May 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH -next] ASoC: imx-card: Make some symbols static
by Wei Yongjun 24 May '21

24 May '21

The sparse tool complains as follows: sound/soc/fsl/imx-card.c:121:27: warning: symbol 'ak4458_fs_mul' was not declared. Should it be static? sound/soc/fsl/imx-card.c:138:31: warning: symbol 'ak4458_tdm_fs_mul' was not declared. Should it be static? sound/soc/fsl/imx-card.c:149:27: warning: symbol 'ak4497_fs_mul' was not declared. Should it be static? sound/soc/fsl/imx-card.c:166:27: warning: symbol 'ak5558_fs_mul' was not declared. Should it be static? sound/soc/fsl/imx-card.c:180:31: warning: symbol 'ak5558_tdm_fs_mul' was not declared. Should it be static? Those symbols are not used outside of imx-card.c, so marks them static. Fixes: aa736700f42f ("ASoC: imx-card: Add imx-card machine driver") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/fsl/imx-card.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index ab424735bbfe..58fd0639a069 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -118,7 +118,7 @@ struct imx_card_data { u32 asrc_format; }; -struct imx_akcodec_fs_mul ak4458_fs_mul[] = { +static struct imx_akcodec_fs_mul ak4458_fs_mul[] = { /* Normal, < 32kHz */ { .rmin = 8000, .rmax = 24000, .wmin = 1024, .wmax = 1024, }, /* Normal, 32kHz */ @@ -135,7 +135,7 @@ struct imx_akcodec_fs_mul ak4458_fs_mul[] = { { .rmin = 705600, .rmax = 768000, .wmin = 16, .wmax = 64, }, }; -struct imx_akcodec_tdm_fs_mul ak4458_tdm_fs_mul[] = { +static struct imx_akcodec_tdm_fs_mul ak4458_tdm_fs_mul[] = { /* * Table 13 - Audio Interface Format * For TDM mode, MCLK should is set to @@ -146,7 +146,7 @@ struct imx_akcodec_tdm_fs_mul ak4458_tdm_fs_mul[] = { { .min = 512, .max = 512, .mul = 1024 }, /* TDM512 */ }; -struct imx_akcodec_fs_mul ak4497_fs_mul[] = { +static struct imx_akcodec_fs_mul ak4497_fs_mul[] = { /** * Table 7 - mapping multiplier and speed mode * Tables 8 & 9 - mapping speed mode and LRCK fs @@ -163,7 +163,7 @@ struct imx_akcodec_fs_mul ak4497_fs_mul[] = { * Auto MCLK selection based on LRCK for Normal Mode * (Table 4 from datasheet) */ -struct imx_akcodec_fs_mul ak5558_fs_mul[] = { +static struct imx_akcodec_fs_mul ak5558_fs_mul[] = { { .rmin = 8000, .rmax = 32000, .wmin = 1024, .wmax = 1024, }, { .rmin = 44100, .rmax = 48000, .wmin = 512, .wmax = 512, }, { .rmin = 88200, .rmax = 96000, .wmin = 256, .wmax = 256, }, @@ -177,7 +177,7 @@ struct imx_akcodec_fs_mul ak5558_fs_mul[] = { * because of SAI we also add the restriction: MCLK >= 2 * BCLK * (Table 9 from datasheet) */ -struct imx_akcodec_tdm_fs_mul ak5558_tdm_fs_mul[] = { +static struct imx_akcodec_tdm_fs_mul ak5558_tdm_fs_mul[] = { { .min = 128, .max = 128, .mul = 256 }, { .min = 256, .max = 256, .mul = 512 }, { .min = 512, .max = 512, .mul = 1024 },

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[PATCH v3 0/7] PinePhone BT audio bringup
by Samuel Holland 24 May '21

24 May '21

This series uses the additional DAIs added to the sun8i-codec driver to add hardware routing for BT SCO (headset) audio on the PinePhone. The BT audio connection is represented by the "dummy" bt-sco codec. The connection to the Quectel EG-25G modem via AIF2 works as well, but I do not include it here because there is no appropriate codec driver in tree. We have been using an out-of-tree "dummy" codec driver for the modem similar to bt-sco, and I'm not sure if such a driver would be desired upstream. Changes from v2: - Also accept #sound-dai-cells in the binding. Since dt-core.yaml already sets the type of this property, it is not possible to use oneOf, nor make a specific value deprecated. Changes from v1: - Fixed DT binding example to follow new binding Arnaud Ferraris (1): arm64: dts: allwinner: pinephone: Set audio card name Samuel Holland (6): ASoC: dt-bindings: sun8i-codec: Increase #sound-dai-cells ARM: dts: sun8i-a33: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Add pinmux nodes for AIF2/AIF3 arm64: dts: allwinner: a64: Allow multiple DAI links arm64: dts: allwinner: pinephone: Add support for Bluetooth audio .../sound/allwinner,sun8i-a33-codec.yaml | 8 +++- arch/arm/boot/dts/sun8i-a33.dtsi | 4 +- .../dts/allwinner/sun50i-a64-pinephone.dtsi | 25 +++++++++++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 37 ++++++++++++++----- 4 files changed, 60 insertions(+), 14 deletions(-) -- 2.26.3

