } |
| |
| static int msm_compr_playback_app_type_cfg_put(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| u64 fe_id = kcontrol->private_value; |
| int session_type = SESSION_TYPE_RX; |
| int be_id = ucontrol->value.integer.value[3]; |
| struct msm_pcm_stream_app_type_cfg cfg_data = {0, 0, 48000}; |
| int ret = 0; |
| |
| cfg_data.app_type = ucontrol->value.integer.value[0]; |
| cfg_data.acdb_dev_id = ucontrol->value.integer.value[1]; |
| if (ucontrol->value.integer.value[2] != 0) |
| cfg_data.sample_rate = ucontrol->value.integer.value[2]; |
| pr_debug("%s: fe_id- %llu session_type- %d be_id- %d app_type- %d acdb_dev_id- %d sample_rate- %d\n", |
| __func__, fe_id, session_type, be_id, |
| cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate); |
| ret = msm_pcm_routing_reg_stream_app_type_cfg(fe_id, session_type, |
| be_id, &cfg_data); |
| if (ret < 0) |
| pr_err("%s: msm_pcm_routing_reg_stream_app_type_cfg failed returned %d\n", |
| __func__, ret); |
| |
| return ret; |
| } |
| |
| static int msm_compr_playback_app_type_cfg_get(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| u64 fe_id = kcontrol->private_value; |
| int session_type = SESSION_TYPE_RX; |
| int be_id = 0; |
| struct msm_pcm_stream_app_type_cfg cfg_data = {0}; |
| int ret = 0; |
| |
| ret = msm_pcm_routing_get_stream_app_type_cfg(fe_id, session_type, |
| &be_id, &cfg_data); |
| if (ret < 0) { |
| pr_err("%s: msm_pcm_routing_get_stream_app_type_cfg failed returned %d\n", |
| __func__, ret); |
| goto done; |
| } |
| |
| ucontrol->value.integer.value[0] = cfg_data.app_type; |
| ucontrol->value.integer.value[1] = cfg_data.acdb_dev_id; |
| ucontrol->value.integer.value[2] = cfg_data.sample_rate; |
| ucontrol->value.integer.value[3] = be_id; |
| pr_debug("%s: fedai_id %llu, session_type %d, be_id %d, app_type %d, acdb_dev_id %d, sample_rate %d\n", |
| __func__, fe_id, session_type, be_id, |
| cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate); |
| done: |
| return ret; |
| } |
| |
| static int msm_compr_capture_app_type_cfg_put(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| u64 fe_id = kcontrol->private_value; |
| int session_type = SESSION_TYPE_TX; |
| int be_id = ucontrol->value.integer.value[3]; |
| struct msm_pcm_stream_app_type_cfg cfg_data = {0, 0, 48000}; |
| int ret = 0; |
| |
| cfg_data.app_type = ucontrol->value.integer.value[0]; |
| cfg_data.acdb_dev_id = ucontrol->value.integer.value[1]; |
| if (ucontrol->value.integer.value[2] != 0) |
| cfg_data.sample_rate = ucontrol->value.integer.value[2]; |
| pr_debug("%s: fe_id- %llu session_type- %d be_id- %d app_type- %d acdb_dev_id- %d sample_rate- %d\n", |
| __func__, fe_id, session_type, be_id, |
| cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate); |
| ret = msm_pcm_routing_reg_stream_app_type_cfg(fe_id, session_type, |
| be_id, &cfg_data); |
| if (ret < 0) |
| pr_err("%s: msm_pcm_routing_reg_stream_app_type_cfg failed returned %d\n", |
| __func__, ret); |
| |
| return ret; |
| } |
| |
| static int msm_compr_capture_app_type_cfg_get(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| u64 fe_id = kcontrol->private_value; |
| int session_type = SESSION_TYPE_TX; |
| int be_id = 0; |
| struct msm_pcm_stream_app_type_cfg cfg_data = {0}; |
| int ret = 0; |
| |
| ret = msm_pcm_routing_get_stream_app_type_cfg(fe_id, session_type, |
| &be_id, &cfg_data); |
| if (ret < 0) { |
| pr_err("%s: msm_pcm_routing_get_stream_app_type_cfg failed returned %d\n", |
| __func__, ret); |
| goto done; |
| } |
| |
| ucontrol->value.integer.value[0] = cfg_data.app_type; |
| ucontrol->value.integer.value[1] = cfg_data.acdb_dev_id; |
| ucontrol->value.integer.value[2] = cfg_data.sample_rate; |
| ucontrol->value.integer.value[3] = be_id; |
| pr_debug("%s: fedai_id %llu, session_type %d, be_id %d, app_type %d, acdb_dev_id %d, sample_rate %d\n", |
| __func__, fe_id, session_type, be_id, |
| cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate); |
| done: |
| return ret; |
| } |
| |
| static int msm_compr_channel_map_put(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| u64 fe_id = kcontrol->private_value; |
| struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| snd_soc_component_get_drvdata(comp); |
| int rc = 0, i; |
| |
| pr_debug("%s: fe_id- %llu\n", __func__, fe_id); |
| |
| if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| pr_err("%s Received out of bounds fe_id %llu\n", |
| __func__, fe_id); |
| rc = -EINVAL; |
| goto end; |
| } |
| |
| if (pdata->ch_map[fe_id]) { |
| pdata->ch_map[fe_id]->set_ch_map = true; |
| for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL; i++) |
| pdata->ch_map[fe_id]->channel_map[i] = |
| (char)(ucontrol->value.integer.value[i]); |
| } else { |
| pr_debug("%s: no memory for ch_map, default will be set\n", |
| __func__); |
| } |
| end: |
| pr_debug("%s: ret %d\n", __func__, rc); |
| return rc; |
| } |
| |
| static int msm_compr_channel_map_get(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| u64 fe_id = kcontrol->private_value; |
| struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| snd_soc_component_get_drvdata(comp); |
| int rc = 0, i; |
| |
| pr_debug("%s: fe_id- %llu\n", __func__, fe_id); |
| if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| pr_err("%s: Received out of bounds fe_id %llu\n", |
| __func__, fe_id); |
| rc = -EINVAL; |
| goto end; |
| } |
| if (pdata->ch_map[fe_id]) { |
| for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL; i++) |
| ucontrol->value.integer.value[i] = |
| pdata->ch_map[fe_id]->channel_map[i]; |
| } |
| end: |
| pr_debug("%s: ret %d\n", __func__, rc); |
| return rc; |
| } |
| |
| static int msm_compr_adsp_stream_cmd_put(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| unsigned long fe_id = kcontrol->private_value; |
| struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| snd_soc_component_get_drvdata(comp); |
| struct snd_compr_stream *cstream = NULL; |
| struct msm_compr_audio *prtd; |
| int ret = 0; |
| struct msm_adsp_event_data *event_data = NULL; |
| |
| if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| pr_err("%s Received invalid fe_id %lu\n", |
| __func__, fe_id); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| cstream = pdata->cstream[fe_id]; |
| if (cstream == NULL) { |
| pr_err("%s cstream is null\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| prtd = cstream->runtime->private_data; |
| if (!prtd) { |
| pr_err("%s: prtd is null\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| if (prtd->audio_client == NULL) { |
| pr_err("%s: audio_client is null\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| event_data = (struct msm_adsp_event_data *)ucontrol->value.bytes.data; |
| if ((event_data->event_type < ADSP_STREAM_PP_EVENT) || |
| (event_data->event_type >= ADSP_STREAM_EVENT_MAX)) { |
| pr_err("%s: invalid event_type=%d", |
| __func__, event_data->event_type); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| if ((sizeof(struct msm_adsp_event_data) + event_data->payload_len) >= |
| sizeof(ucontrol->value.bytes.data)) { |
| pr_err("%s param length=%d exceeds limit", |
| __func__, event_data->payload_len); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| ret = q6asm_send_stream_cmd(prtd->audio_client, event_data); |
| if (ret < 0) |
| pr_err("%s: failed to send stream event cmd, err = %d\n", |
| __func__, ret); |
| done: |
| return ret; |
| } |
| |
| static int msm_compr_ion_fd_map_put(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| unsigned long fe_id = kcontrol->private_value; |
| struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| snd_soc_component_get_drvdata(comp); |
| struct snd_compr_stream *cstream = NULL; |
| struct msm_compr_audio *prtd; |
| int fd; |
| int ret = 0; |
| |
| if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| pr_err("%s Received out of bounds invalid fe_id %lu\n", |
| __func__, fe_id); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| cstream = pdata->cstream[fe_id]; |
| if (cstream == NULL) { |
| pr_err("%s cstream is null\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| prtd = cstream->runtime->private_data; |
| if (!prtd) { |
