August 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/4] drm/i915: Add audio set_ncts callback
by libin.yang＠intel.com 11 Aug '15

11 Aug '15

From: Libin Yang <libin.yang(a)intel.com> Add the set_ncts callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..7305881 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,8 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + int (*set_ncts)(struct device *, int port, int dev_entry, + int rate); } *ops; }; -- 1.9.1

4 11

[alsa-devel] [PATCH v2] ASoC: fsl-sai: add 32 bit word length support
by Zidan Wang 11 Aug '15

11 Aug '15

Add 32 bit word length support. There are no code changes required in the SAI driver since it has already wirten the word width to the corresponding register. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0662809..40280ce 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -1,5 +1,5 @@ /* - * Copyright 2012-2013 Freescale Semiconductor, Inc. + * Copyright 2012-2015 Freescale Semiconductor, Inc. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -13,7 +13,8 @@ #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) /* SAI Register Map Register */ #define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */ -- 1.9.1

2 2

[alsa-devel] jack: Fixing a small memory leak
by Valentin Corfu 10 Aug '15

10 Aug '15

Hello Alsa developers, I observed a small memory leak in alsa-plugins (jack): ... ==1034== ==1034== 8 bytes in 1 blocks are definitely lost in loss record 7 of 108 ==1034== at 0x4027E5A: calloc (in /usr/lib/valgrind/vgpreload_memcheck-x86-linux.so) ==1034== by 0x4035102: snd_pcm_jack_prepare (in /usr/lib/alsa-lib/libasound_module_pcm_jack.so) ==1034== by 0x40D07AB: snd_pcm_ioplug_prepare (pcm_ioplug.c:145) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x409DDB4: snd_pcm_plugin_prepare (pcm_plugin.c:162) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x409DDB4: snd_pcm_plugin_prepare (pcm_plugin.c:162) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x4084A1D: snd_pcm_hw_params (pcm.c:838) ==1034== by 0x4090013: snd_pcm_set_params (pcm.c:8053) ==1034== by 0x80486EB: main (in /root/pcm_min_ok) ... ==1034== LEAK SUMMARY: ==1034== definitely lost: 8 bytes in 1 blocks ==1034== indirectly lost: 0 bytes in 0 blocks ==1034== possibly lost: 31,311 bytes in 1,630 blocks ==1034== still reachable: 5,619 bytes in 24 blocks ==1034== suppressed: 0 bytes in 0 blocks ==1034== Reachable blocks (those to which a pointer was found) are not shown. ==1034== To see them, rerun with: --leak-check=full --show-leak-kinds=all ... Best Regards, Valentin Corfu

2 3

[alsa-devel] [PATCH] ALSA: hda - Fix pin config and mapping on Alienware 15
by Takashi Iwai 10 Aug '15

