February 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH RFC 0/6] ASoC: sgtl5000: fix use of regulators and internal LDO
by Eric Nelson 15 Jun '16

15 Jun '16

This patch set addresses a structural problem in the handling of regulators for VDDIO, VDDA, and VDDD in the SGTL5000 driver. The first two of these power rails must be powered on prior to any I2C communication, and yet the regulators were tied to the codec, which is instantiated only after a fair amount of I2C communication takes place. In other words, these regulators could never have function, and we can surmise that no user of this driver has switched power supply rails connected to them. The third power rail (VDDD) can be derived internally (by using I2C registers) though the data sheet says that if an external VDDD is used, it should be enabled before MCLK is started and I2C activity begins. Patch 1 moves the regulators to the I2C device and initializes them before I2C activity. Patch 2 addresses an issue found during development of the patch and is somewhat unrelated to regulators. It is included for discussion Patch 3 removes the root cause of patch 2 Patch 4 adds return value checks when setting the LINREG and ANA_POWER registers Patch 5 addresses the cause of follow-on failures and the source of the error in patch 3. Patch 6 removes the naive use of the supplies to lower power consumption and replaces it with something more modest. The patch set is being sent as an RFC for discussion, and should probably be squashed into two or three patches covering the regulator updates (patches 1, 3-5 and perhaps 6), initialization error handling (patch 2). Eric Nelson (6): ASoC: sgtl5000: fix regulator support ASoC: sgtl5000: write all default registers ASoC: sgtl5000: initialize CHIP_ANA_POWER to power-on defaults ASoC: sgtl5000: check return values ASoC: sgtl5000: disable internal PLL early ASoC: sgtl5000: Don't disable regulators in SND_SOC_BIAS_OFF sound/soc/codecs/sgtl5000.c | 417 +++++++++++--------------------------------- sound/soc/codecs/sgtl5000.h | 2 + 2 files changed, 102 insertions(+), 317 deletions(-) -- 1.9.1

3 28

[alsa-devel] [PATCH v2 0/6] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 13 May '16

13 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Version 2: - fixing a missing patch - checkpatch.pl - ... well, learning how to send a patchset. Sorry. ;) Bugs are detected and fixed on a imx6sl platform with linux 4.3, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 74 +++++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 63 insertions(+), 11 deletions(-) -- 1.9.1

5 11

[alsa-devel] [PATCH 0/5] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 13 May '16

13 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Bugs are detected and fixed on a imx6sl platform with linux 4.3, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (5): ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 69 ++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 60 insertions(+), 9 deletions(-) -- 1.9.1

4 17

[alsa-devel] [PATCH v3 0/6] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 03 May '16

03 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Version 3: - rebase + fix + test on top of linux 4.4 + 'for-next' Version 2: - fixing a missing patch - checkpatch.pl - ... well, learning how to send a patchset. Sorry. Bugs are detected and fixed on a imx6sl platform with linux 4.4 + 'for-next' branch, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Similar tests were done on a imx6solo platform by Caleb Crome (He will propose a patch concerning fifo depth to fix DMA xrun at highest bitrates) Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 81 ++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 70 insertions(+), 11 deletions(-) -- 1.9.1

6 20

[alsa-devel] [PATCH 1/2] ASoC: rockchip: i2s: separate capture and playback
by John Keeping 29 Apr '16

29 Apr '16

If we only clear the tx/rx state when both are disabled it is not possible to start/stop one multiple times while the other is running. Since the two are independently controlled, treat them as such and remove the false dependency between capture and playback. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/rockchip/rockchip_i2s.c | 72 +++++++++++++++++---------------------- 1 file changed, 32 insertions(+), 40 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 83b1b9c..acc6225 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -82,8 +82,8 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); + I2S_XFER_TXS_START, + I2S_XFER_TXS_START); i2s->tx_start = true; } else { @@ -92,27 +92,23 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); - if (!i2s->rx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START, + I2S_XFER_TXS_STOP); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC, + I2S_CLR_TXC); - regmap_read(i2s->regmap, I2S_CLR, &val); + regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ - while (val) { - regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; - } + /* Should wait for clear operation to finish */ + while (val & I2S_CLR_TXC) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; } } } @@ -128,8 +124,8 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); + I2S_XFER_RXS_START, + I2S_XFER_RXS_START); i2s->rx_start = true; } else { @@ -138,27 +134,23 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); - if (!i2s->tx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_RXS_START, + I2S_XFER_RXS_STOP); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_RXC, + I2S_CLR_RXC); - regmap_read(i2s->regmap, I2S_CLR, &val); + regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ - while (val) { - regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; - } + /* Should wait for clear operation to finish */ + while (val & I2S_CLR_RXC) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; } } } -- 2.6.3.462.gbe2c914

