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

Re: [alsa-devel] ALC256 alias name for Dell
by Takashi Iwai 26 May '15

26 May '15

At Mon, 25 May 2015 09:25:07 +0000, Kailang wrote: > > Hi Takashi, > > Dell has new platform for ALC256 codec. > It calls ALC3246 for codec. Applied, thanks. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-alc256-alias-name-for-dell.patch <application/octet-stream (base64)>] >

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Re: [alsa-devel] Bug 95261 - [SLK][Audio][HDA Analog] Audio capture is noisy if device is in RTD3 mode
by Takashi Iwai 26 May '15

26 May '15

At Mon, 25 May 2015 06:57:09 +0000, Pshou wrote: > > Hi Takashi, > This is when you shut up, occurred about five seconds record noise problems. > The problem, likely to occur in alc286 and sky board. I suppose the COEFs are specific to ALC286? If so, please either create alc286_shutup() or call COEF setups conditionally only for ALC286. thanks, Takashi > > Best regards > pshou > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 patch_realtek.diff <application/octet-stream (base64)>] >

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[alsa-devel] [PATCH v2 0/3] ASoC: topology
by Liam Girdwood 25 May '15

25 May '15

ASoC topology (formerly called dynamic firmware) allows audio DSP topology to be defined as part of the firmware package and not hard coded into the driver. This allows DSP vendors to ship multiple different firmwares or firmwares with programmable topologies for different devices (i.e. phones, tablets, TVs) using a single DSP driver. The DSP topology can be contained as part of the firmware file or exist as separate files. The topology core and file format is generic text and not tied to any single vendor or device. The topology core consists of kernel and userspace components :- Kernel. The kernel core reads in the topology data and builds the driver capabilities and audio topology based on this data. This includes building the DAPM graph, setting PCM and DAI capabilities, registering kcontrols and registering DAPM widgets. The userspace tool allows the creation of binary DSP FW topology files based on a topology text files. The tool is temporarily available on github and its intended to be merged into the alsa-utils package after a clean up. https://github.com/lgirdwood/topology.git Changes since RFC :- o Fixed patch series to survive bisection builds. o Removed widget ID enum from UAPI. o Added channel mappings to controls. o Added TLV macro patch to UAPI (should this be in asound.h ?). o Fixed locking for runtime and init time topology loading. o Moved the UAPI group index field to the block header. Changes since V1 (back in 2012) :- o Name changed to ASoC Topology o Structures from soc.h and soc-dapm.h no longer exposed as UAPIs o Moved to component model from codec/platform model. o Added reserved fields in each UAPI structure. o UAPI structures packed and endianess defined for each structure. o Added support for PCM, DAI and codec <-> codec link capabilities. o Added support for client drivers manifests. o Added support for private data for each dynamic object type. o Single list of dynamic objects rather than list of each object type. Liam

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[alsa-devel] [PATCH 0/5 v2] ALSA: firewire-lib: purge restriction of synchronization for non-blocking mode
by Takashi Sakamoto 25 May '15

25 May '15

This patchset renews my previous one. [alsa-devel] [PATCH 0/4] firewire-lib: purge restriction of synchronization for non-blocking mode http://mailman.alsa-project.org/pipermail/alsa-devel/2015-May/092086.html Changes: * Use stack to keep calculation result per callback of isochronous receive context. * Fix my stupid mistake to compare byte count to quadlet count. (oh...) * Rename flag to proper name. * Add 4th patch newly for better reading. Takashi Sakamoto (5): ALSA: firewire-lib: add buffer-over-run protection at receiving more data blocks than expected ALSA: firewire-lib: simplify function to calculate the number of data blocks ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets ALSA: firewire-lib: set streaming error outside of packetization ALSA: firewire-lib: remove restriction for non-blocking mode sound/firewire/amdtp.c | 151 ++++++++++++++++++++++++-------------- sound/firewire/amdtp.h | 4 + sound/firewire/oxfw/oxfw-stream.c | 10 ++- 3 files changed, 106 insertions(+), 59 deletions(-) -- 2.1.4

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[alsa-devel] [RESEND PATCH v1 0/2] Add dw_hdmi i2s audio driver support
by Yakir Yang 25 May '15

