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

[alsa-devel] Freescale iMX6SL SSI (I2S master mode) Rising edge vs Falling edge
by Xuebing Wang 28 May '15

28 May '15

Nicolin and alsa-devel community: Source code in fsl_ssi.c shows below in function fsl_ssi_set_dai_fmt(): ----------- switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: ... ... /* Data on rising edge of bclk, frame low, 1clk before data */ strcr |= CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TSCKP | CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; break; ----------- According to iMX6SL reference manual, 'TSCKP = 1' means "Data clocked out on *falling* edge of bit clock." (for I2S master mode), rather than "Data on rising edge of bclk in the comments". This means this comment in the source code is *partially* WRONG, am I correct? However, RSCKP is = 1 (for receiving), it means "Data latched on rising edge of bit clock", which is correct. Also, I am not sure why there is inconsistency in iMX6SL reference manual (48.9 SSI Memory Map/Register Definition), that TSCKP = 1 (for transmitting) is FALLING_EDGE, but RSCKP = 1 (for receiving) is RISING_EDGE. Note: SSI (I2S master mode) works perfect on my hardware, I did not observe anything wrong. Thanks. -- Xuebing

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[alsa-devel] [RFC v2 0/13] dw_hdmi cleanups, audio preparation, helpers and ahb audio support
by Russell King - ARM Linux 28 May '15

28 May '15

I'm sending this series for comments, and to allow people to update their code bases (for those who are using my AHB audio driver on iMX6). This is really several series, and I'm not expecting this to be ready for the upcoming merge window. That said, if anyone wants to say that they're happy with some of the initial six patches, I'm happy to queue those for David, if he's willing to take them. This series applies on top of my previous series, with the three patches from the rockchip guys applied, iow: drm: rockchip/dw_hdmi-rockchip: improve for HDMI electrical test drm: bridge/dw_hdmi: separate VLEVCTRL settting into platform driver drm: bridge/dw_hdmi: fixed codec style drm: bridge/dw_hdmi: adjust n/cts setting order drm: bridge/dw_hdmi: protect n/cts setting with a mutex drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() The first four patches are a few cleanups to the dw_hdmi driver. The next two patches introduce new interfaces to the dw_hdmi driver to support AHB audio, including the errata found on iMX6 which requires N to be programmed to zero when the audio FIFO is not full. I'm expecting some discussion with these last two as we try to work out how to deal with the two variants of audio support for this part (AHB audio vs I2S audio). The following three add various helpers to the DRM and ALSA subsystems for audio support which is non-specific to the AHB audio driver. For DRM, this is a helper macro to obtain the pointer into the short audio descriptors. For ALSA, it's a set of helpers to restrict an audio PCM device's capabilities according to the ELD, and to generate the IEC958 channel status data. If these helpers are acceptable, we can convert a number of drivers to them. The ELD helper isn't quite finished, but should be sufficient for initial testing. The last two patches add support for the dw_hdmi audio. This is an ALSA driver rather than an ASoC driver as it's a pretty simple affair. I couldn't find a suitable location in the sound/ subtree to place it, so I've placed it next to the bridge - yes, we probably need to find it a better home in the sound/ subtree, but it currently makes use of dw_hdmi.h in the directory it lives. drivers/gpu/drm/bridge/Kconfig | 11 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 566 +++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.h | 14 + drivers/gpu/drm/bridge/dw_hdmi.c | 321 +++++++++------- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + drivers/gpu/drm/imx/dw_hdmi-imx.c | 5 + include/drm/bridge/dw_hdmi.h | 7 + include/drm/drm_edid.h | 19 + include/sound/pcm_drm_eld.h | 6 + include/sound/pcm_iec958.h | 9 + sound/core/Kconfig | 6 + sound/core/Makefile | 2 + sound/core/pcm_drm_eld.c | 92 +++++ sound/core/pcm_iec958.c | 95 +++++ 15 files changed, 1029 insertions(+), 128 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

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Re: [alsa-devel] [PATCH 1/3] ASoC: max98090: read micbias from device property
by Anish Kumar 28 May '15