| pr_err("%s: prtd is null\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| if (prtd->audio_client == NULL) { |
| pr_err("%s: audio_client is null\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| memcpy(&fd, ucontrol->value.bytes.data, sizeof(fd)); |
| ret = q6asm_send_ion_fd(prtd->audio_client, fd); |
| if (ret < 0) |
| pr_err("%s: failed to register ion fd\n", __func__); |
| done: |
| return ret; |
| } |
| |
| static int msm_compr_rtic_event_ack_put(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| unsigned long fe_id = kcontrol->private_value; |
| struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| snd_soc_component_get_drvdata(comp); |
| struct snd_compr_stream *cstream = NULL; |
| struct msm_compr_audio *prtd; |
| int ret = 0; |
| int param_length = 0; |
| |
| if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| pr_err("%s Received invalid fe_id %lu\n", |
| __func__, fe_id); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| cstream = pdata->cstream[fe_id]; |
| if (cstream == NULL) { |
| pr_err("%s cstream is null\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| prtd = cstream->runtime->private_data; |
| if (!prtd) { |
| pr_err("%s: prtd is null\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| if (prtd->audio_client == NULL) { |
| pr_err("%s: audio_client is null\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| memcpy(¶m_length, ucontrol->value.bytes.data, |
| sizeof(param_length)); |
| if ((param_length + sizeof(param_length)) |
| >= sizeof(ucontrol->value.bytes.data)) { |
| pr_err("%s param length=%d exceeds limit", |
| __func__, param_length); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| ret = q6asm_send_rtic_event_ack(prtd->audio_client, |
| ucontrol->value.bytes.data + sizeof(param_length), |
| param_length); |
| if (ret < 0) |
| pr_err("%s: failed to send rtic event ack, err = %d\n", |
| __func__, ret); |
| done: |
| return ret; |
| } |
| |
| static int msm_compr_gapless_put(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| snd_soc_component_get_drvdata(comp); |
| pdata->use_dsp_gapless_mode = ucontrol->value.integer.value[0]; |
| pr_debug("%s: value: %ld\n", __func__, |
| ucontrol->value.integer.value[0]); |
| |
| return 0; |
| } |
| |
| static int msm_compr_gapless_get(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| struct msm_compr_pdata *pdata = |
| snd_soc_component_get_drvdata(comp); |
| pr_debug("%s:gapless mode %d\n", __func__, pdata->use_dsp_gapless_mode); |
| ucontrol->value.integer.value[0] = pdata->use_dsp_gapless_mode; |
| |
| return 0; |
| } |
| |
| static const struct snd_kcontrol_new msm_compr_gapless_controls[] = { |
| SOC_SINGLE_EXT("Compress Gapless Playback", |
| 0, 0, 1, 0, |
| msm_compr_gapless_get, |
| msm_compr_gapless_put), |
| }; |
| |
| static int msm_compr_probe(struct snd_soc_platform *platform) |
| { |
| struct msm_compr_pdata *pdata; |
| int i; |
| int rc; |
| const char *qdsp_version; |
| |
| pr_debug("%s\n", __func__); |
| pdata = (struct msm_compr_pdata *) |
| kzalloc(sizeof(*pdata), GFP_KERNEL); |
| if (!pdata) |
| return -ENOMEM; |
| |
| snd_soc_platform_set_drvdata(platform, pdata); |
| |
| for (i = 0; i < MSM_FRONTEND_DAI_MAX; i++) { |
| pdata->volume[i][0] = COMPRESSED_LR_VOL_MAX_STEPS; |
| pdata->volume[i][1] = COMPRESSED_LR_VOL_MAX_STEPS; |
| pdata->audio_effects[i] = NULL; |
| pdata->dec_params[i] = NULL; |
| pdata->cstream[i] = NULL; |
| pdata->ch_map[i] = NULL; |
| } |
| |
| snd_soc_add_platform_controls(platform, msm_compr_gapless_controls, |
| ARRAY_SIZE(msm_compr_gapless_controls)); |
| |
| rc = of_property_read_string(platform->dev->of_node, |
| "qcom,adsp-version", &qdsp_version); |
| if (!rc) { |
| if (!strcmp(qdsp_version, "MDSP 1.2")) |
| pdata->use_legacy_api = true; |
| else |
| pdata->use_legacy_api = false; |
| } else |
| pdata->use_legacy_api = false; |
| |
| pr_debug("%s: use legacy api %d\n", __func__, pdata->use_legacy_api); |
| /* |
| * use_dsp_gapless_mode part of platform data(pdata) is updated from HAL |
| * through a mixer control before compress driver is opened. The mixer |