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[PATCH] ASoC: core: Fix Null-point-dereference in fmt_single_name()
by Kefeng Wang 24 May '21

24 May '21

Check the return value of devm_kstrdup() in case of Null-point-dereference. Fixes: 45dd9943fce0 ("ASoC: core: remove artificial component and DAI name constraint") Cc: Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org> Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Kefeng Wang <wangkefeng.wang(a)huawei.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1c0904acb935..a76974ccfce1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2225,6 +2225,8 @@ static char *fmt_single_name(struct device *dev, int *id) return NULL; name = devm_kstrdup(dev, devname, GFP_KERNEL); + if (!name) + return NULL; /* are we a "%s.%d" name (platform and SPI components) */ found = strstr(name, dev->driver->name); -- 2.26.2

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[RFC PATCH 1/2] ASoC: dt-bindings: codecs: Add bindings for nxp, tfa989x
by Stephan Gerhold 24 May '21

24 May '21

NXP/Goodix TFA989X (TFA1) amplifiers are controlled via an I2C bus. Add simple device tree bindings that describe how to set them up in the device tree. Right now only nxp,tfa9895 is supported but this will be extended to at least nxp,tfa9897 in the near future. Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- .../bindings/sound/nxp,tfa989x.yaml | 54 +++++++++++++++++++ 1 file changed, 54 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml new file mode 100644 index 000000000000..45db5776550c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nxp,tfa989x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP/Goodix TFA989X (TFA1) Audio Amplifiers + +maintainers: + - Stephan Gerhold <stephan(a)gerhold.net> + +properties: + compatible: + enum: + - nxp,tfa9895 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + sound-name-prefix: + $ref: /schemas/types.yaml#/definitions/string + description: + Used as prefix for sink/source names of the component. Must be a + unique string among multiple instances of the same component. + +required: + - compatible + - reg + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@34 { + compatible = "nxp,tfa9895"; + reg = <0x34>; + sound-name-prefix = "Speaker Left"; + #sound-dai-cells = <0>; + }; + audio-codec@36 { + compatible = "nxp,tfa9895"; + reg = <0x36>; + sound-name-prefix = "Speaker Right"; + #sound-dai-cells = <0>; + }; + }; -- 2.31.1

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[PATCH -next] ASoC: tlv320aic26: use DEVICE_ATTR_RW macro
by YueHaibing 24 May '21

24 May '21

Use DEVICE_ATTR_RW() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/tlv320aic26.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index c7baef8948d4..077415a57225 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -261,8 +261,8 @@ static const struct snd_kcontrol_new aic26_snd_controls[] = { * SPI device portion of driver: sysfs files for debugging */ -static ssize_t aic26_keyclick_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t keyclick_show(struct device *dev, + struct device_attribute *attr, char *buf) { struct aic26 *aic26 = dev_get_drvdata(dev); int val, amp, freq, len; @@ -276,9 +276,9 @@ static ssize_t aic26_keyclick_show(struct device *dev, } /* Any write to the keyclick attribute will trigger the keyclick event */ -static ssize_t aic26_keyclick_set(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t keyclick_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) { struct aic26 *aic26 = dev_get_drvdata(dev); @@ -288,7 +288,7 @@ static ssize_t aic26_keyclick_set(struct device *dev, return count; } -static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); +static DEVICE_ATTR_RW(keyclick); /* --------------------------------------------------------------------- * SoC CODEC portion of driver: probe and release routines -- 2.17.1

1 0

usb: u_audio: Notifying gadget audio device about starting/stopping capture/playback on the host
by Pavel Hofman 24 May '21