28 May '15

> From: "Fang, Yang A" <yang.a.fang(a)intel.com> > > This patch reads max98090 micbias from acpi or dt > > Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> > --- > sound/soc/codecs/max98090.c | 9 ++++++++- > 1 file changed, 8 insertions(+), 1 deletion(-) > > diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c > index c230626..27df17e 100644 > --- a/sound/soc/codecs/max98090.c > +++ b/sound/soc/codecs/max98090.c > @@ -2419,6 +2419,8 @@ static int max98090_probe(struct snd_soc_codec *codec) > struct max98090_cdata *cdata; > enum max98090_type devtype; > int ret = 0; > + int err; > + unsigned int micbias; > > dev_dbg(codec->dev, "max98090_probe\n"); > > @@ -2503,8 +2505,13 @@ static int max98090_probe(struct snd_soc_codec *codec) > snd_soc_write(codec, M98090_REG_BIAS_CONTROL, > M98090_VCM_MODE_MASK); > > + err = device_property_read_u32(codec->dev, "max,micbias", &micbias); if (err || micbias < M98090_MBVSEL_2V2 || micbias > M98090_MBVSEL_2V8) micbias = M98090_MBVSEL_2V8; Just stick sanity as well.

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[alsa-devel] [PATCH] ALSA: hda - Disable power_save_node for IDT92HD71bxx
by Takashi Iwai 27 May '15

27 May '15

We've got a regression report that 4.1-rc causes noises on a Dell laptop. Similar like Realtek codec, this seems also triggered by the recent power_save_node feature. As this kind of issue is quite hard to debug without actual hardware, disable the power_save_node flag for this codec as a workaround. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98971 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0de0fd95144a..6833c74ed6ff 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4696,7 +4696,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_save_node = 1; + /* disabled power_save_node since it causes noises on a Dell machine */ + /* codec->power_save_node = 1; */ spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; -- 2.4.1

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[alsa-devel] ASoC: regmap_config.max_register value wrong?
by Sergej Sawazki 27 May '15

27 May '15

Dear all, The following snippet is from the wm8741 codec driver. Shouldn't the .max_register field be set to WM8741_REGISTER_COUNT-1 (10) instead of WM8741_MAX_REGISTER (0x20)? [...] static const struct regmap_config wm8741_regmap = { .reg_bits = 7, .val_bits = 9, .max_register = WM8741_MAX_REGISTER, [...] WM8741_MAX_REGISTER is the address of the last register. WM8741_REGISTER_COUNT is the number of registers. The regmap_config documentation says: "@max_register: Optional, specifies the maximum valid register index." So it should be the *register index* and not the register address. Do I understand it correctly? Many thanks in advance. Best regards, Sergej

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[alsa-devel] [PATCH] ALSA: firewire-lib: fix buffer-over-run when detecting packet discontinuity
by Takashi Sakamoto 27 May '15