| * control is used to decide if dsp gapless mode needs to be enabled. |
| * Gapless is disabled by default. |
| */ |
| pdata->use_dsp_gapless_mode = false; |
| return 0; |
| } |
| |
| static int msm_compr_volume_info(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| uinfo->count = 2; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = COMPRESSED_LR_VOL_MAX_STEPS; |
| return 0; |
| } |
| |
| static int msm_compr_audio_effects_config_info(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| uinfo->count = MAX_PP_PARAMS_SZ; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = 0xFFFFFFFF; |
| return 0; |
| } |
| |
| static int msm_compr_query_audio_effect_info(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| uinfo->count = 128; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = 0xFFFFFFFF; |
| return 0; |
| } |
| |
| static int msm_compr_dec_params_info(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| uinfo->count = 128; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = 0xFFFFFFFF; |
| return 0; |
| } |
| |
| static int msm_compr_app_type_cfg_info(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| uinfo->count = 5; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = 0xFFFFFFFF; |
| return 0; |
| } |
| |
| static int msm_compr_channel_map_info(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| uinfo->count = 8; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = 0xFFFFFFFF; |
| return 0; |
| } |
| |
| static int msm_compr_add_volume_control(struct snd_soc_pcm_runtime *rtd) |
| { |
| const char *mixer_ctl_name = "Compress Playback"; |
| const char *deviceNo = "NN"; |
| const char *suffix = "Volume"; |
| char *mixer_str = NULL; |
| int ctl_len; |
| struct snd_kcontrol_new fe_volume_control[1] = { |
| { |
| .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| .name = "?", |
| .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | |
| SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| .info = msm_compr_volume_info, |
| .tlv.p = msm_compr_vol_gain, |
| .get = msm_compr_volume_get, |
| .put = msm_compr_volume_put, |
| .private_value = 0, |
| } |
| }; |
| |
| if (!rtd) { |
| pr_err("%s NULL rtd\n", __func__); |
| return 0; |
| } |
| pr_debug("%s: added new compr FE with name %s, id %d, cpu dai %s, device no %d\n", |
| __func__, rtd->dai_link->name, rtd->dai_link->id, |
| rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1 + |
| strlen(suffix) + 1; |
| mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| if (!mixer_str) { |
| pr_err("failed to allocate mixer ctrl str of len %d", ctl_len); |
| return 0; |
| } |
| snprintf(mixer_str, ctl_len, "%s %d %s", mixer_ctl_name, |
| rtd->pcm->device, suffix); |
| fe_volume_control[0].name = mixer_str; |
| fe_volume_control[0].private_value = rtd->dai_link->id; |
| pr_debug("Registering new mixer ctl %s", mixer_str); |
| snd_soc_add_platform_controls(rtd->platform, fe_volume_control, |
| ARRAY_SIZE(fe_volume_control)); |
| kfree(mixer_str); |
| return 0; |
| } |
| |
| static int msm_compr_add_audio_effects_control(struct snd_soc_pcm_runtime *rtd) |
| { |
| const char *mixer_ctl_name = "Audio Effects Config"; |
| const char *deviceNo = "NN"; |
| char *mixer_str = NULL; |
| int ctl_len; |
| struct snd_kcontrol_new fe_audio_effects_config_control[1] = { |
| { |
| .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| .name = "?", |
| .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| .info = msm_compr_audio_effects_config_info, |
| .get = msm_compr_audio_effects_config_get, |
| .put = msm_compr_audio_effects_config_put, |
| .private_value = 0, |
| } |
| }; |
| |
| |
| if (!rtd) { |
| pr_err("%s NULL rtd\n", __func__); |
| return 0; |
| } |
| |
| pr_debug("%s: added new compr FE with name %s, id %d, cpu dai %s, device no %d\n", |
| __func__, rtd->dai_link->name, rtd->dai_link->id, |
| rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| |
| ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| |
| if (!mixer_str) |
| return 0; |
| |
| snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| |
