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

[alsa-devel] [PATCH] ASoC: nau8825: extend FLL function
by John Hsu 10 Mar '16

10 Mar '16

Extend FLL clock source from MCLK/BCLK/FS. Modify FLL calculation and parameter setting. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 128 ++++++++++++++++++++++++++++++++------------- sound/soc/codecs/nau8825.h | 21 ++++++-- 2 files changed, 108 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index c1b87c5..504c969 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -31,7 +31,7 @@ #include "nau8825.h" #define NAU_FREF_MAX 13500000 -#define NAU_FVCO_MAX 100000000 +#define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 struct nau8825_fll { @@ -874,8 +874,8 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, struct nau8825_fll *fll_param) { - u64 fvco; - unsigned int fref, i; + u64 fvco, fvco_max; + unsigned int fref, i, fvco_sel; /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. @@ -900,18 +900,23 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fll_param->ratio = fll_ratio[i].val; /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. - * FDCO must be within the 90MHz - 100MHz or the FFL cannot be + * FDCO must be within the 90MHz - 124MHz or the FFL cannot be * guaranteed across the full range of operation. * FDCO = freq_out * 2 * mclk_src_scaling */ + fvco_max = 0; + fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { fvco = 256 * fs * 2 * mclk_src_scaling[i].param; - if (NAU_FVCO_MIN < fvco && fvco < NAU_FVCO_MAX) - break; + if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && + fvco_max < fvco) { + fvco_max = fvco; + fvco_sel = i; + } } - if (i == ARRAY_SIZE(mclk_src_scaling)) + if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) return -EINVAL; - fll_param->mclk_src = mclk_src_scaling[i].val; + fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional * input based on FDCO, FREF and FLL ratio. @@ -926,7 +931,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, struct nau8825_fll *fll_param) { regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src); + NAU8825_CLK_SRC_MASK | NAU8825_CLK_MCLK_SRC_MASK, + NAU8825_CLK_SRC_MCLK | fll_param->mclk_src); regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, NAU8825_FLL_RATIO_MASK, fll_param->ratio); /* FLL 16-bit fractional input */ @@ -939,10 +945,25 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, - NAU8825_FLL_FILTER_SW_MASK, 0x0000); - /* FLL sigma delta modulator enable */ - regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, - NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); + NAU8825_FLL_CLK_SW_MASK, NAU8825_FLL_CLK_SW_REF); + /* Disable free-running mode */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + if (fll_param->fll_frac) { + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_MASK, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_FILTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); + } else { + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_MASK, NAU8825_FLL_FTR_SW_ACCU); + regmap_update_bits(nau8825->regmap, + NAU8825_REG_FLL6, NAU8825_SDM_EN, 0); + } } /* freq_out must be 256*Fs in order to achieve the best performance */ @@ -970,6 +991,36 @@ static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source, return 0; } +static int nau8825_mclk_prepare(struct nau8825 *nau8825, unsigned int freq) +{ + int ret = 0; + + /* We selected MCLK source but the clock itself managed externally */ + if (!nau8825->mclk) + return 0; + + if (!nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + return ret; + } + } + + if (nau8825->mclk_freq != freq) { + nau8825->mclk_freq = freq; + + freq = clk_round_rate(nau8825->mclk, freq); + ret = clk_set_rate(nau8825->mclk, freq); + if (ret) { + dev_err(nau8825->dev, "Unable to set mclk rate\n"); + return ret; + } + } + + return 0; +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -981,29 +1032,9 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); - - /* We selected MCLK source but the clock itself managed externally */ - if (!nau8825->mclk) - break; - - if (!nau8825->mclk_freq) { - ret = clk_prepare_enable(nau8825->mclk); - if (ret) { - dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); - return ret; - } - } - - if (nau8825->mclk_freq != freq) { - nau8825->mclk_freq = freq; - - freq = clk_round_rate(nau8825->mclk, freq); - ret = clk_set_rate(nau8825->mclk, freq); - if (ret) { - dev_err(nau8825->dev, "Unable to set mclk rate\n"); - return ret; - } - } + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; break; case NAU8825_CLK_INTERNAL: @@ -1011,7 +1042,32 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, NAU8825_DCO_EN); regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; + case NAU8825_CLK_FLL_MCLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_MCLK); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; + case NAU8825_CLK_FLL_BLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_BLK); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + break; + case NAU8825_CLK_FLL_FS: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_FS); if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index dff8edb..8237693 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -112,20 +112,28 @@ /* FLL3 (0x06) */ #define NAU8825_FLL_INTEGER_MASK (0x3ff << 0) +#define NAU8825_FLL_CLK_SRC_SFT 10 +#define NAU8825_FLL_CLK_SRC_MASK (0x3 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_MCLK (0 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_BLK (0x2 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_FS (0x3 << NAU8825_FLL_CLK_SRC_SFT) /* FLL4 (0x07) */ #define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) /* FLL5 (0x08) */ -#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) +#define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8825_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8825_FLL_CLK_SW_MASK (0x1 << 13) +#define NAU8825_FLL_CLK_SW_N2 (0x1 << 13) +#define NAU8825_FLL_CLK_SW_REF (0x0 << 13) +#define NAU8825_FLL_FTR_SW_MASK (0x1 << 12) +#define NAU8825_FLL_FTR_SW_ACCU (0x1 << 12) +#define NAU8825_FLL_FTR_SW_FILTER (0x0 << 12) /* FLL6 (0x9) */ -#define NAU8825_DCO_EN_MASK (0x1 << 15) #define NAU8825_DCO_EN (0x1 << 15) -#define NAU8825_DCO_DIS (0x0 << 15) -#define NAU8825_SDM_EN_MASK (0x1 << 14) #define NAU8825_SDM_EN (0x1 << 14) -#define NAU8825_SDM_DIS (0x0 << 14) /* HSD_CTRL (0xc) */ #define NAU8825_HSD_AUTO_MODE (1 << 6) @@ -306,6 +314,9 @@ enum { NAU8825_CLK_MCLK = 0, NAU8825_CLK_INTERNAL, + NAU8825_CLK_FLL_MCLK, + NAU8825_CLK_FLL_BLK, + NAU8825_CLK_FLL_FS, }; struct nau8825 { -- 2.6.4

