[PATCH 0/2] ASoC: fsl_sai: Fill Tx FIFO to avoid initial underruns
This series fixes initial underruns that can occur in the TX queue of the fsl_sai interface when starting playback. These patches are around here for quite some time and have proven useful. Time to upstream them.
Sascha
Ahmad Fatoum (2): ASoC: fsl_sai: refactor TDM slots calculation into helper function ASoC: fsl_sai: Fill Tx FIFO to avoid initial underruns
sound/soc/fsl/fsl_sai.c | 38 +++++++++++++++++++++++++++++++++----- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 34 insertions(+), 5 deletions(-)
From: Ahmad Fatoum a.fatoum@pengutronix.de
Splitting the calculation between the initializer and later on makes it harder to follow. A follow-up commit will also need to do this calculation, so move it into a helper function. No functional change.
Signed-off-by: Ahmad Fatoum a.fatoum@pengutronix.de --- sound/soc/fsl/fsl_sai.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-)
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index e3105d48fb651..36f6115469843 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -516,6 +516,19 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) return 0; }
+static unsigned int fsl_sai_get_tdm_slots(struct fsl_sai *sai, + unsigned int channels, + unsigned int slot_width) +{ + if (sai->slots) + return sai->slots; + + if (sai->bclk_ratio) + return sai->bclk_ratio / slot_width; + + return channels == 1 ? 2 : channels; +} + static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -531,7 +544,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, int dl_cfg_cnt = sai->dl_cfg_cnt; u32 dl_type = FSL_SAI_DL_I2S; u32 val_cr4 = 0, val_cr5 = 0; - u32 slots = (channels == 1) ? 2 : channels; + u32 slots; u32 slot_width = word_width; int adir = tx ? RX : TX; u32 pins, bclk; @@ -541,10 +554,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, if (sai->slot_width) slot_width = sai->slot_width;
- if (sai->slots) - slots = sai->slots; - else if (sai->bclk_ratio) - slots = sai->bclk_ratio / slot_width; + slots = fsl_sai_get_tdm_slots(sai, channels, slot_width);
pins = DIV_ROUND_UP(channels, slots);
On Thu, Jun 29, 2023 at 03:58:19PM +0200, Sascha Hauer wrote:
From: Ahmad Fatoum a.fatoum@pengutronix.de
Splitting the calculation between the initializer and later on makes it harder to follow. A follow-up commit will also need to do this calculation, so move it into a helper function. No functional change.
Signed-off-by: Ahmad Fatoum a.fatoum@pengutronix.de
You've not provided a Signed-off-by for this so I can't do anything with it, please see Documentation/process/submitting-patches.rst for details on what this is and why it's important.
From: Ahmad Fatoum a.fatoum@pengutronix.de
JACK handles XRuns by stopping and start the ALSA device. On occasion, this leads to early underruns on start leading to reorderd output channels.
By filling the FIFO initially, we can avoid these early underruns. This is also suggested by the i.MX8MM reference manual:
"If the Transmit FIFO is empty, then to avoid a FIFO underrun, the Transmit Data Register must be written at least 3 bit clocks before the start of the next unmasked word. Before enabling the transmitter, the Transmit FIFO should be initialized with data (since after the transmitter is enabled, the transmitter will start a new frame, and if no data is in the FIFO, then the transmitter will immediately give an error)"
[1]: Rev. 0, 02/2019, 13.9.3.5.2 FIFO pointers Fixes: 435508214942 ("ASoC: Add SAI SoC Digital Audio Interface driver.") Signed-off-by: Ahmad Fatoum a.fatoum@pengutronix.de --- sound/soc/fsl/fsl_sai.c | 18 ++++++++++++++++++ sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 19 insertions(+)
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 36f6115469843..6a4f990110d91 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -755,6 +755,21 @@ static void fsl_sai_config_disable(struct fsl_sai *sai, int dir) } }
+static void fsl_sai_tx_fill_fifo(struct fsl_sai *sai, + struct snd_pcm_runtime *runtime) +{ + u32 slots, slot_width, pins; + int i; + + slot_width = sai->slot_width ?: snd_pcm_format_physical_width(runtime->format); + + slots = fsl_sai_get_tdm_slots(sai, runtime->channels, slot_width); + pins = DIV_ROUND_UP(runtime->channels, slots); + + for (i = 0; i < runtime->channels; i++) + regmap_write(sai->regmap, FSL_SAI_TDR(i % pins), 0x0); +} + static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { @@ -784,6 +799,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* Fill FIFO to avoid initial underruns */ + if (tx) + fsl_sai_tx_fill_fifo(sai, substream->runtime); regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE);
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index a53c4f0e25faf..66a136d97a441 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -34,6 +34,7 @@ #define FSL_SAI_TDR5 0x34 /* SAI Transmit Data 5 */ #define FSL_SAI_TDR6 0x38 /* SAI Transmit Data 6 */ #define FSL_SAI_TDR7 0x3C /* SAI Transmit Data 7 */ +#define FSL_SAI_TDR(ofs) (FSL_SAI_TDR0 + (ofs) * 4) #define FSL_SAI_TFR0 0x40 /* SAI Transmit FIFO 0 */ #define FSL_SAI_TFR1 0x44 /* SAI Transmit FIFO 1 */ #define FSL_SAI_TFR2 0x48 /* SAI Transmit FIFO 2 */
participants (2)
-
Mark Brown -
Sascha Hauer