10 Aug '15

Alienware 15 has CA0132 codec for its onboard sound, but the pin config and mapping seem quite different from other Creative boards. This patch corrects them, at least, for providing the right headphone and mic jack notification, as well as removing the non-existing SPDIF pins. Even with this fix, not all stuff works perfectly yet, mainly because of the badly written ca0132 driver code -- it has too many implicit assumptions of pin configs and maps. Nevertheless, this is a small good step forward. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=101981 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 87 ++++++++++++++++++++++++++++++-------------- 1 file changed, 59 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0f039abe9673..186792fe226e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -763,6 +763,20 @@ enum { QUIRK_ALIENWARE, }; +static const struct hda_pintbl alienware_pincfgs[] = { + { 0x0b, 0x90170110 }, /* Builtin Speaker */ + { 0x0c, 0x411111f0 }, /* N/A */ + { 0x0d, 0x411111f0 }, /* N/A */ + { 0x0e, 0x411111f0 }, /* N/A */ + { 0x0f, 0x0321101f }, /* HP */ + { 0x10, 0x411111f0 }, /* Headset? disabled for now */ + { 0x11, 0x03a11021 }, /* Mic */ + { 0x12, 0xd5a30140 }, /* Builtin Mic */ + { 0x13, 0x411111f0 }, /* N/A */ + { 0x18, 0x411111f0 }, /* N/A */ + {} +}; + static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15", QUIRK_ALIENWARE), {} @@ -3147,7 +3161,7 @@ static int ca0132_select_out(struct hda_codec *codec) auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; if (auto_jack) - jack_present = snd_hda_jack_detect(codec, spec->out_pins[1]); + jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_hp); else jack_present = spec->vnode_lswitch[VNID_HP_SEL - VNODE_START_NID]; @@ -3309,7 +3323,7 @@ static int ca0132_select_mic(struct hda_codec *codec) auto_jack = spec->vnode_lswitch[VNID_AMIC1_ASEL - VNODE_START_NID]; if (auto_jack) - jack_present = snd_hda_jack_detect(codec, spec->input_pins[0]); + jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_amic1); else jack_present = spec->vnode_lswitch[VNID_AMIC1_SEL - VNODE_START_NID]; @@ -4617,37 +4631,54 @@ static void ca0132_config(struct hda_codec *codec) spec->multiout.num_dacs = 3; spec->multiout.max_channels = 2; - spec->num_outputs = 2; - spec->out_pins[0] = 0x0b; /* speaker out */ if (spec->quirk == QUIRK_ALIENWARE) { codec_dbg(codec, "ca0132_config: QUIRK_ALIENWARE applied.\n"); + snd_hda_apply_pincfgs(codec, alienware_pincfgs); + + spec->num_outputs = 2; + spec->out_pins[0] = 0x0b; /* speaker out */ spec->out_pins[1] = 0x0f; - } else{ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = 0x0f; + + spec->adcs[0] = 0x7; /* digital mic / analog mic1 */ + spec->adcs[1] = 0x8; /* analog mic2 */ + spec->adcs[2] = 0xa; /* what u hear */ + + spec->num_inputs = 3; + spec->input_pins[0] = 0x12; + spec->input_pins[1] = 0x11; + spec->input_pins[2] = 0x13; + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = 0x11; + } else { + spec->num_outputs = 2; + spec->out_pins[0] = 0x0b; /* speaker out */ spec->out_pins[1] = 0x10; /* headphone out */ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = spec->out_pins[1]; + + spec->adcs[0] = 0x7; /* digital mic / analog mic1 */ + spec->adcs[1] = 0x8; /* analog mic2 */ + spec->adcs[2] = 0xa; /* what u hear */ + + spec->num_inputs = 3; + spec->input_pins[0] = 0x12; + spec->input_pins[1] = 0x11; + spec->input_pins[2] = 0x13; + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = spec->input_pins[0]; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + spec->dig_in = 0x09; + cfg->dig_in_pin = 0x0e; + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; } - spec->shared_out_nid = 0x2; - spec->unsol_tag_hp = spec->out_pins[1]; - - spec->adcs[0] = 0x7; /* digital mic / analog mic1 */ - spec->adcs[1] = 0x8; /* analog mic2 */ - spec->adcs[2] = 0xa; /* what u hear */ - - spec->num_inputs = 3; - spec->input_pins[0] = 0x12; - spec->input_pins[1] = 0x11; - spec->input_pins[2] = 0x13; - spec->shared_mic_nid = 0x7; - spec->unsol_tag_amic1 = spec->input_pins[0]; - - /* SPDIF I/O */ - spec->dig_out = 0x05; - spec->multiout.dig_out_nid = spec->dig_out; - cfg->dig_out_pins[0] = 0x0c; - cfg->dig_outs = 1; - cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; - spec->dig_in = 0x09; - cfg->dig_in_pin = 0x0e; - cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; } static int ca0132_prepare_verbs(struct hda_codec *codec) -- 2.5.0

1 0

[alsa-devel] [PATCH] ALSA: echoaudio: Use standard C definitions of true and false
by Mark Brown 10 Aug '15

