March 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: rt5682: Check JD status when system resume" to the asoc tree
by Mark Brown 18 Mar '19

18 Mar '19

The patch ASoC: rt5682: Check JD status when system resume has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4834d7070c85a5fb69637265dbbb05d13043280c Mon Sep 17 00:00:00 2001 From: Shuming Fan <shumingf(a)realtek.com> Date: Fri, 8 Mar 2019 11:36:08 +0800 Subject: [PATCH] ASoC: rt5682: Check JD status when system resume The IRQ function may not work when system suspend. We remove snd_soc_dapm_force_enable_pin function call to make sure the bias off when idle and run into suspend/resume function. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5682.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 9d5acd2d04ab..b7e2e9937e6d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -910,13 +910,20 @@ static int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - struct snd_soc_dapm_context *dapm = - snd_soc_component_get_dapm(component); unsigned int val, count; if (jack_insert) { - snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); - snd_soc_dapm_sync(dapm); + + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2, RT5682_PWR_VREF2); + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); + usleep_range(15000, 20000); + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ, RT5682_PWR_CBJ); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH); @@ -944,8 +951,10 @@ static int rt5682_headset_detect(struct snd_soc_component *component, rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - snd_soc_dapm_disable_pin(dapm, "CBJ Power"); - snd_soc_dapm_sync(dapm); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2, 0); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ, 0); rt5682->jack_type = 0; } @@ -1591,8 +1600,6 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, - rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, @@ -1627,9 +1634,6 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("CBJ Power", RT5682_PWR_ANLG_3, - RT5682_PWR_CBJ_BIT, 0, NULL, 0), - /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682_rec1_l_mix, ARRAY_SIZE(rt5682_rec1_l_mix)), @@ -1792,17 +1796,13 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { /*Vref*/ {"MICBIAS1", NULL, "Vref1"}, - {"MICBIAS1", NULL, "Vref2"}, {"MICBIAS2", NULL, "Vref1"}, - {"MICBIAS2", NULL, "Vref2"}, {"CLKDET SYS", NULL, "CLKDET"}, {"IN1P", NULL, "LDO2"}, {"BST1 CBJ", NULL, "IN1P"}, - {"BST1 CBJ", NULL, "CBJ Power"}, - {"CBJ Power", NULL, "Vref2"}, {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, {"RECMIX1L", NULL, "RECMIX1L Power"}, @@ -1912,9 +1912,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"HP Amp", NULL, "Capless"}, {"HP Amp", NULL, "Charge Pump"}, {"HP Amp", NULL, "CLKDET SYS"}, - {"HP Amp", NULL, "CBJ Power"}, {"HP Amp", NULL, "Vref1"}, - {"HP Amp", NULL, "Vref2"}, {"HPOL Playback", "Switch", "HP Amp"}, {"HPOR Playback", "Switch", "HP Amp"}, {"HPOL", NULL, "HPOL Playback"}, @@ -2363,6 +2361,8 @@ static int rt5682_resume(struct snd_soc_component *component) regcache_cache_only(rt5682->regmap, false); regcache_sync(rt5682->regmap); + rt5682_irq(0, rt5682); + return 0; } #else -- 2.20.1

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[alsa-devel] [PATCH 1/2] ASoC: rt5682: fix jack type detection issue
by shumingf＠realtek.com 18 Mar '19

18 Mar '19

From: Shuming Fan <shumingf(a)realtek.com> The jack type detection needs the main bias power of analog. The modification makes sure the main bias power on/off while jack plug/unplug. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index b7e2e9937e6d..b98974edf6ad 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -915,7 +915,8 @@ static int rt5682_headset_detect(struct snd_soc_component *component, if (jack_insert) { snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, - RT5682_PWR_VREF2, RT5682_PWR_VREF2); + RT5682_PWR_VREF2 | RT5682_PWR_MB, + RT5682_PWR_VREF2 | RT5682_PWR_MB); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); usleep_range(15000, 20000); @@ -952,7 +953,7 @@ static int rt5682_headset_detect(struct snd_soc_component *component, snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, - RT5682_PWR_VREF2, 0); + RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, 0); @@ -2301,16 +2302,13 @@ static int rt5682_set_bias_level(struct snd_soc_component *component, switch (level) { case SND_SOC_BIAS_PREPARE: regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, - RT5682_PWR_MB | RT5682_PWR_BG, - RT5682_PWR_MB | RT5682_PWR_BG); + RT5682_PWR_BG, RT5682_PWR_BG); regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO); break; case SND_SOC_BIAS_STANDBY: - regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, - RT5682_PWR_MB, RT5682_PWR_MB); regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, RT5682_DIG_GATE_CTRL, RT5682_DIG_GATE_CTRL); break; @@ -2318,7 +2316,7 @@ static int rt5682_set_bias_level(struct snd_soc_component *component, regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, 0); regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, - RT5682_PWR_MB | RT5682_PWR_BG, 0); + RT5682_PWR_BG, 0); break; default: -- 2.21.0