25 May '15

As Russell have send an series patches for dw_hdmi audio, it make us upload dw_hdmi audio patch more easier, and most important it works well on rockchip platform, so I make this patch that try to add another audio interface support in dw_hdmi driver. Here are the list that I have make in my series: - rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h - add dw_hdmi i2s audio driver Here are the series patches that I depends on: * drm: bridge/dw_hdmi: fixed codec style [Yakir Yang] * drm: bridge/dw_hdmi_rockchip: improve hdmi eye-diagram test * drm: bridge/dw_hdmi: improve hdmi single-end test * drm: imx/dw_hdmi: move phy comments [Russell King] * drm: bridge/dw_hdmi: clean up phy configuration * drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() * drm: bridge/dw_hdmi: protect n/cts setting with a mutex * drm: bridge/dw_hdmi: adjust n/cts setting order * drm: bridge/dw_hdmi: clean up hdmi_set_clk_regenerator() * drm: bridge/dw_hdmi: use drm_hdmi_avi_infoframe_from_display_mode() * drm: bridge/dw_hdmi: simplify hdmi_config_AVI() a little * drm: bridge/dw_hdmi: remove mhsyncpolarity/mvsyncpolarity/minterlaced * drm: bridge/dw_hdmi: introduce interface to setting sample rate * drm: bridge/dw_hdmi: introduce interfaces to enable and disable audio * drm/edid: add function to help find SADs * sound/core: add DRM ELD helper * sound/core: add IEC958 channel status helper * drm: bridge/dw_hdmi-ahb-audio: add audio driver * drm: bridge/dw_hdmi-ahb-audio: parse ELD from HDMI driver Best regards, Yakir Yang Yakir Yang (2): drm: bridge/dw_hdmi: rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h drm: bridge/dw_hdmi-i2s-audio: add audio driver drivers/gpu/drm/bridge/Kconfig | 9 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 2 +- .../{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} | 0 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c | 412 +++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi.c | 26 +- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + 7 files changed, 442 insertions(+), 11 deletions(-) rename drivers/gpu/drm/bridge/{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} (100%) create mode 100644 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c -- 2.1.2

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[alsa-devel] ASoC: How to mix multi input ?
by Kuninori Morimoto 25 May '15

25 May '15

Hi Mark, ALSA SoC I would like to ask you how to mix multi input in ASoC. Renesas sound device(s) can mix multi input to 1 output. +-+ sound A --> |M| sound B --> |I| --> [ codec ] -> speaker sound C --> |X| +-+ Here, sound A/B/C will be used randomly. Does ALSA SoC already has this kind of system ? My image about these are like this. sound A: aplay -D hw:0 ... sound B: aplay -D hw:1 ... sound C: aplay -D hw:2 ... ... codec will be started if one of A/B/C are started, and it will be stopped if all of A/B/C are stopped.

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[alsa-devel] [PATCH] ALSA: bebob: add Digidesign Mbox 2 Pro support
by Takashi Sakamoto 24 May '15

24 May '15

This device is based on DM1000E and BeBoB version 1 is installed. $ cat /proc/asound/cards 0 [Pro ]: BeBoB - Mbox 2 Pro DIGIDESIGN Mbox 2 Pro (id:1, rev:1), GUID 00a07e0100a90000 at fw1.0, S400 $ cat /proc/asound/Pro/firewire/firmware Manufacturer: bridgeCo Protocol Ver: 1 Build Ver: 0 GUID: 0x00A07E0100A90000 Model ID: 0x01 Model Rev: 1 Firmware Date: 20071031 Firmware Time: 034402 Firmware ID: 0xA9 Firmware Ver: 16777215 Base Addr: 0x20080000 Max Size: 1572864 Loader Date: 20051207 Loader Time: 205554 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 1 + sound/firewire/bebob/bebob.c | 3 +++ 2 files changed, 4 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index ecec547..2a0a9aa 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -114,6 +114,7 @@ config SND_BEBOB * M-Audio FireWire410/AudioPhile/Solo * M-Audio Ozonic/NRV10/ProfireLightBridge * M-Audio FireWire 1814/ProjectMix IO + * Digidesign Mbox 2 Pro To compile this driver as a module, choose M here: the module will be called snd-bebob. diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 611b7da..e71bd35 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -57,6 +57,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define VEN_FOCUSRITE 0x0000130e #define VEN_MAUDIO1 0x00000d6c #define VEN_MAUDIO2 0x000007f5 +#define VEN_DIGIDESIGN 0x00a07e #define MODEL_FOCUSRITE_SAFFIRE_BOTH 0x00000000 #define MODEL_MAUDIO_AUDIOPHILE_BOTH 0x00010060 @@ -433,6 +434,8 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* M-Audio ProjectMix */ SND_BEBOB_DEV_ENTRY(VEN_MAUDIO1, MODEL_MAUDIO_PROJECTMIX, &maudio_special_spec), + /* Digidesign Mbox 2 Pro */ + SND_BEBOB_DEV_ENTRY(VEN_DIGIDESIGN, 0x0000a9, &spec_normal), /* IDs are unknown but able to be supported */ /* Apogee, Mini-ME Firewire */ /* Apogee, Mini-DAC Firewire */ -- 2.1.4

