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

[alsa-devel] [PATCH] ASoC: Use __printf markup to silence compiler
by Mathieu Malaterre 12 Feb '19

12 Feb '19

Silence warnings (triggered at W=1) by adding relevant __printf attributes. sound/soc/soc-dapm.c:149:2: warning: function 'pop_dbg' might be a candidate for 'gnu_printf' format attribute [-Wsuggest-attribute=format] Signed-off-by: Mathieu Malaterre <malat(a)debian.org> --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a5178845065b..92a2e7dab5fe 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -133,6 +133,7 @@ static void pop_wait(u32 pop_time) schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time)); } +__printf(3, 4) static void pop_dbg(struct device *dev, u32 pop_time, const char *fmt, ...) { va_list args; -- 2.19.2

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[alsa-devel] [PATCH 0/2][RFC] ASoC: soc-core: modern style dai_link control
by Kuninori Morimoto 12 Feb '19

12 Feb '19

Hi Mark These are [RFC] patches which adds new rule for modern style dai_link. Many sound are using CPU component as Platform component, and then it has below settings on legacy style dai_link dai_link->platform_of_node = dai_link->cpu_of_node But, this duplicated component will be just ignored because it is already selected when CPU component binding. We want to allow "no Platform" on modern style dai_link. And, current legacy style dai_link allows no Platform settings, then, soc-core will select snd-soc-dummy platform for it automatically. In modern style, we need to have CPU/Codec/Platform as array/pointer style. Thus, this kind of "implicit" selection will be just un-understandable. These 2 patches adds new rule for modern style, and have no effect to legacy style. But, I can't test for *all* board/situation. Thus, I added [RFC] for these. Kuninori Morimoto (2): ASoC: soc-core: allow no Platform on dai_link ASoC: soc-core: no more implicit snd-soc-dummy platform on modern dai_link include/sound/soc.h | 2 +- sound/soc/generic/simple-card-utils.c | 4 ++ sound/soc/soc-core.c | 80 ++++++++++++++++++++++------------- 3 files changed, 56 insertions(+), 30 deletions(-) -- 2.7.4

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[alsa-devel] pcm_dshare calculates size incorrectly on buffer rollover
by Brendan Shanks 11 Feb '19