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[alsa-devel] [PATCH 2/2] ASoC: rt5682: recording has no sound after booting
by shumingf＠realtek.com 18 Mar '19

18 Mar '19

From: Shuming Fan <shumingf(a)realtek.com> If ASRC turns on, HW will use clk_dac as the reference clock whether recording or playback. Both of clk_dac and clk_adc should set proper clock while using ASRC. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index b98974edf6ad..86a7fa31c294 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1208,7 +1208,7 @@ static int set_filter_clk(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - int ref, val, reg, sft, mask, idx = -EINVAL; + int ref, val, reg, idx = -EINVAL; static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48}; @@ -1222,15 +1222,10 @@ static int set_filter_clk(struct snd_soc_dapm_widget *w, idx = rt5682_div_sel(rt5682, ref, div_f, ARRAY_SIZE(div_f)); - if (w->shift == RT5682_PWR_ADC_S1F_BIT) { + if (w->shift == RT5682_PWR_ADC_S1F_BIT) reg = RT5682_PLL_TRACK_3; - sft = RT5682_ADC_OSR_SFT; - mask = RT5682_ADC_OSR_MASK; - } else { + else reg = RT5682_PLL_TRACK_2; - sft = RT5682_DAC_OSR_SFT; - mask = RT5682_DAC_OSR_MASK; - } snd_soc_component_update_bits(component, reg, RT5682_FILTER_CLK_DIV_MASK, idx << RT5682_FILTER_CLK_DIV_SFT); @@ -1242,7 +1237,8 @@ static int set_filter_clk(struct snd_soc_dapm_widget *w, } snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1, - mask, idx << sft); + RT5682_ADC_OSR_MASK | RT5682_DAC_OSR_MASK, + (idx << RT5682_ADC_OSR_SFT) | (idx << RT5682_DAC_OSR_SFT)); return 0; } -- 2.21.0

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[alsa-devel] [PATCH] ALSA: firewire-lib: use 8 byte header for IR context to get isochronous cycle
by Takashi Sakamoto 18 Mar '19