27 May '15

When detecting packet discontinuity, handle_in_packet() returns minus value and this value is assigned to unsigned int variable, then the variable has huge value. As a result, the variable causes buffer-over-run in handle_out_packet(). This brings invalid page request and system hangup. This commit fixes the bug to add a new argument into handle_in_packet() and the number of handled data blocks is assignd to it. The function return value is just used to check error. I also considered to change the type of local variable to 'int' in in_stream_callback(). This idea is based on type-conversion in C standard, while it may cause future problems when adding more works. Thus, I dropped this idea. Fixes: 6fc6b9ce41c6('ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets') Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 2b3e8b1..7bb988f 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -688,10 +688,10 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, } static int handle_in_packet(struct amdtp_stream *s, - unsigned int payload_quadlets, __be32 *buffer) + unsigned int payload_quadlets, __be32 *buffer, + unsigned int *data_blocks) { u32 cip_header[2]; - unsigned int data_blocks; unsigned int data_block_quadlets, data_block_counter, dbc_interval; struct snd_pcm_substream *pcm = NULL; bool lost; @@ -709,7 +709,7 @@ static int handle_in_packet(struct amdtp_stream *s, dev_info_ratelimited(&s->unit->device, "Invalid CIP header for AMDTP: %08X:%08X\n", cip_header[0], cip_header[1]); - data_blocks = 0; + *data_blocks = 0; goto end; } @@ -717,7 +717,7 @@ static int handle_in_packet(struct amdtp_stream *s, if (payload_quadlets < 3 || ((cip_header[1] & CIP_FDF_MASK) == (AMDTP_FDF_NO_DATA << CIP_FDF_SHIFT))) { - data_blocks = 0; + *data_blocks = 0; } else { data_block_quadlets = (cip_header[0] & CIP_DBS_MASK) >> CIP_DBS_SHIFT; @@ -731,12 +731,12 @@ static int handle_in_packet(struct amdtp_stream *s, if (s->flags & CIP_WRONG_DBS) data_block_quadlets = s->data_block_quadlets; - data_blocks = (payload_quadlets - 2) / data_block_quadlets; + *data_blocks = (payload_quadlets - 2) / data_block_quadlets; } /* Check data block counter continuity */ data_block_counter = cip_header[0] & CIP_DBC_MASK; - if (data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && + if (*data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && s->data_block_counter != UINT_MAX) data_block_counter = s->data_block_counter; @@ -746,10 +746,10 @@ static int handle_in_packet(struct amdtp_stream *s, } else if (!(s->flags & CIP_DBC_IS_END_EVENT)) { lost = data_block_counter != s->data_block_counter; } else { - if ((data_blocks > 0) && (s->tx_dbc_interval > 0)) + if ((*data_blocks > 0) && (s->tx_dbc_interval > 0)) dbc_interval = s->tx_dbc_interval; else - dbc_interval = data_blocks; + dbc_interval = *data_blocks; lost = data_block_counter != ((s->data_block_counter + dbc_interval) & 0xff); @@ -762,30 +762,30 @@ static int handle_in_packet(struct amdtp_stream *s, return -EIO; } - if (data_blocks > 0) { + if (*data_blocks > 0) { buffer += 2; pcm = ACCESS_ONCE(s->pcm); if (pcm) - s->transfer_samples(s, pcm, buffer, data_blocks); + s->transfer_samples(s, pcm, buffer, *data_blocks); if (s->midi_ports) - read_midi_messages(s, buffer, data_blocks); + read_midi_messages(s, buffer, *data_blocks); } if (s->flags & CIP_DBC_IS_END_EVENT) s->data_block_counter = data_block_counter; else s->data_block_counter = - (data_block_counter + data_blocks) & 0xff; + (data_block_counter + *data_blocks) & 0xff; end: if (queue_in_packet(s) < 0) return -EIO; if (pcm) - update_pcm_pointers(s, pcm, data_blocks); + update_pcm_pointers(s, pcm, *data_blocks); - return data_blocks; + return 0; } static void out_stream_callback(struct fw_iso_context *context, u32 cycle, @@ -853,8 +853,8 @@ static void in_stream_callback(struct fw_iso_context *context, u32 cycle, break; } - data_blocks = handle_in_packet(s, payload_quadlets, buffer); - if (data_blocks < 0) { + if (handle_in_packet(s, payload_quadlets, buffer, + &data_blocks) < 0) { s->packet_index = -1; break; } -- 2.1.4

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Re: [alsa-devel] [PATCH]: No sound when using software based mmap support and dmix plugin
by Takashi Iwai 27 May '15

27 May '15

At Wed, 13 May 2015 17:12:33 -0700, Yogesh Soni wrote: > > Hi all, > > I am trying to have dmix plugin work ( I succeeded) with the audio card > that does not have hardware mmap support and uses pcm_indirect method to > map software buffer for mmap feature. > > Most of the plugins were working quite well with mmap/pcm_indirect combo. > However, dmix plugin did not work (no sound). I debugged and realized that > appl_ptr in pcm_direct.h(snd_pcm_indirect_playback_transfer()) was always 0 > and hence nothing got copied to slave pcm buffer. I then discovered that > dmix operates in freewheeling-mode.. > > I have made a small workaround patch (I believe it can be cleaned up) in > case of free-wheeling mode, basically using sw_ready to hold bytes that > need to be copied. I used rec->sw_ready because it looked safest to use for > this purpose. As a result of this patch, normally period_size bytes are > copied to pcm buffer every period and sound is playing smooth. > > I am a noob in alsa and linux in general. So, I would like your opinion on > the patch. I would much prefer if I could write a patch that is generic and > clean enough, so that it could be used by others. patch is attached. Thanks for reporting and the patch. So, what actually prevents working with the current code in the free wheeling mode? Takashi

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[alsa-devel] [PATCH] ALSA: hda - Fix noise on AMD radeon 290x controller
by Takashi Iwai 27 May '15

27 May '15

A new AMD controller [1002:aac8] seems to need the quirk for other AMD NS HDMI stuff, otherwise it gives noisy sounds. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=99021 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 34040d26c94f..fea198c58196 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2089,6 +2089,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaab0), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaac8), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, -- 2.4.1

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[alsa-devel] [PATCH RESEND v3 0/2] reducing memory footprint of ALSA
by Jie Yang 27 May '15