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[alsa-devel] [PATCH 0/4] ALSA: firewire-lib: code cleanup
by Takashi Sakamoto 24 May '15

24 May '15

Current implementation includes several macros and functions with improper name, hard-coded numerical value and so on. This patchset is for code cleanup. Takashi Sakamoto (4): ALSA: firewire-lib: rename local functions for code cleanup ALSA: firewire-lib: macro arrangement for code cleanup ALSA: firewire-lib: use dev_err() when detecting incoming streaming error ALSA: firewire-lib: use protocol error when detecting wrong value in CIP header sound/firewire/amdtp.c | 104 ++++++++++++++++++++++++++----------------------- 1 file changed, 55 insertions(+), 49 deletions(-) -- 2.1.4

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 24 May '15

24 May '15

Yet another experiment. Orpheus operation manual says there are six modes concerning Sync Source. -------------------------------------- Sync Source Master/Slave Description -------------------------------------- Local Master System is sync'ed to Orpheus' local clock Wordclock Master System is sync'ed to external wordclock applied to Orpheus DI Master System is sync'ed to external S/PDIF applied to Orpheus ADAT Master System is sync'ed to external ADAT (ADAT input must be enabled) PC DAW Slave Orpheus is slaved to system clock from host PC CSP Master System is sync'ed to Orpheus' 1394 clock (not recommended) -------------------------------------- With each of the five modes mentioned above (ADAT input not available), I tried to play music on Windows 7 using Audacity. The results are: ------------------- Local OK Wordclock NG DI NG PC DAW OK CSP OK ------------------- So it seems that if there is no external clock applied, when one is expected, Orpheus stops playing rather than trying to continue operation by free-running its clock. *** Takashi Sakamoto wrote: > Not funny, but headache. It's a customization by hardware vendor, I > mentioned. (in this case, by PrismSound Ltd.) > BridgeCo AG (currently ArchWave AG) produced DM1000/DM1100/DM1500 > chipset with BeBoB firmware. PrismSound Ltd. utilized the chipset and > firmware for streaming functionality of your Orpheus. > BeBoB firmware gives common ways to control itself, while the hadware > vendor added their own cusomization against it. I don't know exactly the > reason because there're few information about your device. I do not think I understand what you said here due to the lack of technical background, but what do you suggest at this moment to a casual applications programmer who wish to use snd-bebob with Orpheus? ------------ Answer 1 ------------ It should work. I received many successful reports for my final call for testing last year. It must be a hardware problem with your PC, or Firewire Cable, or Orpheus, or all of them. ------------ Answer 2 ------------ It should work. Check your software environment. Freshly installed Ubuntu 14.10 or later with the kernel 3.16 or later may not work. Always download the latest sources from GitHub, etc. ------------ Answer 3 ------------ Forget about snd-bebob, and fallback to libffado+jackd. The combination is loosing many packets, but if you say you do not notice any glitches, you should be completely happy with it. *** Regards, Mike Ood

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[alsa-devel] [PATCH 1/3] ASoC: fsl_sai: add sai master mode support
by Zidan Wang 24 May '15