18 Mar '19

In kernel API of Linux FireWire subsystem, handlers of isochronous receive (IR) context can get context headers as an argument of callback. When 4 byte header is used, the context header includes isochronous packet header for each packet. When 8 byte header is used, it includes isochronous cycle as well. ALSA IEC 61883-1/6 engine uses 4 byte header, and computes isochronous cycle from the cycle of interrupt. The usage of 8 byte header can obsolete the computation. Furthermore, this change works well for a case that a series of packet in one interrupt includes skipped isochronous cycle, This commit uses 8 byte header to handle isochronous cycle. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 44 +++++++++++++++-------------------- 1 file changed, 19 insertions(+), 25 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 3ada55ed5381..43f28b813386 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -56,8 +56,9 @@ #define INTERRUPT_INTERVAL 16 #define QUEUE_LENGTH 48 -#define IN_PACKET_HEADER_SIZE 4 +#define IR_HEADER_SIZE 8 // For header and timestamp. #define OUT_PACKET_HEADER_SIZE 0 +#define HEADER_TSTAMP_MASK 0x0000ffff static void pcm_period_tasklet(unsigned long data); @@ -456,7 +457,7 @@ static inline int queue_out_packet(struct amdtp_stream *s, static inline int queue_in_packet(struct amdtp_stream *s) { - return queue_packet(s, IN_PACKET_HEADER_SIZE, s->max_payload_length); + return queue_packet(s, IR_HEADER_SIZE, s->max_payload_length); } static int handle_out_packet(struct amdtp_stream *s, @@ -701,13 +702,6 @@ static inline u32 increment_cycle_count(u32 cycle, unsigned int addend) return cycle; } -static inline u32 decrement_cycle_count(u32 cycle, unsigned int subtrahend) -{ - if (cycle < subtrahend) - cycle += 8 * CYCLES_PER_SECOND; - return cycle - subtrahend; -} - static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) @@ -745,29 +739,26 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, struct amdtp_stream *s = private_data; unsigned int i, packets; unsigned int payload_length, max_payload_length; - __be32 *headers = header; - u32 cycle; + __be32 *ctx_header = header; if (s->packet_index < 0) return; /* The number of packets in buffer */ - packets = header_length / IN_PACKET_HEADER_SIZE; - - cycle = compute_cycle_count(tstamp); - - /* Align to actual cycle count for the last packet. */ - cycle = decrement_cycle_count(cycle, packets); + packets = header_length / IR_HEADER_SIZE; /* For buffer-over-run prevention. */ max_payload_length = s->max_payload_length; for (i = 0; i < packets; i++) { - cycle = increment_cycle_count(cycle, 1); + u32 iso_header = be32_to_cpu(ctx_header[0]); + unsigned int cycle; + + tstamp = be32_to_cpu(ctx_header[1]) & HEADER_TSTAMP_MASK; + cycle = compute_cycle_count(tstamp); /* The number of bytes in this packet */ - payload_length = - (be32_to_cpu(headers[i]) >> ISO_DATA_LENGTH_SHIFT); + payload_length = iso_header >> ISO_DATA_LENGTH_SHIFT; if (payload_length > max_payload_length) { dev_err(&s->unit->device, "Detect jumbo payload: %04x %04x\n", @@ -777,6 +768,8 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, if (s->handle_packet(s, payload_length, cycle, i) < 0) break; + + ctx_header += IR_HEADER_SIZE / sizeof(__be32); } /* Queueing error or detecting invalid payload. */ @@ -797,6 +790,7 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, void *header, void *private_data) { struct amdtp_stream *s = private_data; + __be32 *ctx_header = header; u32 cycle; unsigned int packets; @@ -807,11 +801,10 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, s->callbacked = true; wake_up(&s->callback_wait); - cycle = compute_cycle_count(tstamp); - if (s->direction == AMDTP_IN_STREAM) { - packets = header_length / IN_PACKET_HEADER_SIZE; - cycle = decrement_cycle_count(cycle, packets); + tstamp = be32_to_cpu(ctx_header[1]) & HEADER_TSTAMP_MASK; + cycle = compute_cycle_count(tstamp); + context->callback.sc = in_stream_callback; if (s->flags & CIP_NO_HEADER) s->handle_packet = handle_in_packet_without_header; @@ -819,6 +812,7 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, s->handle_packet = handle_in_packet; } else { packets = header_length / 4; + cycle = compute_cycle_count(tstamp); cycle = increment_cycle_count(cycle, QUEUE_LENGTH - packets); context->callback.sc = out_stream_callback; if (s->flags & CIP_NO_HEADER) @@ -880,7 +874,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) if (s->direction == AMDTP_IN_STREAM) { dir = DMA_FROM_DEVICE; type = FW_ISO_CONTEXT_RECEIVE; - header_size = IN_PACKET_HEADER_SIZE; + header_size = IR_HEADER_SIZE; } else { dir = DMA_TO_DEVICE; type = FW_ISO_CONTEXT_TRANSMIT; -- 2.19.1

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[alsa-devel] [PATCH] ALSA: firewire-motu: add support MOTU 8pre FireWire
by Takashi Sakamoto 18 Mar '19