10 Aug '15

The echoaudio locally defines TRUE and FALSE. Not only is this redundant given that C now has a boolean type it results in lots of warnings as other headers also define these macros, causing duplicate definitions. Fix this by removing the local defines and converting all local users to use the standard C true and false instead, simply removing the macros is less safe due to implicit inclusion of the other definitons. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/pci/echoaudio/darla20_dsp.c | 6 +++--- sound/pci/echoaudio/darla24_dsp.c | 6 +++--- sound/pci/echoaudio/echo3g_dsp.c | 16 ++++++++-------- sound/pci/echoaudio/echoaudio.c | 2 +- sound/pci/echoaudio/echoaudio.h | 7 ++----- sound/pci/echoaudio/echoaudio_3g.c | 12 ++++++------ sound/pci/echoaudio/echoaudio_dsp.c | 26 +++++++++++++------------- sound/pci/echoaudio/gina20_dsp.c | 8 ++++---- sound/pci/echoaudio/gina24_dsp.c | 22 +++++++++++----------- sound/pci/echoaudio/indigo_dsp.c | 6 +++--- sound/pci/echoaudio/indigodj_dsp.c | 6 +++--- sound/pci/echoaudio/indigodjx_dsp.c | 6 +++--- sound/pci/echoaudio/indigoio_dsp.c | 6 +++--- sound/pci/echoaudio/indigoiox_dsp.c | 6 +++--- sound/pci/echoaudio/layla20_dsp.c | 12 ++++++------ sound/pci/echoaudio/layla24_dsp.c | 26 +++++++++++++------------- sound/pci/echoaudio/mia_dsp.c | 8 ++++---- sound/pci/echoaudio/mona_dsp.c | 20 ++++++++++---------- 18 files changed, 99 insertions(+), 102 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index febee5b..320837b 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -44,18 +44,18 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_DARLA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index 7b4f6fd..8736b5e 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -44,17 +44,17 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_DARLA24_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_ESYNC; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index ae11ce1..6deb80c 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -59,8 +59,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2); chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; - chip->has_midi = TRUE; + chip->bad_board = true; + chip->has_midi = true; chip->dsp_code_to_load = FW_ECHO3G_DSP; /* Load the DSP code and the ASIC on the PCI card and get @@ -78,8 +78,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->px_analog_in = chip->bx_analog_in = 14; chip->px_digital_in = chip->bx_digital_in = 16; chip->px_num = chip->bx_num = 24; - chip->has_phantom_power = TRUE; - chip->hasnt_input_nominal_level = TRUE; + chip->has_phantom_power = true; + chip->hasnt_input_nominal_level = true; } else if (err == E3G_LAYLA3G_BOX_TYPE) { chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | @@ -106,10 +106,10 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->non_audio_spdif = FALSE; - chip->bad_board = FALSE; - chip->phantom_power = FALSE; + chip->professional_spdif = false; + chip->non_audio_spdif = false; + chip->bad_board = false; + chip->phantom_power = false; return init_line_levels(chip); } diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 862db9a..1cb85ae 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2245,7 +2245,7 @@ static int snd_echo_resume(struct device *dev) #ifdef ECHOCARD_HAS_MIDI if (chip->midi_input_enabled) - enable_midi_input(chip, TRUE); + enable_midi_input(chip, true); if (chip->midi_out) snd_echo_midi_output_trigger(chip->midi_out, 1); #endif diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 3251522..152ce15 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -153,9 +153,6 @@ #define _ECHOAUDIO_H_ -#define TRUE 1 -#define FALSE 0 - #include "echoaudio_dsp.h" @@ -378,8 +375,8 @@ struct echoaudio { */ u8 output_clock; /* Layla20 only */ char meters_enabled; /* VU-meters status */ - char asic_loaded; /* Set TRUE when ASIC loaded */ - char bad_board; /* Set TRUE if DSP won't load */ + char asic_loaded; /* Set true when ASIC loaded */ + char bad_board; /* Set true if DSP won't load */ char professional_spdif; /* 0 = consumer; 1 = professional */ char non_audio_spdif; /* 3G - only */ char digital_in_automute; /* Gina24, Layla24, Mona - only */ diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 2fa66dc..22c786b 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -41,7 +41,7 @@ static int check_asic_status(struct echoaudio *chip) return -EIO; chip->comm_page->ext_box_status = cpu_to_le32(E3G_ASIC_NOT_LOADED); - chip->asic_loaded = FALSE; + chip->asic_loaded = false; clear_handshake(chip); send_vector(chip, DSP_VC_TEST_ASIC); @@ -55,7 +55,7 @@ static int check_asic_status(struct echoaudio *chip) if (box_status == E3G_ASIC_NOT_LOADED) return -ENODEV; - chip->asic_loaded = TRUE; + chip->asic_loaded = true; return box_status & E3G_BOX_TYPE_MASK; } @@ -243,7 +243,7 @@ static int load_asic(struct echoaudio *chip) * 48 kHz, internal clock, S/PDIF RCA mode */ if (box_type >= 0) { err = write_control_reg(chip, E3G_48KHZ, - E3G_FREQ_REG_DEFAULT, TRUE); + E3G_FREQ_REG_DEFAULT, true); if (err < 0) return err; } @@ -378,16 +378,16 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) int err, incompatible_clock; /* Set clock to "internal" if it's not compatible with the new mode */ - incompatible_clock = FALSE; + incompatible_clock = false; switch (mode) { case DIGITAL_MODE_SPDIF_OPTICAL: case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) - incompatible_clock = TRUE; + incompatible_clock = true; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) - incompatible_clock = TRUE; + incompatible_clock = true; break; default: dev_err(chip->card->dev, diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 1a9427a..15aae2f 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -103,8 +103,8 @@ static int write_dsp(struct echoaudio *chip, u32 