| fe_audio_effects_config_control[0].name = mixer_str; |
| fe_audio_effects_config_control[0].private_value = rtd->dai_link->id; |
| pr_debug("Registering new mixer ctl %s\n", mixer_str); |
| snd_soc_add_platform_controls(rtd->platform, |
| fe_audio_effects_config_control, |
| ARRAY_SIZE(fe_audio_effects_config_control)); |
| kfree(mixer_str); |
| return 0; |
| } |
| |
| static int msm_compr_add_query_audio_effect_control( |
| struct snd_soc_pcm_runtime *rtd) |
| { |
| const char *mixer_ctl_name = "Query Audio Effect Param"; |
| const char *deviceNo = "NN"; |
| char *mixer_str = NULL; |
| int ctl_len; |
| struct snd_kcontrol_new fe_query_audio_effect_control[1] = { |
| { |
| .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| .name = "?", |
| .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| .info = msm_compr_query_audio_effect_info, |
| .get = msm_compr_query_audio_effect_get, |
| .put = msm_compr_query_audio_effect_put, |
| .private_value = 0, |
| } |
| }; |
| if (!rtd) { |
| pr_err("%s NULL rtd\n", __func__); |
| return 0; |
| } |
| pr_debug("%s: added new compr FE with name %s, id %d, cpu dai %s, device no %d\n", |
| __func__, rtd->dai_link->name, rtd->dai_link->id, |
| rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| if (!mixer_str) { |
| pr_err("failed to allocate mixer ctrl str of len %d", ctl_len); |
| return 0; |
| } |
| snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| fe_query_audio_effect_control[0].name = mixer_str; |
| fe_query_audio_effect_control[0].private_value = rtd->dai_link->id; |
| pr_debug("%s: registering new mixer ctl %s\n", __func__, mixer_str); |
| snd_soc_add_platform_controls(rtd->platform, |
| fe_query_audio_effect_control, |
| ARRAY_SIZE(fe_query_audio_effect_control)); |
| kfree(mixer_str); |
| return 0; |
| } |
| |
| static int msm_compr_add_audio_adsp_stream_cmd_control( |
| struct snd_soc_pcm_runtime *rtd) |
| { |
| const char *mixer_ctl_name = DSP_STREAM_CMD; |
| const char *deviceNo = "NN"; |
| char *mixer_str = NULL; |
| int ctl_len = 0, ret = 0; |
| struct snd_kcontrol_new fe_audio_adsp_stream_cmd_config_control[1] = { |
| { |
| .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| .name = "?", |
| .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| .info = msm_adsp_stream_cmd_info, |
| .put = msm_compr_adsp_stream_cmd_put, |
| .private_value = 0, |
| } |
| }; |
| |
| if (!rtd) { |
| pr_err("%s NULL rtd\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| if (!mixer_str) { |
| ret = -ENOMEM; |
| goto done; |
| } |
| |
| snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| fe_audio_adsp_stream_cmd_config_control[0].name = mixer_str; |
| fe_audio_adsp_stream_cmd_config_control[0].private_value = |
| rtd->dai_link->id; |
| pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| ret = snd_soc_add_platform_controls(rtd->platform, |
| fe_audio_adsp_stream_cmd_config_control, |
| ARRAY_SIZE(fe_audio_adsp_stream_cmd_config_control)); |
| if (ret < 0) |
| pr_err("%s: failed to add ctl %s. err = %d\n", |
| __func__, mixer_str, ret); |
| |
| kfree(mixer_str); |
| done: |
| return ret; |
| } |
| |
| static int msm_compr_add_audio_adsp_stream_callback_control( |
| struct snd_soc_pcm_runtime *rtd) |
| { |
| const char *mixer_ctl_name = DSP_STREAM_CALLBACK; |
| const char *deviceNo = "NN"; |
| char *mixer_str = NULL; |
| int ctl_len = 0, ret = 0; |
| struct snd_kcontrol *kctl; |
| |
| struct snd_kcontrol_new fe_audio_adsp_callback_config_control[1] = { |
| { |
| .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| .name = "?", |
| .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| .info = msm_adsp_stream_callback_info, |
| .get = msm_adsp_stream_callback_get, |
| .private_value = 0, |
| } |
| }; |
| |
| if (!rtd) { |
| pr_err("%s: rtd is NULL\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| if (!mixer_str) { |
| ret = -ENOMEM; |
| goto done; |
| } |
| |
| snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| fe_audio_adsp_callback_config_control[0].name = mixer_str; |
| fe_audio_adsp_callback_config_control[0].private_value = |