27 May '15

We are planning to work on reducing memory footprint of ALSA, targeting on small memory embedded devices(e.g. IoT), this series focuses on disabling procfs, hw/sw params refinement, mmap, dpcm, dapm, compressed API, compat code APIs... We plan to enable those reducing items only for EXPERT, and add knobs in sound/core/Kconfig. These first two patches is a start, adding SND_PROC_FS to config disable/enable ALSA proc FS, this disabling can save about 9KB code size on x86_64 platform. Changes in v3: 1. drop SND_REDUCED_MEMORY_FOOTPRINT and make configuration available only for EXPERT; 2. move the configure into sound/core/Kconfig; 3. Add missing Signed-off-by for PATCH 2/2. Changes in v2: 1. Prompt "Sound Proc FS Support" appears only when SND_REDUCED_MEMORY_FOOTPRINT is true. Jie Yang (2): ALSA: Kconfig: add config item SND_PROC_FS for expert ALSA: replace CONFIG_PROC_FS with CONFIG_SND_PROC_FS include/sound/info.h | 8 ++++---- sound/core/Kconfig | 11 ++++++++++- sound/core/Makefile | 2 +- sound/core/hwdep.c | 6 +++--- sound/core/init.c | 8 ++++---- sound/core/oss/mixer_oss.c | 6 +++--- sound/core/pcm.c | 6 +++--- sound/core/seq/oss/seq_oss.c | 6 +++--- sound/core/seq/oss/seq_oss_init.c | 5 ++--- sound/core/seq/oss/seq_oss_midi.c | 4 ++-- sound/core/seq/oss/seq_oss_readq.c | 4 ++-- sound/core/seq/oss/seq_oss_synth.c | 4 ++-- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_device.c | 2 +- sound/core/seq/seq_info.h | 2 +- sound/core/seq/seq_queue.c | 4 ++-- sound/core/seq/seq_timer.c | 4 ++-- sound/core/sound.c | 4 ++-- sound/core/sound_oss.c | 4 ++-- sound/core/timer.c | 4 ++-- sound/pci/hda/Makefile | 3 ++- sound/pci/hda/hda_eld.c | 4 ++-- sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- 25 files changed, 62 insertions(+), 53 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: codecs: use SNDRV_PCM_FMTBIT_* for format bitmask
by Maciej S. Szmigiero 27 May '15

27 May '15

snd_soc_pcm_stream.formats is a bitmask of SNDRV_PCM_FMTBIT_*, not of SNDRV_PCM_FORMAT_* (which are sequential integers), however some of ASoC CODEC drivers use these values instead. Found out by sparse on 0-day kernel tester. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index ee31fa7..38b3dad 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1186,16 +1186,16 @@ static struct snd_soc_dai_driver pm860x_dai[] = { .channels_min = 2, .channels_max = 2, .rates = PM860X_RATES, - .formats = SNDRV_PCM_FORMAT_S16_LE | \ - SNDRV_PCM_FORMAT_S18_3LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE, }, .capture = { .stream_name = "PCM Capture", .channels_min = 2, .channels_max = 2, .rates = PM860X_RATES, - .formats = SNDRV_PCM_FORMAT_S16_LE | \ - SNDRV_PCM_FORMAT_S18_3LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE, }, .ops = &pm860x_pcm_dai_ops, }, { @@ -1207,16 +1207,16 @@ static struct snd_soc_dai_driver pm860x_dai[] = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FORMAT_S16_LE | \ - SNDRV_PCM_FORMAT_S18_3LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE, }, .capture = { .stream_name = "I2S Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FORMAT_S16_LE | \ - SNDRV_PCM_FORMAT_S18_3LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE, }, .ops = &pm860x_i2s_dai_ops, }, diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 2341e8e..ed4cca7 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -320,7 +320,7 @@ static struct snd_soc_dai_driver stac9766_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, }, /* alsa ops */ .ops = &stac9766_dai_ops_digital, diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index e7d2ecd..493faf5 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -998,8 +998,8 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute) SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) #define WM8900_PCM_FORMATS \ - (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ - SNDRV_PCM_FORMAT_S24_LE) + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops wm8900_dai_ops = { .hw_params = wm8900_hw_params, diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 9d18a0e..89cd2d6 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1054,8 +1054,8 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_48000) #define WM9713_PCM_FORMATS \ - (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ - SNDRV_PCM_FORMAT_S24_LE) + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops wm9713_dai_ops_hifi = { .prepare = ac97_hifi_prepare,

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