data) cond_resched(); } - chip->bad_board = TRUE; /* Set TRUE until DSP re-loaded */ - dev_dbg(chip->card->dev, "write_dsp: Set bad_board to TRUE\n"); + chip->bad_board = true; /* Set true until DSP re-loaded */ + dev_dbg(chip->card->dev, "write_dsp: Set bad_board to true\n"); return -EIO; } @@ -126,8 +126,8 @@ static int read_dsp(struct echoaudio *chip, u32 *data) cond_resched(); } - chip->bad_board = TRUE; /* Set TRUE until DSP re-loaded */ - dev_err(chip->card->dev, "read_dsp: Set bad_board to TRUE\n"); + chip->bad_board = true; /* Set true until DSP re-loaded */ + dev_err(chip->card->dev, "read_dsp: Set bad_board to true\n"); return -EIO; } @@ -166,7 +166,7 @@ static int read_sn(struct echoaudio *chip) /* This card has no ASIC, just return ok */ static inline int check_asic_status(struct echoaudio *chip) { - chip->asic_loaded = TRUE; + chip->asic_loaded = true; return 0; } @@ -341,11 +341,11 @@ static int load_dsp(struct echoaudio *chip, u16 *code) dev_warn(chip->card->dev, "DSP is already loaded!\n"); return 0; } - chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ + chip->bad_board = true; /* Set true until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ - chip->asic_loaded = FALSE; /* Loading the DSP code will reset the ASIC */ + chip->asic_loaded = false; /* Loading the DSP code will reset the ASIC */ - dev_dbg(chip->card->dev, "load_dsp: Set bad_board to TRUE\n"); + dev_dbg(chip->card->dev, "load_dsp: Set bad_board to true\n"); /* If this board requires a resident loader, install it. */ #ifdef DSP_56361 @@ -471,7 +471,7 @@ static int load_dsp(struct echoaudio *chip, u16 *code) } chip->dsp_code = code; /* Show which DSP code loaded */ - chip->bad_board = FALSE; /* DSP OK */ + chip->bad_board = false; /* DSP OK */ return 0; } udelay(100); @@ -951,10 +951,10 @@ static int rest_in_peace(struct echoaudio *chip) /* Stops all active pipes (just to be sure) */ stop_transport(chip, chip->active_mask); - set_meters_on(chip, FALSE); + set_meters_on(chip, false); #ifdef ECHOCARD_HAS_MIDI - enable_midi_input(chip, FALSE); + enable_midi_input(chip, false); #endif /* Go to sleep */ @@ -981,9 +981,9 @@ static int init_dsp_comm_page(struct echoaudio *chip) /* Init all the basic stuff */ chip->card_name = ECHOCARD_NAME; - chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ + chip->bad_board = true; /* Set true until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ - chip->asic_loaded = FALSE; + chip->asic_loaded = false; memset(chip->comm_page, 0, sizeof(struct comm_page)); /* Init the comm page */ diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index 5dafe92..b237757 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -48,19 +48,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_GINA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } @@ -69,7 +69,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { - chip->professional_spdif = FALSE; + chip->professional_spdif = false; return init_line_levels(chip); } diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index 6971766..8eff2b4 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -52,7 +52,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96 | @@ -76,7 +76,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } @@ -86,8 +86,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; + chip->professional_spdif = false; + chip->digital_in_automute = true; return init_line_levels(chip); } @@ -152,7 +152,7 @@ static int load_asic(struct echoaudio *chip) 48 kHz, internal clock, S/PDIF RCA mode */ if (!err) { control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; - err = write_control_reg(chip, control_reg, TRUE); + err = write_control_reg(chip, control_reg, true); } return err; } @@ -226,7 +226,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) chip->sample_rate = rate; dev_dbg(chip->card->dev, "set_sample_rate: %d clock %d\n", rate, clock); - return write_control_reg(chip, control_reg, FALSE); + return write_control_reg(chip, control_reg, false); } @@ -274,7 +274,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) } chip->input_clock = clock; - return write_control_reg(chip, control_reg, TRUE); + return write_control_reg(chip, control_reg, true); } @@ -285,17 +285,17 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) int err, incompatible_clock; /* Set clock to "internal" if it's not compatible with the new mode */ - incompatible_clock = FALSE; + incompatible_clock = false; switch (mode) { case DIGITAL_MODE_SPDIF_OPTICAL: case DIGITAL_MODE_SPDIF_CDROM: case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) - incompatible_clock = TRUE; + incompatible_clock = true; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) - incompatible_clock = TRUE; + incompatible_clock = true; break; default: dev_err(chip->card->dev, @@ -333,7 +333,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) break; } - err = write_control_reg(chip, control_reg, TRUE); + err = write_control_reg(chip, control_reg, true); spin_unlock_irq(&chip->lock); if (err < 0) return err; diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index 54edd67..c97dc83 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -49,16 +49,16 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 2cf5cc0..2428b35 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -49,16 +49,16 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_DJ_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 5252863..5fbd4a3 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -47,17 +47,17 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_DJX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; err = load_firmware(chip); if (err < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 4e81787..79b68ba 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -49,16 +49,16 