18 Mar '19

This commit adds support for MOTU 8pre FireWire, which was shipped 2007 and nowadays already discontinued. Userspace applications can transmit and receive PCM frames and MIDI messages for this model via ALSA PCM interface and RawMidi/Sequencer interfaces. Like the other models of MOTU FireWire series, this model has many quirks in its CIP. At first, data channels for two pairs of optical interfaces. At lower sampling transmission frequency, i.e. 44.1 and 48.0 kHz, one pair is available for ADAT data, thus 8 data chunks are transferred by CIP. At middle sampling transmission frequency, i.e. 88.2 and 96.0 kHz, two pairs are available to keep 8 chunks for ADAT data, thus CIP still includes 8 data chunks. Apart from data chunks for optical interface, CIP includes fixed number of data chunks. In tx stream, two chunks for status message, eight chunks for samples from analog 1-8 input, two chunks for mix-return. In rx stream, two chunks for control message, two chunks for main 1-2 output, two chunks for phone 1-2 output, two chunks for dummy 1-2. CIP header in tx stream includes quirks for its dbs and dbc fields. The value of dbs field is fixed to 0x13, against its actual size. The value of dbc field is firstly updated to 0x07 from zero, then it's incremented continuously according to actual number of data h blocks. Finally, the model has own bits to disable frame fetch. This commit uses several options to absorb the above quirks. $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 0410b57d bus_info_length 4, crc_length 16, crc 46461 404 31333934 bus_name "1394" 408 20001000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 0, max_rec 1 (4) 40c 0001f200 company_id 0001f2 | 410 00083dfb device_id 0000083dfb | EUI-64 0001f20000083dfb root directory ----------------------------------------------------------------- 414 0004c65c directory_length 4, crc 50780 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 8d000006 --> eui-64 leaf at 438 424 d1000001 --> unit directory at 428 unit directory at 428 ----------------------------------------------------------------- 428 0003991c directory_length 3, crc 39196 42c 120001f2 specifier id 430 1300000f version 434 17103800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 00022681 leaf_length 2, crc 9857 43c 0001f200 company_id 0001f2 | 440 00083dfb device_id 0000083dfb | EUI-64 0001f20000083dfb Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/amdtp-motu.c | 6 ++++ sound/firewire/motu/motu-protocol-v2.c | 43 +++++++++++++++++++------- sound/firewire/motu/motu.c | 15 +++++++++ sound/firewire/motu/motu.h | 1 + 4 files changed, 53 insertions(+), 12 deletions(-) diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index 6c9b743ea74b..cb0c967dea63 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -412,6 +412,12 @@ int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit, CIP_HEADER_WITHOUT_EOH; fmt = CIP_FMT_MOTU_TX_V3; } + + if (protocol == &snd_motu_protocol_v2) { + // 8pre has some quirks. + flags |= CIP_WRONG_DBS | + CIP_SKIP_DBC_ZERO_CHECK; + } } else { process_data_blocks = process_rx_data_blocks; flags |= CIP_DBC_IS_END_EVENT; diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index 453fc29fade7..848fffe7387e 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -15,6 +15,8 @@ #define V2_CLOCK_SRC_SHIFT 0 #define V2_CLOCK_TRAVELER_FETCH_DISABLE 0x04000000 #define V2_CLOCK_TRAVELER_FETCH_ENABLE 0x03000000 +#define V2_CLOCK_8PRE_FETCH_DISABLE 0x02000000 +#define V2_CLOCK_8PRE_FETCH_ENABLE 0x00000000 #define V2_IN_OUT_CONF_OFFSET 0x0c04 #define V2_OPT_OUT_IFACE_MASK 0x00000c00 @@ -132,20 +134,31 @@ static int v2_switch_fetching_mode(struct snd_motu *motu, bool enable) u32 data; int err = 0; - if (motu->spec == &snd_motu_spec_traveler) { + if (motu->spec == &snd_motu_spec_traveler || + motu->spec == &snd_motu_spec_8pre) { err = snd_motu_transaction_read(motu, V2_CLOCK_STATUS_OFFSET, &reg, sizeof(reg)); if (err < 0) return err; data = be32_to_cpu(reg); - data &= ~(V2_CLOCK_TRAVELER_FETCH_DISABLE | - V2_CLOCK_TRAVELER_FETCH_ENABLE); - - if (enable) - data |= V2_CLOCK_TRAVELER_FETCH_ENABLE; - else - data |= V2_CLOCK_TRAVELER_FETCH_DISABLE; + if (motu->spec == &snd_motu_spec_traveler) { + data &= ~(V2_CLOCK_TRAVELER_FETCH_DISABLE | + V2_CLOCK_TRAVELER_FETCH_ENABLE); + + if (enable) + data |= V2_CLOCK_TRAVELER_FETCH_ENABLE; + else + data |= V2_CLOCK_TRAVELER_FETCH_DISABLE; + } else if (motu->spec == &snd_motu_spec_8pre) { + data &= ~(V2_CLOCK_8PRE_FETCH_DISABLE | + V2_CLOCK_8PRE_FETCH_ENABLE); + + if (enable) + data |= V2_CLOCK_8PRE_FETCH_DISABLE; + else + data |= V2_CLOCK_8PRE_FETCH_ENABLE; + } reg = cpu_to_be32(data); err = snd_motu_transaction_write(motu, V2_CLOCK_STATUS_OFFSET, @@ -220,10 +233,16 @@ static void calculate_differed_part(struct snd_motu_packet_format *formats, * interfaces. */ data = (data & mask) >> shift; - if ((flags & SND_MOTU_SPEC_HAS_OPT_IFACE_A) && - data == V2_OPT_IFACE_MODE_ADAT) { - pcm_chunks[0] += 8; - pcm_chunks[1] += 4; + if (data == V2_OPT_IFACE_MODE_ADAT) { + if (flags & SND_MOTU_SPEC_HAS_OPT_IFACE_A) { + pcm_chunks[0] += 8; + pcm_chunks[1] += 4; + } + // 8pre has two sets of optical interface and doesn't reduce + // chunks for ADAT signals. + if (flags & SND_MOTU_SPEC_HAS_OPT_IFACE_B) { + pcm_chunks[1] += 4; + } } /* At mode x4, no data chunks are supported in this part. */ diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 513291ba0ab0..201539d4488c 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -203,6 +203,20 @@ const struct snd_motu_spec snd_motu_spec_traveler = { .analog_out_ports = 8, }; +const struct snd_motu_spec snd_motu_spec_8pre = { + .name = "8pre", + .protocol = &snd_motu_protocol_v2, + // In tx, use coax chunks for mix-return 1/2. In rx, use coax chunks for + // dummy 1/2. + .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | + SND_MOTU_SPEC_HAS_OPT_IFACE_A | + SND_MOTU_SPEC_HAS_OPT_IFACE_B | + SND_MOTU_SPEC_RX_MIDI_2ND_Q | + SND_MOTU_SPEC_TX_MIDI_2ND_Q, + .analog_in_ports = 8, + .analog_out_ports = 2, +}; + static const struct snd_motu_spec motu_828mk3 = { .name = "828mk3", .protocol = &snd_motu_protocol_v3, @@ -248,6 +262,7 @@ static const struct snd_motu_spec motu_audio_express = { static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x000003, &motu_828mk2), SND_MOTU_DEV_ENTRY(0x000009, &snd_motu_spec_traveler), + SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), SND_MOTU_DEV_ENTRY(0x000015, &motu_828mk3), /* FireWire only. */ SND_MOTU_DEV_ENTRY(0x000035, &motu_828mk3), /* Hybrid. */ SND_MOTU_DEV_ENTRY(0x000033, &motu_audio_express), diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index fd5327d30ab1..1cd112be7dad 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -130,6 +130,7 @@ extern const struct snd_motu_protocol snd_motu_protocol_v2; extern const struct snd_motu_protocol snd_motu_protocol_v3; extern const struct snd_motu_spec snd_motu_spec_traveler; +extern const struct snd_motu_spec snd_motu_spec_8pre; int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, -- 2.19.1