3 5

[alsa-devel] [PATCH] ASoC: Make aux_dev more like a generic component
by mengdong.lin＠linux.intel.com 25 Apr '16

25 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> aux_dev is mainly used by the machine driver to specify analog devices, which are registered as codecs. Making it more like a generic component can help the machine driver to use it to specify any component with topology info by name. Details: - Remove the stub 'rtd_aux' array from the soc card. - Add a list 'aux_comp_list' to store the components of aux_devs. And add a list head 'list_aux' to struct snd_soc_component, for adding such components to the above list. - Add a 'init' ops to a component for machine specific init. soc_bind_aux_dev() will set it to be aux_dev's init. And it will be called when probing the component. - soc_bind_aux_dev() will also search components by name of an aux_dev, since it may not be a codec. - Move probing of aux_devs before checking new DAI links brought by topology. - Move removal of aux_devs later than removal of links. Because topology of aux components may register DAIs and the DAI drivers will go with removal of the aux components, we want soc_remove_link_dais() to remove the DAIs at first. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 49011c3..c03e07a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -798,6 +798,7 @@ struct snd_soc_component { unsigned int registered_as_component:1; struct list_head list; + struct list_head list_aux; /* for auxiliary component of the card */ struct snd_soc_dai_driver *dai_drv; int num_dai; @@ -841,6 +842,9 @@ struct snd_soc_component { int (*probe)(struct snd_soc_component *); void (*remove)(struct snd_soc_component *); + /* machine specific init */ + int (*init)(struct snd_soc_component *component); + #ifdef CONFIG_DEBUG_FS void (*init_debugfs)(struct snd_soc_component *component); const char *debugfs_prefix; @@ -1144,8 +1148,7 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; - struct snd_soc_pcm_runtime *rtd_aux; - int num_aux_rtd; + struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1553,6 +1556,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); + INIT_LIST_HEAD(&card->aux_comp_list); } static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e03e82..790ee2b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1413,6 +1413,16 @@ static int soc_probe_component(struct snd_soc_card *card, component->name); } + /* machine specific init */ + if (component->init) { + ret = component->init(component); + if (ret < 0) { + dev_err(component->dev, + "Failed to do machine specific init %d\n", ret); + goto err_probe; + } + } + if (component->controls) snd_soc_add_component_controls(component, component->controls, component->num_controls); @@ -1657,65 +1667,81 @@ static int soc_probe_link_dais(struct snd_soc_card *card, static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - const char *name = aux_dev->codec_name; - - rtd->component = soc_find_component(aux_dev->codec_of_node, name); - if (!rtd->component) { - if (aux_dev->codec_of_node) - name = of_node_full_name(aux_dev->codec_of_node); - - dev_err(card->dev, "ASoC: %s not registered\n", name); - return -EPROBE_DEFER; + struct snd_soc_component *component; + const char *name; + struct device_node *codec_of_node; + + if (aux_dev->codec_of_node || aux_dev->codec_name) { + /* codecs, usually analog devices */ + name = aux_dev->codec_name; + codec_of_node = aux_dev->codec_of_node; + component = soc_find_component(codec_of_node, name); + if (!component) { + if (codec_of_node) + name = of_node_full_name(codec_of_node); + goto err_defer; + } + } else if (aux_dev->name) { + /* generic components */ + name = aux_dev->name; + component = soc_find_component(NULL, name); + if (!component) + goto err_defer; + } else { + dev_err(card->dev, "ASoC: Invalid auxiliary device\n"); + return -EINVAL; } - /* - * Some places still reference rtd->codec, so we have to keep that - * initialized if the component is a CODEC. Once all those references - * have been removed, this code can be removed as well. - */ - rtd->codec = rtd->component->codec; - + component->init = aux_dev->init; + list_add(&component->list_aux, &card->aux_comp_list); return 0; + +err_defer: + dev_err(card->dev, "ASoC: %s not registered\n", name); + return -EPROBE_DEFER; } -static int soc_probe_aux_dev(struct snd_soc_card *card, int num) +static int soc_probe_aux_devices(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; + struct snd_soc_component *comp; + int order; int ret; - ret = soc_probe_component(card, rtd->component); - if (ret < 0) - return ret; - - /* do machine specific initialization */ - if (aux_dev->init) { - ret = aux_dev->init(rtd->component); - if (ret < 0) { - dev_err(card->dev, "ASoC: failed to init %s: %d\n", - aux_dev->name, ret); - return ret; + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + list_for_each_entry(comp, &card->aux_comp_list, list_aux) { + if (comp->driver->probe_order == order) { + ret = soc_probe_component(card, comp); + if (ret < 0) { + dev_err(card->dev, + "ASoC: failed to probe aux component %s %d\n", + comp->name, ret); + return ret; + } + } } } - return soc_post_component_init(rtd, aux_dev->name); + return 0; } -static void soc_remove_aux_dev(struct snd_soc_card *card, int num) +static void soc_remove_aux_devices(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_component *component = rtd->component; + struct snd_soc_component *comp, *_comp; + int order; - /* unregister the rtd device */ - if (rtd->dev_registered) { - device_unregister(rtd->dev); - rtd->dev_registered = 0; + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + list_for_each_entry_safe(comp, _comp, + &card->aux_comp_list, list_aux) { + if (comp->driver->remove_order == order) { + soc_remove_component(comp); + /* remove it from the card's aux_comp_list */ + list_del(&comp->list_aux); + } + } } - - if (component) - soc_remove_component(component); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) @@ -1894,6 +1920,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } + /* probe auxiliary components */ + ret = soc_probe_aux_devices(card); + if (ret < 0) + goto probe_dai_err; + /* Find new DAI links added during probing components and bind them. * Components with topology may bring new DAIs and DAI links. */ @@ -1923,16 +1954,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } - for (i = 0; i < card->num_aux_devs; i++) { - ret = soc_probe_aux_dev(card, i); - if (ret < 0) { - dev_err(card->dev, - "ASoC: failed to add auxiliary devices %d\n", - ret); - goto probe_aux_dev_err; - } - } - snd_soc_dapm_link_dai_widgets(card); snd_soc_dapm_connect_dai_link_widgets(card); @@ -1992,8 +2013,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) return 0; probe_aux_dev_err: - for (i = 0; i < card->num_aux_devs; i++) - soc_remove_aux_dev(card, i); + soc_remove_aux_devices(card); probe_dai_err: soc_remove_dai_links(card); @@ -2039,20 +2059,18 @@ static int soc_probe(struct platform_device *pdev) static int soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; - int i; /* make sure any delayed work runs */ list_for_each_entry(rtd, &card->rtd_list, list) flush_delayed_work(&rtd->delayed_work); - /* remove auxiliary devices */ - for (i = 0; i < card->num_aux_devs; i++) - soc_remove_aux_dev(card, i); - /* remove and free each DAI */ soc_remove_dai_links(card); soc_remove_pcm_runtimes(card); + /* remove auxiliary devices */ + soc_remove_aux_devices(card); + soc_cleanup_card_debugfs(card); /* remove the card */ @@ -2608,16 +2626,6 @@ int snd_soc_register_card(struct snd_soc_card *card) INIT_LIST_HEAD(&card->rtd_list); card->num_rtd = 0; - card->rtd_aux = devm_kzalloc(card->dev, - sizeof(struct snd_soc_pcm_runtime) * - card->num_aux_devs, - GFP_KERNEL); - if (card->rtd_aux == NULL) - return -ENOMEM; - - for (i = 0; i < card->num_aux_devs; i++) - card->rtd_aux[i].card = card; - INIT_LIST_HEAD(&card->dapm_dirty); INIT_LIST_HEAD(&card->dobj_list); card->instantiated = 0; -- 2.5.0

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[alsa-devel] [PATCH 0/4] ASoC: simple-card/davinci-mcasp: System clock configuration support
by Peter Ujfalusi 22 Apr '16