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_IO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 6de3f9b..1ae394e 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -47,17 +47,17 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_IOX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; err = load_firmware(chip); if (err < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index f2024a3..5e5b6e2 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -51,8 +51,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; - chip->has_midi = TRUE; + chip->bad_board = true; + chip->has_midi = true; chip->dsp_code_to_load = FW_LAYLA20_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | @@ -62,7 +62,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } @@ -71,7 +71,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { - chip->professional_spdif = FALSE; + chip->professional_spdif = false; return init_line_levels(chip); } @@ -113,7 +113,7 @@ static int check_asic_status(struct echoaudio *chip) u32 asic_status; int goodcnt, i; - chip->asic_loaded = FALSE; + chip->asic_loaded = false; for (i = goodcnt = 0; i < 5; i++) { send_vector(chip, DSP_VC_TEST_ASIC); @@ -127,7 +127,7 @@ static int check_asic_status(struct echoaudio *chip) if (asic_status == ASIC_ALREADY_LOADED) { if (++goodcnt == 3) { - chip->asic_loaded = TRUE; + chip->asic_loaded = true; return 0; } } diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index 4f11e81..df28e51 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -51,8 +51,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; - chip->has_midi = TRUE; + chip->bad_board = true; + chip->has_midi = true; chip->dsp_code_to_load = FW_LAYLA24_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | @@ -64,7 +64,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; if ((err = init_line_levels(chip)) < 0) return err; @@ -77,8 +77,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; + chip->professional_spdif = false; + chip->digital_in_automute = true; return init_line_levels(chip); } @@ -135,7 +135,7 @@ static int load_asic(struct echoaudio *chip) err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC, FW_LAYLA24_2S_ASIC); if (err < 0) - return FALSE; + return false; /* Now give the external ASIC a little time to set up */ mdelay(10); @@ -147,7 +147,7 @@ static int load_asic(struct echoaudio *chip) 48 kHz, internal clock, S/PDIF RCA mode */ if (!err) err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ, - TRUE); + true); return err; } @@ -241,7 +241,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) dev_dbg(chip->card->dev, "set_sample_rate: %d clock %d\n", rate, control_reg); - return write_control_reg(chip, control_reg, FALSE); + return write_control_reg(chip, control_reg, false); } @@ -287,7 +287,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) } chip->input_clock = clock; - return write_control_reg(chip, control_reg, TRUE); + return write_control_reg(chip, control_reg, true); } @@ -334,17 +334,17 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) short asic; /* Set clock to "internal" if it's not compatible with the new mode */ - incompatible_clock = FALSE; + incompatible_clock = false; switch (mode) { case DIGITAL_MODE_SPDIF_OPTICAL: case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) - incompatible_clock = TRUE; + incompatible_clock = true; asic = FW_LAYLA24_2S_ASIC; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) - incompatible_clock = TRUE; + incompatible_clock = true; asic = FW_LAYLA24_2A_ASIC; break; default: @@ -383,7 +383,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) break; } - err = write_control_reg(chip, control_reg, TRUE); + err = write_control_reg(chip, control_reg, true); spin_unlock_irq(&chip->lock); if (err < 0) return err; diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index fdad079..8f612a0 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -52,19 +52,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_MIA_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; if ((subdevice_id & 0x0000f) == MIA_MIDI_REV) - chip->has_midi = TRUE; + chip->has_midi = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index 843c7a5..dce9e57 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -52,7 +52,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT; @@ -69,7 +69,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } @@ -79,8 +79,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; + chip->professional_spdif = false; + chip->digital_in_automute = true; return init_line_levels(chip); } @@ -148,7 +148,7 @@ static int load_asic(struct echoaudio *chip) 48 kHz, internal clock, S/PDIF RCA mode */ if (!err) { control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; - err = write_control_reg(chip, control_reg, TRUE); + err = write_control_reg(chip, control_reg, true); } return err; @@ -356,7 +356,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) } chip->input_clock = clock; - return write_control_reg(chip, control_reg, TRUE); + return write_control_reg(chip, control_reg, true); } @@ -367,16 +367,16 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) int err, incompatible_clock; /* Set clock to "internal" if it's not compatible with the new mode */ - incompatible_clock = FALSE; + incompatible_clock = false; switch (mode) { case DIGITAL_MODE_SPDIF_OPTICAL: case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) - incompatible_clock = TRUE; + incompatible_clock = true; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) - incompatible_clock = TRUE; + incompatible_clock = true; break; default: dev_err(chip->card->dev, @@ -415,7 +415,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) break; } - err = write_control_reg(chip, control_reg, FALSE); + err = write_control_reg(chip, control_reg, false); spin_unlock_irq(&chip->lock); if (err < 0) return err; -- 2.5.0