2 5

[alsa-devel] [PATCH v6 0/6] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 10 Mar '16

10 Mar '16

There is currently two other patch series[1][2] that depend on the three first (ALSA-) patches of this series. It would be nice to get at least those three applied sooner rather than later, so please tell me if I can do anything to get those in. BTW, I applied the relevant patches from the two series to on top of this one. Obviously the two series conflicted with each other a bit and also the IEC958 helper patch from [2] conflicted with the IEC958 patch from my series (which is weird since Arnaud's series depends on my series...). After resolving the conflicts everything compiled fine, but for some reason neither the 'ELD' or IEC958 Playback Default' appeared on my BBB. I'll try to find time to debug this further at some point. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105509.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105502.html Changes since v5 - Rebased on top of the latest drm-next branch - Allow 32 bit samplewidth in snd_pcm_create_iec958_consumer() and snd_pcm_create_iec958_consumer_hw_params() - Propose new simpler DT binding for tda998x audio - Squash tda998x audio DT binding together with hdmi-codec integration Changes since RFC v4, - Rebased on top of the latest drm-next branch - Split the hdmi-codec abort functionality into a separate patch for better visibility of what it is all about - This does not affect the tda998x patches as the abort functionality is not used - Drop S18_3* formats from I2S_FORMATS and add a comment about formats not supported by HDMI Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 Jyri Sarha (6): ALSA: pcm: add IEC958 channel status helper for hw_params ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add audio abort() callback for video side to use drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 18 + arch/arm/boot/dts/am335x-boneblack.dts | 71 +++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 271 ++++++++++++-- include/drm/i2c/tda998x.h | 24 +- include/dt-bindings/display/tda998x.h | 7 + include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 53 ++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec.c | 411 +++++++++++++++++++++ 12 files changed, 898 insertions(+), 72 deletions(-) create mode 100644 include/dt-bindings/display/tda998x.h create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c -- 1.9.1