22 Apr '16

Hi, Currently davinci-mcasp is hardwired for two configuration regarding to reference clock: McASP is master: AHCLKX is selected as source of the reference clock used to generate the FS and BCLK McASP is slave: AHCLKX is selected as output and the default AUXCLK is routed there. This works in most of the boards, but not all configuration can be supported with this limitation. McASP uses it's internal AUXCLK as a reference clock. AUXCLK can be sourced from the external AHCLKX pin or from internal functional clock. If the board uses McASP as master and AHCLKX is not receiving clock, we need to be able to select the internal clock source to be able to generate the FS and BCLK. Since we try to move all of our boards to simple-card instead of the davinci-evm custom machine driver, I have included Jyri's two patch to add the needed support for it to be able to select clocks and directions. I remember the discussion regarding to clock IDs, and to try to move to common clock framework. I have taken a look at that path and I'm not convinced that by trying to add ASoC level common clock handling will simplify or makes things cleaner/simpler. Regards, Peter --- Jyri Sarha (2): ASoC: simple-card: Add system-clock-direction DT parameter to dai nodes ASoC: simple-card: Support for selecting system clocks by ID Peter Ujfalusi (2): ASoC: davinci-mcasp: Use defines for clkdiv IDs via DT binding header ASoC: davinci-mcasp: Improve the sysclk selection .../devicetree/bindings/sound/simple-card.txt | 3 ++ include/dt-bindings/sound/ti-mcasp.h | 13 +++++++ include/sound/simple_card.h | 2 ++ sound/soc/davinci/davinci-mcasp.c | 42 ++++++++++++++++------ sound/soc/generic/simple-card.c | 17 ++++++++- 5 files changed, 66 insertions(+), 11 deletions(-) create mode 100644 include/dt-bindings/sound/ti-mcasp.h -- 2.7.1

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[alsa-devel] [PATCH v8 0/2] ASoc: sun4i-codec: Add FM, Line and Mic inputs
by Danny Milosavljevic 21 Apr '16

21 Apr '16

Hi, this is the eighth version of the patch set that adds inputs to sun4i-codec. Changes compared to v7 are: - fix the routes for line and mic capturing. The patches are on top of the branch "sunxi-next" in <git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git>. Regards, Danny --- Danny Milosavljevic (2): sun4i-codec: Add FM, Line and Mic inputs sun4i-codec: make it possible to use different codec_widgets for A10 and A20. b/sound/soc/sunxi/sun4i-codec.c | 230 +++++++++++++++++++++++++++++++++++++--- 1 file changed, 215 insertions(+), 15 deletions(-)

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[alsa-devel] Help with wm8731 support for the Utilite (imx6q)
by Jonathan Bennett 06 Apr '16

06 Apr '16

Hey, I'm working with a compulab utilite, trying to get everything working with 3.17, with the eventual goal of getting better support for it upstream. Currently fighting the sound. It's a imx6q talking to a wm8731 over i2c/i2s. I am booting Fedora 21 arm Alpha, with my own dtb ( http://pastebin.com/4a4PBKwx) and driver (http://pastebin.com/1CsesuVV) It shows up in aplay -l, but trying to play anything to it gives me a div by 0 in a driver: [<c04f27e4>] (Ldiv0) from [<bf50512c>] (fsl_ssi_hw_params+0x194/0x364 [snd_soc_fsl_ssi]) This div0 happens in sound/soc/fsl/fsl_ssi.c, in the function fsl_ssi_set_bclk. My driver file is an almost direct copy of what compulab shipped with a 3.10 kernel. This is my first attempt at kernel hacking, so any help would be appreciated. ~Jonathan Bennett

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[alsa-devel] [PATCH v2] ASoC: fsl-asoc-card: add wm8960 support
by Zidan Wang 05 Apr '16

05 Apr '16

add wm8960 support for fsl-asoc-card Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl-asoc-card.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 040362f..5aeb6ed 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -23,6 +23,7 @@ #include "../codecs/sgtl5000.h" #include "../codecs/wm8962.h" +#include "../codecs/wm8960.h" #define RX 0 #define TX 1 @@ -479,6 +480,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { + codec_dai_name = "wm8960-hifi"; + priv->card.set_bias_level = fsl_asoc_card_set_bias_level; + priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; + priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; @@ -582,6 +589,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, + { .compatible = "fsl,imx-audio-wm8960", }, {} }; -- 1.9.1

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