2 2

[alsa-devel] [RFC PATCH 00/10] ASoC: support adding PCM dynamically from topology
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch set allows the topology core to create PCM devices dynamically. The user can define different DAIs in the topology files for different versions of firmware, but share a generic platform and machine driver. A dummy DAI and DAI link can be used to register the soc card and specify the platform with topology. Then real DAIs are created in platform probing phase by the topology core, and the machine driver will be notified to create relavant DAI links. We tested using topology to create FE DAI/DAI links on Broadwell. Mengdong Lin (10): ASoC: change the PCM runtime array to a list ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: soc_bind_dai_link() change 2nd argument to DAI link pointer ASoC: implement DAI links in a list. ASoC: Add support for dummy DAI links and PCM runtimes ASoC: bind new DAI links after probing components. ASoC: support adding a DAI dynamically ASoC: topology: change pass number of DAI smaller than graph ASoC: Topology: change stream formats to bitwise flag ASOC: topology: Add PCM DAIs dynamically when loading them include/sound/soc-dai.h | 1 + include/sound/soc-topology.h | 1 - include/sound/soc.h | 28 +- include/uapi/sound/asoc.h | 2 +- sound/soc/fsl/fsl-asoc-card.c | 10 +- sound/soc/fsl/imx-wm8962.c | 10 +- sound/soc/generic/simple-card.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 12 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/pxa/mioa701_wm9713.c | 6 +- sound/soc/samsung/bells.c | 40 +- sound/soc/samsung/littlemill.c | 32 +- sound/soc/samsung/odroidx2_max98090.c | 9 +- sound/soc/samsung/snow.c | 9 +- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tobermory.c | 21 +- sound/soc/soc-core.c | 642 +++++++++++++++++++-------- sound/soc/soc-dapm.c | 7 +- sound/soc/soc-pcm.c | 22 +- sound/soc/soc-topology.c | 99 ++++- sound/soc/tegra/tegra_wm8903.c | 3 +- 22 files changed, 701 insertions(+), 291 deletions(-) -- 1.9.1

3 2

[alsa-devel] writing an alsa driver
by Johnny 10 Aug '15

10 Aug '15

Hi, I have been following your progress with great anticipation. > SSP2 > http://permalink.gmane.org/gmane.linux.alsa.devel/140328 > future versions of the firmware should support SSP1 So, I'm guessing, until the firmware supports SSP1, there will not be any sound? I also tried using the firmware that supports SSP0, which Vinod posted in another thread, and get no sound. Regards, John

1 0

[alsa-devel] [PATCH v3] ASoC: wm8960: update pll and clock setting function
by Zidan Wang 10 Aug '15