5 16

[alsa-devel] [PATCH] ASoC: hdac_hdmi: Fix infoframe programming
by Subhransu S. Prusty 10 Mar '16

10 Mar '16

Audio infoframe used incorrect buffer, so fix it. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 7851dd2..4f7bd1f 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -303,7 +303,6 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac, if (ret < 0) return ret; - dip = (u8 *)&frame; break; case DRM_ELD_CONN_TYPE_DP: @@ -332,9 +331,9 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac, /* Fill infoframe. Index auto-incremented */ hdac_hdmi_set_dip_index(hdac, pin_nid, 0x0, 0x0); if (conn_type == DRM_ELD_CONN_TYPE_HDMI) { - for (i = 0; i < sizeof(frame); i++) + for (i = 0; i < sizeof(buffer); i++) snd_hdac_codec_write(&hdac->hdac, pin_nid, 0, - AC_VERB_SET_HDMI_DIP_DATA, dip[i]); + AC_VERB_SET_HDMI_DIP_DATA, buffer[i]); } else { for (i = 0; i < sizeof(dp_ai); i++) snd_hdac_codec_write(&hdac->hdac, pin_nid, 0, -- 1.9.1

2 1

[alsa-devel] [PATCH 0/4] ALSA: dice: enable to handle several streams
by Takashi Sakamoto 09 Mar '16

09 Mar '16

Hi, This patchset updates my previous RFT, and goes for merging. [alsa-devel] [RFT][PATCH 0/4] ALSA: dice: enabled to handle several streams http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105387.html For the previous one, Stefan Richter worked as a tester with his devices (Focusrite Saffire Pro 24 and 40), while current patchset includes changes related to userspace interface. Therefore, I don't add Tested-by. In previous one, ALSA dice driver uses subdevice for userspace to access any PCM substreams. On the other hand, the design of PCM subdevice leaves a bit cumbersome for userspace to handle several subdevices simultaneously in one process. These PCM substreams on a batch of AMDTP streams and it's better for ALSA dice driver to enable one process to handle them with simpler way. Based on this concept, this patchset adds PCM character devices, instead of PCM subdevice. Thus, userspace applications can utilize PCM substreams via individual PCM character devices. In configuration space of alsa-lib, via 'hw:0,0' and 'hw:0,1' if available, like: $ arecord -l ... card 1: Pro26000f5a [Pro26-000f5a], device 0: DICE [Pro26-000f5a] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Pro26000f5a [Pro26-000f5a], device 1: DICE [Pro26-000f5a] Subdevices: 1/1 Subdevice #0: subdevice #0 For the other parts, there's no code changes. Therefore, Stefan's result still has meanings. Changes: * Add additional PCM character device instead of PCM subdevice * Correct string entries to force-pcm * Improve comments Takashi Sakamoto (4): ALSA: dice: have two sets of isochronous resources/streams ALSA: dice: handle whole available isochronous streams ALSA: dice: handle several PCM substreams when any isochronous streams are available ALSA: dice: force to add two pcm devices for listed models sound/firewire/dice/dice-midi.c | 8 +- sound/firewire/dice/dice-pcm.c | 147 +++++++++----- sound/firewire/dice/dice-stream.c | 398 ++++++++++++++++++++++++-------------- sound/firewire/dice/dice.c | 41 ++++ sound/firewire/dice/dice.h | 33 +++- 5 files changed, 424 insertions(+), 203 deletions(-) -- 2.7.0

3 8

[alsa-devel] Scheduled alsa-lib release?
by Takashi Iwai 09 Mar '16

09 Mar '16

Hi, in the previous meeting, there have been requests about the regular "predictable" release of alsa-lib. We thought of aligning with kernel release cycle, judging from the nature of alsa-lib. alsa-lib 1.1.0 was released in the beginning of November, which was just after 4.3 kernel release. Now 4.5 kernel release will be likely in the next week or one week after that. Jaroslav, how about releasing alsa-lib 1.1.1 around that? There are a few pending patches, but they should be available soon. thanks, Takashi