11 Feb '19

I'm working on an embedded system based on an Ambarella H1 SoC (32-bit ARM Cortex A9). Audio playback through ALSA (and GStreamer) works fine when playing to the raw hw device. When playing through dshare (my normal configuration), playback stops after exactly 7 hours 16 minutes, for about 3 minutes. During the 3 minutes, the playback thread consumes an entire CPU core. After the 3 minutes, GStreamer reports an xrun, recovers from it, and playback goes back to normal. After some debugging, the problem seems to be in snd_pcm_dshare_sync_area(). When dshare->appl_ptr rolls over to 0, 'size' becomes huge, 3036676576. 'slave_size' is also much bigger than it should be, 1258291680. This is what 'size' is set to when the for loop starts, and I believe the for loop then spends ~3 minutes copying a huge amount of samples. This also explains the 7h16m time, it's linked to the PCM boundary which is 1258291200. At 48 kHz, 1258291200 samples takes 7h16m54s. I'm not sure what the fix should be though. Is this really a bug in dshare, or are bad values being set somewhere else? GStreamer? Or maybe the period/buffer size (480/9600) is causing problems? Any advice is appreciated, my modified snd_pcm_dshare_sync_area() and the output are below. Brendan Shanks $ more /proc/asound/card0/pcm1p/sub0/hw_params access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 4 rate: 48000 (48000/1) period_size: 480 buffer_size: 9600 $ more /proc/asound/card0/pcm1p/sub0/sw_params tstamp_mode: ENABLE period_step: 1 avail_min: 480 start_threshold: 1 stop_threshold: 1258291200 silence_threshold: 0 silence_size: 1258291200 boundary: 1258291200 --- output from running 'gst-launch-1.0 -e audiotestsrc ! alsasink' with prints included as below snd_pcm_dshare_sync_area size 3036676576, dshare->appl_ptr 0 dshare->last_appl_ptr 1258290720 slave_hw_ptr1 1258282080 slave_period_size 480 slave_hw_ptr2 1258282080 slave_buffer_size 9600 slave_hw_ptr3 1258291680 slave_boundary 1258291200 slave_hw_ptr4 1258291680 slave_appl_ptr 0 slave_size1 1258291680 size 1258291680 appl_ptr 9120 size 1258291680 boundary 1258291200 last_appl_ptr 0 slave_appl_ptr 0 dshare->slave_appl_ptr 480 exiting snd_pcm_dshare_sync_area size 3036676576, dshare->appl_ptr 0 dshare->last_appl_ptr 1258290720 slave_hw_ptr1 8965920 slave_period_size 480 slave_hw_ptr2 8965920 slave_buffer_size 9600 slave_hw_ptr3 8975520 slave_boundary 1258291200 slave_hw_ptr4 8975520 slave_appl_ptr 8975040 slave_size1 480 size 480 appl_ptr 9120 size 480 boundary 1258291200 last_appl_ptr 0 slave_appl_ptr 8640 dshare->slave_appl_ptr 8975520 exiting --- snd_pcm_dshare_sync_area() from alsa-lib-1.1.6 with prints added: static void snd_pcm_dshare_sync_area(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; snd_pcm_uframes_t slave_hw_ptr, slave_appl_ptr, slave_size; snd_pcm_uframes_t appl_ptr, size; const snd_pcm_channel_area_t *src_areas, *dst_areas; int print = 0; /* calculate the size to transfer */ size = dshare->appl_ptr - dshare->last_appl_ptr; if (! size) return; if (size > 9600) { printf("snd_pcm_dshare_sync_area size %lu, dshare->appl_ptr %lu dshare->last_appl_ptr %lu\n", size, dshare->appl_ptr, dshare->last_appl_ptr); print = 1; } slave_hw_ptr = dshare->slave_hw_ptr; if (print) printf("slave_hw_ptr1 %lu slave_period_size %lu", slave_hw_ptr, dshare->slave_period_size); /* don't write on the last active period - this area may be cleared * by the driver during write operation... */ slave_hw_ptr -= slave_hw_ptr % dshare->slave_period_size; if (print) printf("slave_hw_ptr2 %lu slave_buffer_size %lu", slave_hw_ptr, dshare->slave_buffer_size); slave_hw_ptr += dshare->slave_buffer_size; if (print) printf("slave_hw_ptr3 %lu slave_boundary %lu", slave_hw_ptr, dshare->slave_boundary); if (dshare->slave_hw_ptr > dshare->slave_boundary) slave_hw_ptr -= dshare->slave_boundary; if (print) printf("slave_hw_ptr4 %lu slave_appl_ptr %lu", slave_hw_ptr, dshare->slave_appl_ptr); if (slave_hw_ptr < dshare->slave_appl_ptr) slave_size = slave_hw_ptr + (dshare->slave_boundary - dshare->slave_appl_ptr); else slave_size = slave_hw_ptr - dshare->slave_appl_ptr; if (print) printf("slave_size1 %lu", slave_size); if (slave_size < size) size = slave_size; if (! size) return; if (print) printf("size %lu", size); /* add sample areas here */ src_areas = snd_pcm_mmap_areas(pcm); dst_areas = snd_pcm_mmap_areas(dshare->spcm); appl_ptr = dshare->last_appl_ptr % pcm->buffer_size; if (print) printf("appl_ptr %lu size %lu boundary %lu", appl_ptr, size, pcm->boundary); dshare->last_appl_ptr += size; dshare->last_appl_ptr %= pcm->boundary; if (print) printf("last_appl_ptr %lu", dshare->last_appl_ptr); slave_appl_ptr = dshare->slave_appl_ptr % dshare->slave_buffer_size; if (print) printf("slave_appl_ptr %lu", slave_appl_ptr); dshare->slave_appl_ptr += size; dshare->slave_appl_ptr %= dshare->slave_boundary; if (print) printf("dshare->slave_appl_ptr %lu", dshare->slave_appl_ptr); for (;;) { snd_pcm_uframes_t transfer = size; if (appl_ptr + transfer > pcm->buffer_size) transfer = pcm->buffer_size - appl_ptr; if (slave_appl_ptr + transfer > dshare->slave_buffer_size) transfer = dshare->slave_buffer_size - slave_appl_ptr; share_areas(dshare, src_areas, dst_areas, appl_ptr, slave_appl_ptr, transfer); size -= transfer; if (! size) break; slave_appl_ptr += transfer; slave_appl_ptr %= dshare->slave_buffer_size; appl_ptr += transfer; appl_ptr %= pcm->buffer_size; } if (print) printf("exiting");