10 Aug '15

Add sysclk auto mode. When it's sysclk auto mode, if the MCLK is available for clock configure, using MCLK to provide sysclk directly, otherwise, search a available pll out frequcncy and set pll. Configure clock in hw_params may cause problems when using bypass style paths without hw_params in machine driver getting called. So add configure clock to set_bias_level. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 217 +++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8960.h | 1 + 2 files changed, 177 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 94c5c46..e9f8e41 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -48,6 +48,9 @@ #define WM8960_DISOP 0x40 #define WM8960_DRES_MASK 0x30 +static bool is_pll_freq_available(unsigned int source, unsigned int target); +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out); /* * wm8960 register cache * We can't read the WM8960 register space when we are @@ -126,9 +129,12 @@ struct wm8960_priv { struct snd_soc_dapm_widget *rout1; struct snd_soc_dapm_widget *out3; bool deemph; - int playback_fs; + int lrclk; int bclk; int sysclk; + int clk_id; + int freq_in; + bool is_stream_in_use[2]; struct wm8960_data pdata; }; @@ -164,8 +170,8 @@ static int wm8960_set_deemph(struct snd_soc_codec *codec) if (wm8960->deemph) { best = 1; for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - wm8960->playback_fs) < - abs(deemph_settings[best] - wm8960->playback_fs)) + if (abs(deemph_settings[i] - wm8960->lrclk) < + abs(deemph_settings[best] - wm8960->lrclk)) best = i; } @@ -565,6 +571,9 @@ static struct { { 8000, 5 }, }; +/* -1 for reserved value */ +static const int sysclk_divs[] = { 1, -1, 2, -1 }; + /* Multiply 256 for internal 256 div */ static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 }; @@ -574,61 +583,109 @@ static const int bclk_divs[] = { 120, 160, 220, 240, 320, 320, 320 }; -static void wm8960_configure_clocking(struct snd_soc_codec *codec, - bool tx, int lrclk) +static int wm8960_configure_clocking(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int sysclk, bclk, lrclk, freq_out, freq_in; u16 iface1 = snd_soc_read(codec, WM8960_IFACE1); - u16 iface2 = snd_soc_read(codec, WM8960_IFACE2); - u32 sysclk; - int i, j; + int i, j, k; if (!(iface1 & (1<<6))) { dev_dbg(codec->dev, "Codec is slave mode, no need to configure clock\n"); - return; + return 0; + } + + if (wm8960->clk_id != WM8960_SYSCLK_MCLK && !wm8960->freq_in) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; } - if (!wm8960->sysclk) { - dev_dbg(codec->dev, "No SYSCLK configured\n"); - return; + freq_in = wm8960->freq_in; + bclk = wm8960->bclk; + lrclk = wm8960->lrclk; + /* + * If it's sysclk auto mode, check if the MCLK can provide sysclk or + * not. If MCLK can provide sysclk, using MCLK to provide sysclk + * directly. Otherwise, auto select a available pll out frequency + * and set PLL. + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO) { + /* disable the PLL and using MCLK to provide sysclk */ + wm8960_set_pll(codec, 0, 0); + freq_out = freq_in; + } else if (wm8960->sysclk) { + freq_out = wm8960->sysclk; + } else { + dev_err(codec->dev, "No SYSCLK configured\n"); + return -EINVAL; } - if (!wm8960->bclk || !lrclk) { - dev_dbg(codec->dev, "No audio clocks configured\n"); - return; + /* check if the sysclk frequency is available. */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + sysclk = freq_out / sysclk_divs[i]; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + if (sysclk == dac_divs[j] * lrclk) { + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) + if (sysclk == bclk * bclk_divs[k] / 10) + break; + if (k != ARRAY_SIZE(bclk_divs)) + break; + } + } + if (j != ARRAY_SIZE(dac_divs)) + break; } - for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { - if (wm8960->sysclk == lrclk * dac_divs[i]) { - for (j = 0; j < ARRAY_SIZE(bclk_divs); ++j) { - sysclk = wm8960->bclk * bclk_divs[j] / 10; - if (wm8960->sysclk == sysclk) + if (i != ARRAY_SIZE(sysclk_divs)) { + goto configure_clock; + } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; + } + /* get a available pll out frequency and set pll */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + sysclk = lrclk * dac_divs[j]; + freq_out = sysclk * sysclk_divs[i]; + + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) { + if (sysclk == bclk * bclk_divs[k] / 10 && + is_pll_freq_available(freq_in, freq_out)) { + wm8960_set_pll(codec, freq_in, freq_out); break; + } else { + continue; + } } - if(j != ARRAY_SIZE(bclk_divs)) + if (k != ARRAY_SIZE(bclk_divs)) break; } + if (j != ARRAY_SIZE(dac_divs)) + break; } - if (i == ARRAY_SIZE(dac_divs)) { - dev_err(codec->dev, "Unsupported sysclk %d\n", wm8960->sysclk); - return; + if (i == ARRAY_SIZE(sysclk_divs)) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; } - /* - * configure frame clock. If ADCLRC configure as GPIO pin, DACLRC - * pin is used as a frame clock for ADCs and DACs. - */ - if (iface2 & (1<<6)) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (!tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, i << 6); +configure_clock: + /* configure sysclk clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 3 << 1, i << 1); + + /* configure frame clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, j << 3); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, j << 6); /* configure bit clock */ - snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, j); + snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, k); + + return 0; } static int wm8960_hw_params(struct snd_pcm_substream *substream, @@ -667,9 +724,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + wm8960->lrclk = params_rate(params); /* Update filters for the new rate */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - wm8960->playback_fs = params_rate(params); + if (tx) { wm8960_set_deemph(codec); } else { for (i = 0; i < ARRAY_SIZE(alc_rates); i++) @@ -682,7 +739,23 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); - wm8960_configure_clocking(codec, tx, params_rate(params)); + wm8960->is_stream_in_use[tx] = true; + + if (codec->dapm.bias_level == SND_SOC_BIAS_ON && + !wm8960->is_stream_in_use[!tx]) + return wm8960_configure_clocking(codec); + + return 0; +} + +static int wm8960_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + wm8960->is_stream_in_use[tx] = false; return 0; } @@ -702,6 +775,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int ret; switch (level) { @@ -711,6 +785,9 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: switch (snd_soc_codec_get_bias_level(codec)) { case SND_SOC_BIAS_STANDBY: + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; if (!IS_ERR(wm8960->mclk)) { ret = clk_prepare_enable(wm8960->mclk); if (ret) { @@ -726,6 +803,13 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); break; @@ -780,6 +864,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int reg, ret; switch (level) { @@ -822,6 +907,10 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, msleep(100); + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; + if (!IS_ERR(wm8960->mclk)) { ret = clk_prepare_enable(wm8960->mclk); if (ret) { @@ -834,6 +923,13 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); @@ -892,6 +988,28 @@ struct _pll_div { u32 k:24; }; +static bool is_pll_freq_available(unsigned int source, unsigned int target) +{ + unsigned int Ndiv; + + if (source == 0 || target == 0) + return false; + + /* Scale up target to PLL operating frequency */ + target *= 4; + Ndiv = target / source; + + if (Ndiv < 6) { + source >>= 1; + Ndiv = target / source; + } + + if ((Ndiv < 6) || (Ndiv > 12)) + return false; + + return true; +} + /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) @@ -943,10 +1061,9 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out) { - struct snd_soc_codec *codec = codec_dai->codec; u16 reg; static struct _pll_div pll_div; int ret; @@ -986,6 +1103,20 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, return 0; } +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + wm8960->freq_in = freq_in; + + if (pll_id == WM8960_SYSCLK_AUTO) + return 0; + + return wm8960_set_pll(codec, freq_in, freq_out); +} + static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { @@ -1043,11 +1174,14 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, WM8960_SYSCLK_PLL); break; + case WM8960_SYSCLK_AUTO: + break; default: return -EINVAL; } wm8960->sysclk = freq; + wm8960->clk_id = clk_id; return 0; } @@ -1060,6 +1194,7 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, static const struct snd_soc_dai_ops wm8960_dai_ops = { .hw_params = wm8960_hw_params, + .hw_free = wm8960_hw_free, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, .set_clkdiv = wm8960_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h index 2d8163d..ab3220d 100644 --- a/sound/soc/codecs/wm8960.h +++ b/sound/soc/codecs/wm8960.h @@ -82,6 +82,7 @@ #define WM8960_SYSCLK_MCLK (0 << 0) #define WM8960_SYSCLK_PLL (1 << 0) +#define WM8960_SYSCLK_AUTO (2 << 0) #define WM8960_DAC_DIV_1 (0 << 3) #define WM8960_DAC_DIV_1_5 (1 << 3) -- 1.9.1