4 6

[alsa-devel] alsa-plugins: pcm_a52.c uses recently removed avcodec_{alloc, free}_frame
by Daniel Kirchner 09 Mar '16

09 Mar '16

pcm_a52.c uses avcodec_alloc_frame and avcodec_free_frame that have been replaced by av_frame_alloc and av_frame_free in recent versions of libavcodec. The following patch should fix this while remaining backwards compatible. --- a/a52/pcm_a52.c.orig 2016-03-04 18:21:12.606805770 +0100 +++ b/a52/pcm_a52.c 2016-03-04 18:34:22.653240501 +0100 @@ -62,6 +62,11 @@ #define AV_CODEC_ID_AC3 CODEC_ID_AC3 #endif +#if LIBAVCODEC_VERSION_INT < 0x371c01 +#define av_frame_alloc avcodec_alloc_frame +#define av_frame_free avcodec_free_frame +#endif + struct a52_ctx { snd_pcm_ioplug_t io; snd_pcm_t *slave; @@ -513,7 +518,7 @@ rec->inbuf = NULL; } #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0) - avcodec_free_frame(&rec->frame); + av_frame_free(&rec->frame); #else av_freep(&rec->frame); #endif @@ -557,7 +562,7 @@ { struct a52_ctx *rec = io->private_data; #ifdef USE_AVCODEC_FRAME - rec->frame = avcodec_alloc_frame(); + rec->frame = av_frame_alloc(); if (!rec->frame) return -ENOMEM; if (av_samples_alloc(rec->frame->data, rec->frame->linesize,

2 1

[alsa-devel] [PATCH] Show sequencer sound card numer/PID via aconnect
by Martin Koegler 09 Mar '16

09 Mar '16

From: Martin Koegler <martin.koegler(a)chello.at> rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. This patch adds support for displaying the sound card number/PID to aconnect. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- configure.ac | 9 +++++++++ seq/aconnect/aconnect.c | 17 ++++++++++++++++- 2 files changed, 25 insertions(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index bdb133c..9e48190 100644 --- a/configure.ac +++ b/configure.ac @@ -41,6 +41,15 @@ AC_CHECK_HEADERS([alsa/topology.h], [have_topology="yes"], [have_topology="no"], AC_CHECK_HEADERS([samplerate.h], [have_samplerate="yes"], [have_samplerate="no"], [#include <samplerate.h>]) +AC_CHECK_LIB([asound], [snd_seq_client_info_get_card], [HAVE_SEQ_CLIENT_INFO_GET_CARD="yes"]) +if test "$HAVE_SEQ_CLIENT_INFO_GET_CARD" = "yes" ; then + AC_DEFINE([HAVE_SEQ_CLIENT_INFO_GET_CARD], 1, [alsa-lib supports snd_seq_client_info_get_card]) +fi +AC_CHECK_LIB([asound], [snd_seq_client_info_get_pid], [HAVE_SEQ_CLIENT_INFO_GET_PID="yes"]) +if test "$HAVE_SEQ_CLIENT_INFO_GET_PID" = "yes" ; then + AC_DEFINE([HAVE_SEQ_CLIENT_INFO_GET_PID], 1, [alsa-lib supports snd_seq_client_info_get_pid]) +fi + AM_CONDITIONAL(HAVE_PCM, test "$have_pcm" = "yes") AM_CONDITIONAL(HAVE_MIXER, test "$have_mixer" = "yes") AM_CONDITIONAL(HAVE_RAWMIDI, test "$have_rawmidi" = "yes") diff --git a/seq/aconnect/aconnect.c b/seq/aconnect/aconnect.c index 8d6cebb..85a7770 100644 --- a/seq/aconnect/aconnect.c +++ b/seq/aconnect/aconnect.c @@ -166,11 +166,26 @@ static void print_port(snd_seq_t *seq, snd_seq_client_info_t *cinfo, snd_seq_port_info_t *pinfo, int count) { if (! count) { - printf(_("client %d: '%s' [type=%s]\n"), + int card = -1, pid = -1; + + printf(_("client %d: '%s' [type=%s"), snd_seq_client_info_get_client(cinfo), snd_seq_client_info_get_name(cinfo), (snd_seq_client_info_get_type(cinfo) == SND_SEQ_USER_CLIENT ? _("user") : _("kernel"))); + +#ifdef HAVE_SEQ_CLIENT_INFO_GET_CARD + card = snd_seq_client_info_get_card(cinfo); +#endif + if (card != -1) + printf(",card=%d", card); + +#ifdef HAVE_SEQ_CLIENT_INFO_GET_PID + pid = snd_seq_client_info_get_pid(cinfo); +#endif + if (pid != -1) + printf(",pid=%d", pid); + printf("]\n"); } printf(" %3d '%-16s'\n", snd_seq_port_info_get_port(pinfo), -- 2.1.4