24 May '21

Hi, I am trying to notify the user-space code running on USB audio gadget that the USB host has started/stopped capture/playback. For stopping action, I am enclosing a patch snippet. I used snd_pcm_stop code when alsa SPDIF receivers detect a change in the stream https://github.com/torvalds/linux/blob/master/sound/i2c/other/ak4117.c#L505 When testing the code, aplay/arecord ends, sox just reports alsa device warning and does not stop. IMO that is the desired outcome - the snd_pcm_read/write methods returned errors when capture/playback on the host side stopped. For starting the capture/playback on the host, perhaps some snd_ctl_notify on some dedicated alsa control could be used, notifying the user space application that it can open the gadget audio device? The patch below does not compile on upstream kernel as it uses changes not included in upstream yet (explicit feedback EP, samplerate switching), but I am just discussing principles and options now. Thank you very much for your opinion and recommendations. With regards, Pavel. diff --git a/drivers/usb/gadget/function/u_audio.c b/drivers/usb/gadget/function/u_audio.c index 45367d650c5a..c6cdb844fec1 100644 --- a/drivers/usb/gadget/function/u_audio.c +++ b/drivers/usb/gadget/function/u_audio.c @@ -565,6 +565,7 @@ int u_audio_start_capture(struct g_audio *audio_dev) struct uac_params *params = &audio_dev->params; int req_len, i; + dev_dbg(dev, "starting capture with rate %d\n", params->c_srate_active); ep = audio_dev->out_ep; prm = &uac->c_prm; config_ep_by_speed(gadget, &audio_dev->func, ep); @@ -635,6 +636,23 @@ EXPORT_SYMBOL_GPL(u_audio_start_capture); void u_audio_stop_capture(struct g_audio *audio_dev) { struct snd_uac_chip *uac = audio_dev->uac; + unsigned long _flags; + struct snd_pcm_substream *substream; + struct uac_rtd_params *prm; + + dev_dbg(uac->card->dev, "stopping capture\n"); + prm = &uac->c_prm; + if (prm) { + substream = prm->ss; + if (substream) { + dev_dbg(uac->card->dev, "stopping capture substream\n"); + snd_pcm_stream_lock_irqsave(substream, _flags); + if (snd_pcm_running(substream)) { + snd_pcm_stop(substream, SNDRV_PCM_STATE_DRAINING); + } + snd_pcm_stream_unlock_irqrestore(substream, _flags); + } + } if (audio_dev->in_ep_fback) free_ep_fback(&uac->c_prm, audio_dev->in_ep_fback); @@ -655,7 +673,7 @@ int u_audio_start_playback(struct g_audio *audio_dev) const struct usb_endpoint_descriptor *ep_desc; int req_len, i; - dev_dbg(dev, "start playback with rate %d\n", params->p_srate_active); + dev_dbg(dev, "starting playback with rate %d\n", params->p_srate_active); ep = audio_dev->in_ep; prm = &uac->p_prm; config_ep_by_speed(gadget, &audio_dev->func, ep); @@ -715,6 +733,23 @@ EXPORT_SYMBOL_GPL(u_audio_start_playback); void u_audio_stop_playback(struct g_audio *audio_dev) { struct snd_uac_chip *uac = audio_dev->uac; + unsigned long _flags; + struct snd_pcm_substream *substream; + struct uac_rtd_params *prm; + + dev_dbg(uac->card->dev, "stopping playback\n"); + prm = &uac->p_prm; + if (prm) { + substream = prm->ss; + if (substream) { + dev_dbg(uac->card->dev, "stopping playback substream\n"); + snd_pcm_stream_lock_irqsave(substream, _flags); + if (snd_pcm_running(substream)) { + snd_pcm_stop(substream, SNDRV_PCM_STATE_DRAINING); + } + snd_pcm_stream_unlock_irqrestore(substream, _flags); + } + } free_ep(&uac->p_prm, audio_dev->in_ep); }

1 1

Add UCMs for Google Asus Nexus 7 and Acer Iconia A500
by GitHub pull_request - reopened 24 May '21

24 May '21

alsa-project/alsa-ucm-conf pull request #92 was reopened from digetx: @perexg As was discussed previouly at https://www.spinics.net/lists/alsa-devel/msg126382.html I'm opening PR with the UCMs for Nexus 7 and A500. Both these devices are supported by the mainline Linux kernel. You're right that Tegra ASoC driver doesn't set card's name properly. In particular I found that we need to set the card's `driver_name` to `tegra`, then UCM get a proper lookup path. I rebased UCMs on top of the recent alsa-ucm-conf master and switched Nexus 7 UCM to use generic RT5640 UCM. Everything works great once the card name of the kernel driver is corrected. I'll send the kernel patches ASAP for review. Nexus 7 alsa-info: https://alsa-project.org/db/?f=0a29f1dab39ec5b33e01215dee42dbfd1dbbf08e A500 alsa-info: https://alsa-project.org/db/?f=bde4f7938a39b1d26904b414969fc811e726242f The alsa-info is taken using the updated kernel ASoC driver. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/92 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/92.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Srinivasa Rao Mandadapu 24 May '21