| rtd->dai_link->id; |
| pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| ret = snd_soc_add_platform_controls(rtd->platform, |
| fe_audio_adsp_callback_config_control, |
| ARRAY_SIZE(fe_audio_adsp_callback_config_control)); |
| if (ret < 0) { |
| pr_err("%s: failed to add ctl %s. err = %d\n", |
| __func__, mixer_str, ret); |
| ret = -EINVAL; |
| goto free_mixer_str; |
| } |
| |
| kctl = snd_soc_card_get_kcontrol(rtd->card, mixer_str); |
| if (!kctl) { |
| pr_err("%s: failed to get kctl %s.\n", __func__, mixer_str); |
| ret = -EINVAL; |
| goto free_mixer_str; |
| } |
| |
| kctl->private_data = NULL; |
| |
| free_mixer_str: |
| kfree(mixer_str); |
| done: |
| return ret; |
| } |
| |
| static int msm_compr_add_dec_runtime_params_control( |
| struct snd_soc_pcm_runtime *rtd) |
| { |
| const char *mixer_ctl_name = "Audio Stream"; |
| const char *deviceNo = "NN"; |
| const char *suffix = "Dec Params"; |
| char *mixer_str = NULL; |
| int ctl_len; |
| struct snd_kcontrol_new fe_dec_params_control[1] = { |
| { |
| .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| .name = "?", |
| .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| .info = msm_compr_dec_params_info, |
| .get = msm_compr_dec_params_get, |
| .put = msm_compr_dec_params_put, |
| .private_value = 0, |
| } |
| }; |
| |
| if (!rtd) { |
| pr_err("%s NULL rtd\n", __func__); |
| return 0; |
| } |
| |
| pr_debug("%s: added new compr FE with name %s, id %d, cpu dai %s, device no %d\n", |
| __func__, rtd->dai_link->name, rtd->dai_link->id, |
| rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| |
| ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1 + |
| strlen(suffix) + 1; |
| mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| |
| if (!mixer_str) |
| return 0; |
| |
| snprintf(mixer_str, ctl_len, "%s %d %s", mixer_ctl_name, |
| rtd->pcm->device, suffix); |
| |
| fe_dec_params_control[0].name = mixer_str; |
| fe_dec_params_control[0].private_value = rtd->dai_link->id; |
| pr_debug("Registering new mixer ctl %s", mixer_str); |
| snd_soc_add_platform_controls(rtd->platform, |
| fe_dec_params_control, |
| ARRAY_SIZE(fe_dec_params_control)); |
| kfree(mixer_str); |
| return 0; |
| } |
| |
| static int msm_compr_add_app_type_cfg_control(struct snd_soc_pcm_runtime *rtd) |
| { |
| const char *playback_mixer_ctl_name = "Audio Stream"; |
| const char *capture_mixer_ctl_name = "Audio Stream Capture"; |
| const char *deviceNo = "NN"; |
| const char *suffix = "App Type Cfg"; |
| char *mixer_str = NULL; |
| int ctl_len; |
| struct snd_kcontrol_new fe_app_type_cfg_control[1] = { |
| { |
| .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| .name = "?", |
| .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| .info = msm_compr_app_type_cfg_info, |
| .put = msm_compr_playback_app_type_cfg_put, |
| .get = msm_compr_playback_app_type_cfg_get, |
| .private_value = 0, |
| } |
| }; |
| |
| if (!rtd) { |
| pr_err("%s NULL rtd\n", __func__); |
| return 0; |
| } |
| |
| pr_debug("%s: added new compr FE ctl with name %s, id %d, cpu dai %s, device no %d\n", |
| __func__, rtd->dai_link->name, rtd->dai_link->id, |
| rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) |
| ctl_len = strlen(playback_mixer_ctl_name) + 1 + strlen(deviceNo) |
| + 1 + strlen(suffix) + 1; |
| else |
| ctl_len = strlen(capture_mixer_ctl_name) + 1 + strlen(deviceNo) |
| + 1 + strlen(suffix) + 1; |
| |
| mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| |
| if (!mixer_str) |
| return 0; |
| |
| if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) |
| snprintf(mixer_str, ctl_len, "%s %d %s", |
| playback_mixer_ctl_name, rtd->pcm->device, suffix); |
| else |
| snprintf(mixer_str, ctl_len, "%s %d %s", |
| capture_mixer_ctl_name, rtd->pcm->device, suffix); |
| |
| fe_app_type_cfg_control[0].name = mixer_str; |
| fe_app_type_cfg_control[0].private_value = rtd->dai_link->id; |
| |
| if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) { |
| fe_app_type_cfg_control[0].put = |
| msm_compr_playback_app_type_cfg_put; |