2 1

[alsa-devel] alsa-plugins: pcm_a52.c uses recently removed avcodec_{alloc, free}_frame
by Daniel Kirchner 09 Mar '16

09 Mar '16

pcm_a52.c uses avcodec_alloc_frame and avcodec_free_frame that have been replaced by av_frame_alloc and av_frame_free in recent versions of libavcodec. The following patch should fix this while remaining backwards compatible. --- a/a52/pcm_a52.c.orig 2016-03-04 18:21:12.606805770 +0100 +++ b/a52/pcm_a52.c 2016-03-04 18:34:22.653240501 +0100 @@ -62,6 +62,11 @@ #define AV_CODEC_ID_AC3 CODEC_ID_AC3 #endif +#if LIBAVCODEC_VERSION_INT < 0x371c01 +#define av_frame_alloc avcodec_alloc_frame +#define av_frame_free avcodec_free_frame +#endif + struct a52_ctx { snd_pcm_ioplug_t io; snd_pcm_t *slave; @@ -513,7 +518,7 @@ rec->inbuf = NULL; } #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0) - avcodec_free_frame(&rec->frame); + av_frame_free(&rec->frame); #else av_freep(&rec->frame); #endif @@ -557,7 +562,7 @@ { struct a52_ctx *rec = io->private_data; #ifdef USE_AVCODEC_FRAME - rec->frame = avcodec_alloc_frame(); + rec->frame = av_frame_alloc(); if (!rec->frame) return -ENOMEM; if (av_samples_alloc(rec->frame->data, rec->frame->linesize,

1 0

[alsa-devel] [PATCH] ASoC: rt5640: remove unused variable
by Sudip Mukherjee 09 Mar '16

09 Mar '16

From: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> We are getting build warning about: sound/soc/codecs/rt5640.c:1892:11: warning: unused variable 'dai_sel' The use of the variable was removed but the variable itself was not removed. Fixes: c467fc0e010b ("ASoC: rt5640: Set PLL src according to source") Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/soc/codecs/rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index af9b5f1..e8b5ba0 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1889,7 +1889,7 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); struct rl6231_pll_code pll_code; - int ret, dai_sel; + int ret; if (source == rt5640->pll_src && freq_in == rt5640->pll_in && freq_out == rt5640->pll_out) -- 1.9.1

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[alsa-devel] [PATCH v4] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 08 Mar '16