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[alsa-devel] [PATCH] ALSA: hda - call runtime_resume after S3 and S4 for each codec
by Hui Wang 18 Mar '19

18 Mar '19

Recently we found the audio jack detection doesn't work after suspend on many machines with Realtek codec. Sometimes the audio selection dialogue didn't show up after users plugged headhphone/headset into the headset jack, sometimes after uses plugged headphone/headset, then click the sound icon on the upper-right corner of gnome-desktop, it also showed the speaker rather than the headphone. The root cause is that before suspend, the codec already call the runtime_suspend since this codec is not used by any apps, then in resume, it will not call runtime_resume for this codec. But for some realtek codec (so far, alc236, alc255 and alc891) with the specific BIOS, if it doesn't run runtime_resume after suspend, all codec functions including jack detection stop working anymore. This problem existed for a long time, but it was not exposed, that is because if users is playing sound or recording sound, and at the same time they run suspend, the runtime_suspend will be called in pm_suspend and runtime_resume will be called in pm_resume; if audio codec is not used by any apps and users run suspend, the runtime_resume will not be called in pm_resume, then codec stops working, but if users play sound or open sound-setting to check audio device, this will trigger calling to runtime_resume ( via snd_hda_power_up), then the codec starts working again before users notice this problem. Since we don't know how many codec and BIOS combinations have this problem, to fix it, let the driver call runtime_resume for all codecs in pm_resume, maybe for some codecs, this is not needed, but it is harmless. After a codec is runtime resumed, if it is not used by any apps, it will be runtime suspended soon and furthermore we don't run suspend frequently, this change will not add much power consumption. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- include/sound/hda_codec.h | 3 +++ sound/pci/hda/hda_codec.c | 26 +++++++++++++++++++++++--- 2 files changed, 26 insertions(+), 3 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index cc7c8d42d4fd..a4e26d1d18bc 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -262,6 +262,9 @@ struct hda_codec { unsigned int auto_runtime_pm:1; /* enable automatic codec runtime pm */ unsigned int force_pin_prefix:1; /* Add location prefix */ unsigned int link_down_at_suspend:1; /* link down at runtime suspend */ +#ifdef CONFIG_PM_SLEEP + unsigned int already_rt_suspend:1; /* already runtime suspend before sleep */ +#endif #ifdef CONFIG_PM unsigned long power_on_acct; unsigned long power_off_acct; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5f2005098a60..7c2bbe25adde 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2939,34 +2939,54 @@ static int hda_codec_runtime_resume(struct device *dev) #endif /* CONFIG_PM */ #ifdef CONFIG_PM_SLEEP +static int hda_codec_force_resume(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + + if (codec->already_rt_suspend) { + int ret; + + pm_runtime_get_noresume(dev); + ret = pm_runtime_force_resume(dev); + pm_runtime_put(dev); + return ret; + } else + return pm_runtime_force_resume(dev); +} static int hda_codec_pm_suspend(struct device *dev) { + struct hda_codec *codec = dev_to_hda_codec(dev); + dev->power.power_state = PMSG_SUSPEND; + codec->already_rt_suspend = pm_runtime_suspended(dev); return pm_runtime_force_suspend(dev); } static int hda_codec_pm_resume(struct device *dev) { dev->power.power_state = PMSG_RESUME; - return pm_runtime_force_resume(dev); + return hda_codec_force_resume(dev); } static int hda_codec_pm_freeze(struct device *dev) { + struct hda_codec *codec = dev_to_hda_codec(dev); + dev->power.power_state = PMSG_FREEZE; + codec->already_rt_suspend = pm_runtime_suspended(dev); return pm_runtime_force_suspend(dev); } static int hda_codec_pm_thaw(struct device *dev) { dev->power.power_state = PMSG_THAW; - return pm_runtime_force_resume(dev); + return hda_codec_force_resume(dev); } static int hda_codec_pm_restore(struct device *dev) { dev->power.power_state = PMSG_RESTORE; - return pm_runtime_force_resume(dev); + return hda_codec_force_resume(dev); } #endif /* CONFIG_PM_SLEEP */ -- 2.17.1

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[alsa-devel] [PATCH] ALSA: opl3: fix mismatch between snd_opl3_drum_switch definition and declaration
by Colin King 18 Mar '19