24 May '21

This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start and clock enable. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since V1: -- Enableed BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 48 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 46 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..c5bb3f16d25f 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,9 +93,18 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); - clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); + /* To ensure BCLK/LRCLK disabled even in device node validation. + Will not impact if disabled in trigger suspend */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + clk_disable_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -275,6 +284,8 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* To ensure lpass BCLK/LRCLK is enabled during + device resume. Will not impact if enabled in prepare */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +307,8 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* To ensure lpass BCLK/LRCLK is disabled during + device suspend */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -308,19 +321,50 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, ret); clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); - + break; + default: break; } return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret = -EINVAL; + /* To ensure lpass BCLK/LRCLK is enabled bit before + playback/capture data flow starts */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", + ret); + + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + return ret; + } + return 0; +} + + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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Dmix plugin does not update state when sound card is removed.
by GitHub issues - opened 23 May '21

23 May '21

alsa-project/alsa-lib issue #137 was opened from MHoyer12345678: Playing sound using snd_pcm_writei in a dmix setup might result in an infinite loop when sound card is removed. **Setup:** - dmix setup - pcm dmix device opened non blocking - snd_pcm_writei used with poll (POLLERR is not evaluated) **Trigger:** - usb sound card is removed (e.g. by "echo 0 > /sys/bus/usb/x-x/authorized") **Observation:** - snd_pcm_writei returns -EAGAIN on each call - snd_pcm_state report state RUNNING **Expected:** - snd_pcm_writei returns -ENODEV (as happening in case a pure hw configuration is used without dmix plugin) - snd_pcm_state report state DISCONNECTED (as happening in case a pure hw configuration is used without dmix plugin) **Root Cause:** - snd_pcm_writei calls snd_pcm_write_areas - snd_pcm_write_areas requests state of dmix pcm device (state = __snd_pcm_state(pcm);) - dmix implementation of snd_pcm_state requests state from slave hw device (state = snd_pcm_state(dmix->spcm);) - hw implementation of snd_pcm_state executes an ioctl (query_status -> sync_ptr1) - the ioctl call fails with -ENODEV (after the device has been removed) - pcm_hw returns -ENODEV as state - dmix implementation of snd_pcm_state ends in default section of the switch block - this leads to the unchanged dmix->state returned - this way, the -ENODEV from the slave device is finally ignored in dmix plugin **Potential Fix:** - Add the following three lines to snd_pcm_hw_state in pcm_hw.c after query_status_data ``` //disconnected HW is reported via a failed IOCTL return -ENODEV if (err == -ENODEV) return SND_PCM_STATE_DISCONNECTED; ``` - The fix takes -ENODEV as trigger to return state DISCONNECTED Hope this helps ... Issue URL : https://github.com/alsa-project/alsa-lib/issues/137 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/3] ALSA: firewire-lib: drop initial NODATA packets or empty packets
by Takashi Sakamoto 23 May '21

23 May '21

Hi, This patchset is for convenience of ALSA bebob and firewire-tascam drivers. The devices based on BeBoB ASICs or the devices in Tascam FireWire series transfer a batch of NODATA packet or empty packet in initial step of streaming. To avoid processing them, current implementation uses an option to skip processing content of tx packet during some initial cycles. However, the hard-coded number is not enough useful. In 1st patch, ALSA IEC 61883-1/6 packet streaming engine becomes to drop the initial packets. As a result, The tx_init_skip_cycles argument of amdtp_domain_start() function changes its meaning. In the following patches, ALSA bebob driver is refactored. Takashi Sakamoto (3): ALSA: firewire-lib: drop initial tx packets with NODATA or empty ALSA: bebob: cancel switching connection order ALSA: bebob: distinguish M-Audio ProFire Lightbridge quirk sound/firewire/amdtp-stream.c | 130 +++++++++++++++++++++------- sound/firewire/amdtp-stream.h | 3 + sound/firewire/bebob/bebob.c | 10 ++- sound/firewire/bebob/bebob.h | 1 + sound/firewire/bebob/bebob_stream.c | 25 ++---- 5 files changed, 119 insertions(+), 50 deletions(-) -- 2.27.0

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