08 Mar '16

This is an initial imx-wm8958 device-tree-only machine driver. The sound card driver support three dai link, HIFI, VOICE and BT. We can configure the cpu dai from device tree, if present it will create corresponding sound card subdevice. So at least one cpu dai phandle should be set from device tree. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v3->v4: remove the dummy cpu dai, support set cpu dai1/2/3 from device tree. When cpu-dai1/2/3 present it will create croresponding sound card subdevice. At least one cpu dai should be configured. .../devicetree/bindings/sound/imx-audio-wm8958.txt | 56 ++++ arch/arm/configs/imx_v6_v7_defconfig | 1 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 327 +++++++++++++++++++++ 5 files changed, 400 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..c013287 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,56 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - audio-codec : The phandle of the WM8958 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being + the connection's sink, the second being the + connection's source. Valid names could be power + supplies, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + +Optional properties: + + - cpu-dai1 : The phandle of CPU DAI1 controller. At least one + cpu dai is required. + + - cpu-dai2 : The phandle of CPU DAI2 controller. At least one + cpu dai is required. + + - cpu-dai3 : The phandle of CPU DAI3 controller. At least one + cpu dai is required. + + - fsl,cpu-dai1-master : If present, cpu dai1 works as master, and will + provide bit clock and frame clock. Otherwise, + cpu dai1 works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-evk-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8958-audio"; + cpu-dai1 = <&sai1>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; +}; diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index 978c5de..7ac5856 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -251,6 +251,7 @@ CONFIG_SND_SOC_FSL_ASRC=y CONFIG_SND_IMX_SOC=y CONFIG_SND_SOC_PHYCORE_AC97=y CONFIG_SND_SOC_EUKREA_TLV320=y +CONFIG_SND_SOC_IMX_WM8958=y CONFIG_SND_SOC_IMX_WM8962=y CONFIG_SND_SOC_IMX_SGTL5000=y CONFIG_SND_SOC_IMX_SPDIF=y diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..2a781c0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..0e8a9b2 --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,327 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../fsl/fsl_sai.h" +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define HIFI_DAI (0) +#define VOICE_DAI (1) +#define BT_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == HIFI_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == HIFI_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == HIFI_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link dai_link[DAI_LINK_NUM]; + int num_links; + struct snd_soc_card card; + struct clk *mclk[WM8958_MCLK_MAX]; + u32 mclk_freq[WM8958_MCLK_MAX]; + bool is_cpu_dai1_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int imx_wm8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == HIFI_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == HIFI_DAI) { + if (data->is_cpu_dai1_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (data->is_cpu_dai1_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == HIFI_DAI && data->is_cpu_dai1_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_ops imx_voice_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [HIFI_DAI] = { + .name = "HiFi", + .stream_name = "HiFi", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_hifi_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [VOICE_DAI] = { + .name = "Voice", + .stream_name = "Voice", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .ops = &imx_voice_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [BT_DAI] = { + .name = "Bluetooth", + .stream_name = "Bluetooth", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np[DAI_LINK_NUM], *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev[DAI_LINK_NUM]; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + char tmp[8]; + int ret, i; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + for (i = 0; i < DAI_LINK_NUM; i++) { + sprintf(tmp, "cpu-dai%d", i + 1); + cpu_np[i] = of_parse_phandle(np, tmp, 0); + if (cpu_np[i]) { + cpu_pdev[i] = of_find_device_by_node(cpu_np[i]); + if (!cpu_pdev[i]) { + dev_err(&pdev->dev, + "failed to get cpu dai%d platform device\n", i); + ret = -EINVAL; + goto fail; + } + + data->dai_link[data->num_links] = imx_wm8958_dai_link[i]; + data->dai_link[data->num_links].cpu_dai_name = dev_name(&cpu_pdev[i]->dev); + data->dai_link[data->num_links++].platform_of_node = cpu_np[i]; + } + } + + if (!data->num_links) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + /* Hifi dai link support codec master and slave mode */ + if (of_property_read_bool(pdev->dev.of_node, "fsl,cpu-dai1-master")) { + data->is_cpu_dai1_master = true; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else { + data->is_cpu_dai1_master = false; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "MCLK%d", i + 1); + data->mclk[i] = devm_clk_get(&codec_dev->dev, tmp); + if (!IS_ERR(data->mclk[i])) + data->mclk_freq[i] = clk_get_rate(data->mclk[i]); + } + + if (!data->mclk_freq[0] && !data->mclk_freq[1]) { + dev_err(&pdev->dev, "failed to get mclk clock\n"); + ret = -EINVAL; + goto fail; + } + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = data->num_links; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + +fail: + of_node_put(cpu_np[HIFI_DAI]); + of_node_put(cpu_np[VOICE_DAI]); + of_node_put(cpu_np[BT_DAI]); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

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