| fe_app_type_cfg_control[0].get = |
| msm_compr_playback_app_type_cfg_get; |
| } else { |
| fe_app_type_cfg_control[0].put = |
| msm_compr_capture_app_type_cfg_put; |
| fe_app_type_cfg_control[0].get = |
| msm_compr_capture_app_type_cfg_get; |
| } |
| pr_debug("Registering new mixer ctl %s", mixer_str); |
| snd_soc_add_platform_controls(rtd->platform, |
| fe_app_type_cfg_control, |
| ARRAY_SIZE(fe_app_type_cfg_control)); |
| kfree(mixer_str); |
| return 0; |
| } |
| |
| static int msm_compr_add_channel_map_control(struct snd_soc_pcm_runtime *rtd) |
| { |
| const char *mixer_ctl_name = "Playback Channel Map"; |
| const char *deviceNo = "NN"; |
| char *mixer_str = NULL; |
| struct msm_compr_pdata *pdata = NULL; |
| int ctl_len; |
| struct snd_kcontrol_new fe_channel_map_control[1] = { |
| { |
| .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| .name = "?", |
| .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| .info = msm_compr_channel_map_info, |
| .get = msm_compr_channel_map_get, |
| .put = msm_compr_channel_map_put, |
| .private_value = 0, |
| } |
| }; |
| |
| if (!rtd) { |
| pr_err("%s: NULL rtd\n", __func__); |
| return -EINVAL; |
| } |
| |
| pr_debug("%s: added new compr FE with name %s, id %d, cpu dai %s, device no %d\n", |
| __func__, rtd->dai_link->name, rtd->dai_link->id, |
| rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| |
| ctl_len = strlen(mixer_ctl_name) + strlen(deviceNo) + 1; |
| mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| |
| if (!mixer_str) |
| return -ENOMEM; |
| |
| snprintf(mixer_str, ctl_len, "%s%d", mixer_ctl_name, rtd->pcm->device); |
| |
| fe_channel_map_control[0].name = mixer_str; |
| fe_channel_map_control[0].private_value = rtd->dai_link->id; |
| pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| snd_soc_add_platform_controls(rtd->platform, |
| fe_channel_map_control, |
| ARRAY_SIZE(fe_channel_map_control)); |
| |
| pdata = snd_soc_platform_get_drvdata(rtd->platform); |
| pdata->ch_map[rtd->dai_link->id] = |
| kzalloc(sizeof(struct msm_compr_ch_map), GFP_KERNEL); |
| if (!pdata->ch_map[rtd->dai_link->id]) { |
| pr_err("%s: Could not allocate memory for channel map\n", |
| __func__); |
| kfree(mixer_str); |
| return -ENOMEM; |
| } |
| kfree(mixer_str); |
| return 0; |
| } |
| |
| static int msm_compr_add_io_fd_cmd_control(struct snd_soc_pcm_runtime *rtd) |
| { |
| const char *mixer_ctl_name = "Playback ION FD"; |
| const char *deviceNo = "NN"; |
| char *mixer_str = NULL; |
| int ctl_len = 0, ret = 0; |
| struct snd_kcontrol_new fe_ion_fd_config_control[1] = { |
| { |
| .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| .name = "?", |
| .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| .info = msm_adsp_stream_cmd_info, |
| .put = msm_compr_ion_fd_map_put, |
| .private_value = 0, |
| } |
| }; |
| |
| if (!rtd) { |
| pr_err("%s NULL rtd\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| if (!mixer_str) { |
| ret = -ENOMEM; |
| goto done; |
| } |
| |
| snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| fe_ion_fd_config_control[0].name = mixer_str; |
| fe_ion_fd_config_control[0].private_value = rtd->dai_link->id; |
| pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| ret = snd_soc_add_platform_controls(rtd->platform, |
| fe_ion_fd_config_control, |
| ARRAY_SIZE(fe_ion_fd_config_control)); |
| if (ret < 0) |
| pr_err("%s: failed to add ctl %s\n", __func__, mixer_str); |
| |
| kfree(mixer_str); |
| done: |
| return ret; |
| } |
| |
| static int msm_compr_add_event_ack_cmd_control(struct snd_soc_pcm_runtime *rtd) |
| { |
| const char *mixer_ctl_name = "Playback Event Ack"; |
| const char *deviceNo = "NN"; |
| char *mixer_str = NULL; |
| int ctl_len = 0, ret = 0; |
| struct snd_kcontrol_new fe_event_ack_config_control[1] = { |
| { |
| .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| .name = "?", |
| .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| .info = msm_adsp_stream_cmd_info, |
| .put = msm_compr_rtic_event_ack_put, |
| .private_value = 0, |
| } |
| }; |
| |
| if (!rtd) { |