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[alsa-devel] Support for snd_pcm_link() on the rate plugin
by Christian Gruber 10 Aug '15

10 Aug '15

Dear ALSA community, I have a question regarding the usage of the function snd_pcm_link() on the rate plugin. As can be seen from its implementation (see pcm.c, line 1368), the function snd_pcm_link() calls the fast_op "link" of the first pcm device/plugin to be linked, which is a function pointer, whose destination depends on the type of pcm device/plugin to be linked. If the device is a HW-device, the function snd_pcm_hw_link() is called for instance (see pcm_hw.c). Several other plugins use the generic function snd_pcm_generic_link() (see pcm_generic.c, line 212), which forwards the function call to their slave devices/plugins, e.g. the plugins mmap_emul, linear, lfloat, mulaw, alaw, route, etc. The rate plugin instead does not support snd_pcm_link(), since the function pointer "link" is not assigned in the definition of snd_pcm_rate_fast_ops (see pcm_rate.c, line 1153). Why does the rate plugin not support snd_pcm_link()? Is their any reason to not use the generic function snd_pcm_generic_link() within the rate plugin? If yes, what's the reason for it? Thanks in advance for answering this question, Christian

1 0

[alsa-devel] [PATCH 1/2] ASoC: extend seq_notifier callback interface
by yang.a.fang＠intel.com 10 Aug '15

10 Aug '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> This is needed for some codec driver to know whether it is power up/down sequence call from dapm Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/soc-dapm.c | 5 ++--- 6 files changed, 7 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 37d95a8..1d017a3 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -607,7 +607,7 @@ struct snd_soc_dapm_context { /* Go to BIAS_OFF in suspend if the DAPM context is idle */ unsigned int suspend_bias_off:1; void (*seq_notifier)(struct snd_soc_dapm_context *, - enum snd_soc_dapm_type, int); + enum snd_soc_dapm_type, int, bool); struct device *dev; /* from parent - for debug */ struct snd_soc_component *component; /* parent component */ diff --git a/include/sound/soc.h b/include/sound/soc.h index add5097..0072e21 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -875,7 +875,7 @@ struct snd_soc_codec_driver { bool suspend_bias_off; void (*seq_notifier)(struct snd_soc_dapm_context *, - enum snd_soc_dapm_type, int); + enum snd_soc_dapm_type, int, bool); bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */ }; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 8120b0c..6f4e6c3 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -733,7 +733,7 @@ WM5100_MIXER_CONTROLS("LHPF4", WM5100_HPLP4MIX_INPUT_1_SOURCE), }; static void wm5100_seq_notifier(struct snd_soc_dapm_context *dapm, - enum snd_soc_dapm_type event, int subseq) + enum snd_soc_dapm_type event, int subseq, bool power_up) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b011253..12da3b1 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -280,7 +280,7 @@ static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, #define WM8903_DCS_MODE_START_STOP 2 static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, - enum snd_soc_dapm_type event, int subseq) + enum snd_soc_dapm_type event, int subseq, bool power_up) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 2a696db..f12a900 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -686,7 +686,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) } static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, - enum snd_soc_dapm_type event, int subseq) + enum snd_soc_dapm_type event, int subseq, bool power_up) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1992568..ebea2d3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1528,8 +1528,7 @@ static void dapm_seq_run(struct snd_soc_card *card, for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) if (sort[i] == cur_sort) cur_dapm->seq_notifier(cur_dapm, - i, - cur_subseq); + i, cur_subseq, power_up); } if (cur_dapm && w->dapm != cur_dapm) @@ -1591,7 +1590,7 @@ static void dapm_seq_run(struct snd_soc_card *card, for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) if (sort[i] == cur_sort) cur_dapm->seq_notifier(cur_dapm, - i, cur_subseq); + i, cur_subseq, power_up); } list_for_each_entry(d, &card->dapm_list, list) { -- 1.7.9.5

4 7