18 Mar '19

From: Colin Ian King <colin.king(a)canonical.com> The function snd_opl3_drum_switch declaration in the header file has the order of the two arguments on_off and vel swapped when compared to the definition arguments of vel and on_off. Fix this by swapping them around to match the definition. This error predates the git history, so no idea when this error was introduced. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/drivers/opl3/opl3_voice.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/opl3/opl3_voice.h b/sound/drivers/opl3/opl3_voice.h index 5b02bd49fde4..4e4ecc21760b 100644 --- a/sound/drivers/opl3/opl3_voice.h +++ b/sound/drivers/opl3/opl3_voice.h @@ -41,7 +41,7 @@ void snd_opl3_timer_func(struct timer_list *t); /* Prototypes for opl3_drums.c */ void snd_opl3_load_drums(struct snd_opl3 *opl3); -void snd_opl3_drum_switch(struct snd_opl3 *opl3, int note, int on_off, int vel, struct snd_midi_channel *chan); +void snd_opl3_drum_switch(struct snd_opl3 *opl3, int note, int vel, int on_off, struct snd_midi_channel *chan); /* Prototypes for opl3_oss.c */ #if IS_ENABLED(CONFIG_SND_SEQUENCER_OSS) -- 2.20.1

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Re: [alsa-devel] [PATCH 1/2] ASoC: rt5682: fix jack type detection issue
by Mark Brown 18 Mar '19

18 Mar '19

On Mon, Mar 18, 2019 at 11:33:01AM +0000, Shuming [范書銘] wrote: > Hi Mark, > > This patch based on "ASoC: rt5682: Check JD status when system resume" patch. > But "ASoC: rt5682: Check JD status when system resume" patch doesn't apply yet. > Could you please check "ASoC: rt5682: Check JD status when system resume" patch first ? > https://patchwork.kernel.org/patch/10844139/ I have seen that, it's really hard to understand. In any case, please resend like I said in the mail.

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[alsa-devel] [PATCH] ASoC: simple-amplifier: Make gpio property optional
by Maxime Ripard 18 Mar '19

18 Mar '19

From: Mylène Josserand <mylene.josserand(a)bootlin.com> Some amplifier may not have a GPIO to control the power, but instead simply rely on the regulator to power up and down the amplifier. In order to support those setups, let's make the GPIO optional. Signed-off-by: Mylène Josserand <mylene.josserand(a)bootlin.com> Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- Documentation/devicetree/bindings/sound/simple-amplifier.txt | 2 +- sound/soc/codecs/simple-amplifier.c | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt index 7182ac4f1e65..b1b097cc9b68 100644 --- a/Documentation/devicetree/bindings/sound/simple-amplifier.txt +++ b/Documentation/devicetree/bindings/sound/simple-amplifier.txt @@ -2,9 +2,9 @@ Simple Amplifier Audio Driver Required properties: - compatible : "dioo,dio2125" or "simple-audio-amplifier" -- enable-gpios : the gpio connected to the enable pin of the simple amplifier Optional properties: +- enable-gpios : the gpio connected to the enable pin of the simple amplifier - VCC-supply : power supply for the device, as covered in Documentation/devicetree/bindings/regulator/regulator.txt diff --git a/sound/soc/codecs/simple-amplifier.c b/sound/soc/codecs/simple-amplifier.c index c07e8a80b4b7..351aa55c384e 100644 --- a/sound/soc/codecs/simple-amplifier.c +++ b/sound/soc/codecs/simple-amplifier.c @@ -89,7 +89,8 @@ static int simple_amp_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, priv); - priv->gpiod_enable = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); + priv->gpiod_enable = devm_gpiod_get_optional(dev, "enable", + GPIOD_OUT_LOW); if (IS_ERR(priv->gpiod_enable)) { err = PTR_ERR(priv->gpiod_enable); if (err != -EPROBE_DEFER) -- 2.20.1

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[alsa-devel] [PATCH 5/5] ASoC: rsnd: dma: use extended audio dmac registers when available
by Jiada Wang 18 Mar '19

18 Mar '19

Some of SoCs have both basic and extended dmac registers set basic set only supports busif0 ~ busif3, in order to use busif4 ~ busif7, extended audio dmac registers need to be used. This patch changes to use extended dmac registers set when it is available in device-tree. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/dma.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0324a5c39619..905502ccedba 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -830,7 +830,10 @@ int rsnd_dma_probe(struct rsnd_priv *priv) /* * for Gen2 or later */ - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "audmapp"); + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "extaudmapp"); + if (!res) + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, + "audmapp"); dmac = devm_kzalloc(dev, sizeof(*dmac), GFP_KERNEL); if (!dmac || !res) { dev_err(dev, "dma allocate failed\n"); -- 2.19.2

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