24 May '15

When sai works on master mode, set its bit clock and frame clock. SAI has 4 MCLK source, bus clock, MCLK1, MCLK2 and MCLK3. fsl_sai_set_bclk will select proper MCLK source, then calculate and set the bit clock divider. After fsl_sai_set_bclk, enable the selected mclk in hw_params(), and add hw_free() to disable the mclk. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 117 ++++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 9 +++- 2 files changed, 121 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ee2671b..1ccc10d1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1,7 +1,7 @@ /* * Freescale ALSA SoC Digital Audio Interface (SAI) driver. * - * 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 as published by the @@ -251,12 +251,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr4 |= FSL_SAI_CR4_FSD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFM: + sai->is_slave_mode = true; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFS: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + sai->is_slave_mode = true; break; default: return -EINVAL; @@ -288,6 +290,79 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) return ret; } +static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai); + unsigned long clk_rate; + u32 savediv = 0, ratio, savesub = freq; + u32 id; + int ret = 0; + + /* Don't apply to slave mode */ + if (sai->is_slave_mode) + return 0; + + for (id = 0; id < FSL_SAI_MCLK_MAX; id++) { + clk_rate = clk_get_rate(sai->mclk_clk[id]); + if (!clk_rate) + continue; + + ratio = clk_rate / freq; + + ret = clk_rate - ratio * freq; + + /* + * Drop the source that can not be + * divided into the required rate. + */ + if (ret != 0 && clk_rate / ret < 1000) + continue; + + dev_dbg(dai->dev, + "ratio %d for freq %dHz based on clock %ldHz\n", + ratio, freq, clk_rate); + + if (ratio % 2 == 0 && ratio >= 2 && ratio <= 512) + ratio /= 2; + else + continue; + + if (ret < savesub) { + savediv = ratio; + sai->mclk_id[tx] = id; + savesub = ret; + } + + if (ret == 0) + break; + } + + if (savediv == 0) { + dev_err(dai->dev, "failed to derive required %cx rate: %d\n", + tx ? 'T' : 'R', freq); + return -EINVAL; + } + + if ((tx && sai->synchronous[TX]) || (!tx && !sai->synchronous[RX])) { + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, + FSL_SAI_CR2_MSEL_MASK, + FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, + FSL_SAI_CR2_DIV_MASK, savediv - 1); + } else { + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, + FSL_SAI_CR2_MSEL_MASK, + FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, + FSL_SAI_CR2_DIV_MASK, savediv - 1); + } + + dev_dbg(dai->dev, "best fit: clock id=%d, div=%d, deviation =%d\n", + sai->mclk_id[tx], savediv, savesub); + + return 0; +} + static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -297,6 +372,24 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(params); u32 word_width = snd_pcm_format_width(params_format(params)); u32 val_cr4 = 0, val_cr5 = 0; + int ret; + + if (!sai->is_slave_mode) { + ret = fsl_sai_set_bclk(cpu_dai, tx, + 2 * word_width * params_rate(params)); + if (ret) + return ret; + + /* Do not enable the clock if it is already enabled */ + if (!(sai->mclk_streams & BIT(substream->stream))) { + ret = clk_prepare_enable(sai->mclk_clk[sai->mclk_id[tx]]); + if (ret) + return ret; + + sai->mclk_streams |= BIT(substream->stream); + } + + } if (!sai->is_dsp_mode) val_cr4 |= FSL_SAI_CR4_SYWD(word_width); @@ -322,6 +415,22 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, return 0; } +static int fsl_sai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + if (!sai->is_slave_mode && + sai->mclk_streams & BIT(substream->stream)) { + clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]); + sai->mclk_streams &= ~BIT(substream->stream); + } + + return 0; +} + + static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { @@ -428,6 +537,7 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .set_sysclk = fsl_sai_set_dai_sysclk, .set_fmt = fsl_sai_set_dai_fmt, .hw_params = fsl_sai_hw_params, + .hw_free = fsl_sai_hw_free, .trigger = fsl_sai_trigger, .startup = fsl_sai_startup, .shutdown = fsl_sai_shutdown, @@ -600,8 +710,9 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->bus_clk = NULL; } - for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { - sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[0] = sai->bus_clk; + for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i); sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 3466720..0662809 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -72,13 +72,15 @@ /* SAI Transmit and Recieve Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) -#define FSL_SAI_CR2_MSEL_MASK (0xff << 26) +#define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 #define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) #define FSL_SAI_CR2_MSEL_MCLK2 BIT(27) #define FSL_SAI_CR2_MSEL_MCLK3 (BIT(26) | BIT(27)) +#define FSL_SAI_CR2_MSEL(ID) ((ID) << 26) #define FSL_SAI_CR2_BCP BIT(25) #define FSL_SAI_CR2_BCD_MSTR BIT(24) +#define FSL_SAI_CR2_DIV_MASK 0xff /* SAI Transmit and Recieve Configuration 3 Register */ #define FSL_SAI_CR3_TRCE BIT(16) @@ -120,7 +122,7 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 -#define FSL_SAI_MCLK_MAX 3 +#define FSL_SAI_MCLK_MAX 4 /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 @@ -132,11 +134,14 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; + bool is_slave_mode; bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; bool synchronous[2]; + unsigned int mclk_id[2]; + unsigned int mclk_streams; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; -- 1.9.1

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