| pr_err("%s NULL rtd\n", __func__); |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| if (!mixer_str) { |
| ret = -ENOMEM; |
| goto done; |
| } |
| |
| snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| fe_event_ack_config_control[0].name = mixer_str; |
| fe_event_ack_config_control[0].private_value = rtd->dai_link->id; |
| pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| ret = snd_soc_add_platform_controls(rtd->platform, |
| fe_event_ack_config_control, |
| ARRAY_SIZE(fe_event_ack_config_control)); |
| if (ret < 0) |
| pr_err("%s: failed to add ctl %s\n", __func__, mixer_str); |
| |
| kfree(mixer_str); |
| done: |
| return ret; |
| } |
| |
| static int msm_compr_new(struct snd_soc_pcm_runtime *rtd) |
| { |
| int rc; |
| |
| rc = msm_compr_add_volume_control(rtd); |
| if (rc) |
| pr_err("%s: Could not add Compr Volume Control\n", __func__); |
| |
| rc = msm_compr_add_audio_effects_control(rtd); |
| if (rc) |
| pr_err("%s: Could not add Compr Audio Effects Control\n", |
| __func__); |
| |
| rc = msm_compr_add_audio_adsp_stream_cmd_control(rtd); |
| if (rc) |
| pr_err("%s: Could not add Compr ADSP Stream Cmd Control\n", |
| __func__); |
| |
| rc = msm_compr_add_audio_adsp_stream_callback_control(rtd); |
| if (rc) |
| pr_err("%s: Could not add Compr ADSP Stream Callback Control\n", |
| __func__); |
| |
| rc = msm_compr_add_io_fd_cmd_control(rtd); |
| if (rc) |
| pr_err("%s: Could not add Compr ion fd Control\n", |
| __func__); |
| |
| rc = msm_compr_add_event_ack_cmd_control(rtd); |
| if (rc) |
| pr_err("%s: Could not add Compr event ack Control\n", |
| __func__); |
| |
| rc = msm_compr_add_query_audio_effect_control(rtd); |
| if (rc) |
| pr_err("%s: Could not add Compr Query Audio Effect Control\n", |
| __func__); |
| |
| rc = msm_compr_add_dec_runtime_params_control(rtd); |
| if (rc) |
| pr_err("%s: Could not add Compr Dec runtime params Control\n", |
| __func__); |
| rc = msm_compr_add_app_type_cfg_control(rtd); |
| if (rc) |
| pr_err("%s: Could not add Compr App Type Cfg Control\n", |
| __func__); |
| rc = msm_compr_add_channel_map_control(rtd); |
| if (rc) |
| pr_err("%s: Could not add Compr Channel Map Control\n", |
| __func__); |
| return 0; |
| } |
| |
| static struct snd_compr_ops msm_compr_ops = { |
| .open = msm_compr_open, |
| .free = msm_compr_free, |
| .trigger = msm_compr_trigger, |
| .pointer = msm_compr_pointer, |
| .set_params = msm_compr_set_params, |
| .set_metadata = msm_compr_set_metadata, |
| .get_metadata = msm_compr_get_metadata, |
| .set_next_track_param = msm_compr_set_next_track_param, |
| .ack = msm_compr_ack, |
| .copy = msm_compr_copy, |
| .get_caps = msm_compr_get_caps, |
| .get_codec_caps = msm_compr_get_codec_caps, |
| }; |
| |
| static struct snd_soc_platform_driver msm_soc_platform = { |
| .probe = msm_compr_probe, |
| .compr_ops = &msm_compr_ops, |
| .pcm_new = msm_compr_new, |
| }; |
| |
| static int msm_compr_dev_probe(struct platform_device *pdev) |
| { |
| |
| pr_debug("%s: dev name %s\n", __func__, dev_name(&pdev->dev)); |
| return snd_soc_register_platform(&pdev->dev, |
| &msm_soc_platform); |
| } |
| |
| static int msm_compr_remove(struct platform_device *pdev) |
| { |
| snd_soc_unregister_platform(&pdev->dev); |
| return 0; |
| } |
| |
| static const struct of_device_id msm_compr_dt_match[] = { |
| {.compatible = "qcom,msm-compress-dsp"}, |
| {} |
| }; |
| MODULE_DEVICE_TABLE(of, msm_compr_dt_match); |
| |
| static struct platform_driver msm_compr_driver = { |
| .driver = { |
| .name = "msm-compress-dsp", |
| .owner = THIS_MODULE, |
| .of_match_table = msm_compr_dt_match, |
| }, |
| .probe = msm_compr_dev_probe, |
| .remove = msm_compr_remove, |
| }; |
| |
| static int __init msm_soc_platform_init(void) |
| { |
| return platform_driver_register(&msm_compr_driver); |
| } |
| module_init(msm_soc_platform_init); |
| |
| static void __exit msm_soc_platform_exit(void) |
| { |
| platform_driver_unregister(&msm_compr_driver); |
| } |
| module_exit(msm_soc_platform_exit); |
| |
| MODULE_DESCRIPTION("Compress Offload platform driver"); |
| MODULE_LICENSE("GPL v2"); |