// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) // // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // // Copyright(c) 2018 Intel Corporation // // Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com> // Ranjani Sridharan <ranjani.sridharan@linux.intel.com> // Rander Wang <rander.wang@intel.com> // Keyon Jie <yang.jie@linux.intel.com> //
/* * Hardware interface for generic Intel audio DSP HDA IP
*/
u32 hda_dsp_get_mult_div(struct snd_sof_dev *sdev, int rate)
{ switch (rate) { case 8000: return SDnFMT_DIV(6); case 9600: return SDnFMT_DIV(5); case 11025: return SDnFMT_BASE(1) | SDnFMT_DIV(4); case 16000: return SDnFMT_DIV(3); case 22050: return SDnFMT_BASE(1) | SDnFMT_DIV(2); case 32000: return SDnFMT_DIV(3) | SDnFMT_MULT(2); case 44100: return SDnFMT_BASE(1); case 48000: return 0; case 88200: return SDnFMT_BASE(1) | SDnFMT_MULT(2); case 96000: return SDnFMT_MULT(2); case 176400: return SDnFMT_BASE(1) | SDnFMT_MULT(4); case 192000: return SDnFMT_MULT(4); default:
dev_warn(sdev->dev, "can't find div rate %d using 48kHz\n",
rate); return 0; /* use 48KHz if not found */
}
};
u32 hda_dsp_get_bits(struct snd_sof_dev *sdev, int sample_bits)
{ switch (sample_bits) { case 8: return SDnFMT_BITS(0); case 16: return SDnFMT_BITS(1); case 20: return SDnFMT_BITS(2); case 24: return SDnFMT_BITS(3); case 32: return SDnFMT_BITS(4); default:
dev_warn(sdev->dev, "can't find %d bits using 16bit\n",
sample_bits); return SDnFMT_BITS(1); /* use 16bits format if not found */
}
};
/* * Use the codec required format val (which is link_bps adjusted) when * the DSP is not in use
*/ if (!sdev->dspless_mode_selected) {
u32 rate = hda_dsp_get_mult_div(sdev, params_rate(params));
u32 bits = hda_dsp_get_bits(sdev, params_width(params));
spcm = snd_sof_find_spcm_dai(scomp, rtd); if (!spcm) {
dev_warn_ratelimited(sdev->dev, "warn: can't find PCM with DAI ID %d\n",
rtd->dai_link->id); return 0;
}
if (hda && !hda->no_ipc_position) { /* read position from IPC position */
pos = spcm->stream[substream->stream].posn.host_posn; goto found;
}
spcm = snd_sof_find_spcm_dai(scomp, rtd); if (!spcm) {
dev_err(sdev->dev, "error: can't find PCM with DAI ID %d\n", rtd->dai_link->id); return -EINVAL;
}
/* * if we want the .ack to work, we need to prevent the control from being mapped. * The status can still be mapped.
*/ if (hda_disable_rewinds)
runtime->hw.info |= SNDRV_PCM_INFO_NO_REWINDS | SNDRV_PCM_INFO_SYNC_APPLPTR;
/* * All playback streams are DMI L1 capable, capture streams need * pause push/release to be disabled
*/ if (hda_always_enable_dmi_l1 && direction == SNDRV_PCM_STREAM_CAPTURE)
runtime->hw.info &= ~SNDRV_PCM_INFO_PAUSE;
/* * Do not advertise the PAUSE support if it is forced to be disabled via * module parameter or if the pause_supported is false for the PCM * device
*/ if (hda_force_pause_support == 0 ||
(hda_force_pause_support == -1 &&
!spcm->stream[substream->stream].pause_supported))
runtime->hw.info &= ~SNDRV_PCM_INFO_PAUSE;
if (hda_always_enable_dmi_l1 ||
direction == SNDRV_PCM_STREAM_PLAYBACK ||
spcm->stream[substream->stream].d0i3_compatible)
flags |= SOF_HDA_STREAM_DMI_L1_COMPATIBLE;
dsp_stream = hda_dsp_stream_get(sdev, direction, flags); if (!dsp_stream) {
dev_err(sdev->dev, "error: no stream available\n"); return -ENODEV;
}
/* minimum as per HDA spec */
snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 4);
/* avoid circular buffer wrap in middle of period */
snd_pcm_hw_constraint_integer(substream->runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
/* Limit the maximum number of periods to not exceed the BDL entries count */ if (runtime->hw.periods_max > HDA_DSP_MAX_BDL_ENTRIES)
snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS,
runtime->hw.periods_min,
HDA_DSP_MAX_BDL_ENTRIES);
/* Only S16 and S32 supported by HDA hardware when used without DSP */ if (sdev->dspless_mode_selected)
snd_pcm_hw_constraint_mask64(substream->runtime, SNDRV_PCM_HW_PARAM_FORMAT,
SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32);
/* * The dsp_max_burst_size_in_ms is the length of the maximum burst size * of the host DMA in the ALSA buffer. * * On playback start the DMA will transfer dsp_max_burst_size_in_ms * amount of data in one initial burst to fill up the host DMA buffer. * Consequent DMA burst sizes are shorter and their length can vary. * To avoid immediate xrun by the initial burst we need to place * constraint on the period size (via PERIOD_TIME) to cover the size of * the host buffer. * We need to add headroom of max 10ms as the firmware needs time to * settle to the 1ms pacing and initially it can run faster for few * internal periods. * * On capture the DMA will transfer 1ms chunks.
*/ if (spcm->stream[direction].dsp_max_burst_size_in_ms) { unsignedint period_time = spcm->stream[direction].dsp_max_burst_size_in_ms;
/* * add headroom over the maximum burst size to cover the time * needed for the DMA pace to settle. * Limit the headroom time to HDA_MAX_PERIOD_TIME_HEADROOM
*/
period_time += min(period_time, HDA_MAX_PERIOD_TIME_HEADROOM);
/* * Reset the llp cache values (they are used for LLP compensation in * case the counter is not reset)
*/
dsp_stream->pplcllpl = 0;
dsp_stream->pplcllpu = 0;
int hda_dsp_pcm_close(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream)
{ struct hdac_stream *hstream = substream->runtime->private_data; int direction = substream->stream; int ret;
ret = hda_dsp_stream_put(sdev, direction, hstream->stream_tag);
if (ret) {
dev_dbg(sdev->dev, "stream %s not opened!\n", substream->name); return -ENODEV;
}
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