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[alsa-devel] [PATCH 1/1] seq: arecordmidi: Add num-events option
by pmalani＠chromium.org 11 Feb '19

11 Feb '19

From: Prashant Malani <pmalani(a)chromium.org> Add a command line option to automatically exit after recording a fixed number of MIDI events. This allows a program using arecordmidi to expect a MIDI file to be written automatically when the specified number of events have been received, instead of having to send a SIGINT or SIGTERM programmatically. It also avoids the need to have the arecordmidi process running in the background, and then constantly stat the output file to check if any bytes have been written to it (this makes for less predictable and longer-running tests). This functionality finds use in Chrome OS functional testing, since having to send SIGTERM/SIGINT programmatically and then wait for the output file adds unpredictability and delay to the tests. The addition of this command-line option should (hopefully) not break any existing usage. Signed-off-by: Prashant Malani <pmalani(a)chromium.org> diff --git a/seq/aplaymidi/arecordmidi.c b/seq/aplaymidi/arecordmidi.c index 19dbb7d..ca25bfa 100644 --- a/seq/aplaymidi/arecordmidi.c +++ b/seq/aplaymidi/arecordmidi.c @@ -690,7 +690,8 @@ static void help(const char *argv0) " -t,--ticks=ticks resolution in ticks per beat or frame\n" " -s,--split-channels create a track for each channel\n" " -m,--metronome=client:port play a metronome signal\n" - " -i,--timesig=nn:dd time signature\n", + " -i,--timesig=nn:dd time signature\n" + " -n,--num-events=events fixed number of events to record, then exit\n", argv0); } @@ -706,7 +707,7 @@ static void sighandler(int sig) int main(int argc, char *argv[]) { - static const char short_options[] = "hVlp:b:f:t:sdm:i:"; + static const char short_options[] = "hVlp:b:f:t:sdm:i:n:"; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, @@ -719,6 +720,7 @@ int main(int argc, char *argv[]) {"dump", 0, NULL, 'd'}, {"metronome", 1, NULL, 'm'}, {"timesig", 1, NULL, 'i'}, + {"num-events", 1, NULL, 'n'}, { } }; @@ -727,6 +729,9 @@ int main(int argc, char *argv[]) struct pollfd *pfds; int npfds; int c, err; + /* If |num_events| isn't specified, leave it at 0. */ + int num_events = 0; + int events_received = 0; init_seq(); @@ -775,6 +780,11 @@ int main(int argc, char *argv[]) case 'i': time_signature(optarg); break; + case 'n': + num_events = atoi(optarg); + if (num_events <= 0) + fatal("num_events must be greater than 0"); + break; default: help(argv[0]); return 1; @@ -864,13 +874,20 @@ int main(int argc, char *argv[]) err = snd_seq_event_input(seq, &event); if (err < 0) break; - if (event) + if (event) { record_event(event); + events_received++; + } } while (err > 0); if (stop) break; + if (num_events && (events_received == num_events)) + break; } + if (num_events && events_received < num_events) + fputs("Warning: Received signal before num_events\n", stdout); + finish_tracks(); write_file(); -- 2.20.1.495.gaa96b0ce6b-goog

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[alsa-devel] Async ops SOC compress issue
by b_lkasam＠codeaurora.org 08 Feb '19

08 Feb '19

Hi ALSA team, Need inputs wrt one issue seen with soc-compress code. Please help check below issue seen in soc-compress file wrt handling multiple BEs with async ops ASYNC_DPCM_SND_SOC_HW_PARAMS set for given FE dailink/BE dailinks also. Issue : pcm_prepare failure when 2 BE's on same FE enabled during compress playback, seen only when both BE's have SOC_HW_PARAM defined as an async_op. Analysis: - With SOC_HW_PARAMS defined as async_op on both BE's, hw_params api for both of these BE's are scheduled for asynchronous execution. - It is observed that dpcm_be_hw_params_prepare is getting invoked twice, but for the same BE. Below are the issue logs taken for above scenario. 12-21 16:38:32.074 0 0 D SDM660 Compress1: connected new DPCM playback path SDM660 Compress1 -> INT4_MI2S_RX 12-21 16:38:32.075 0 0 D SDM660 Compress1: connected new DPCM playback path SDM660 Compress1 -> INT0_MI2S_RX 12-21 16:38:32.075 0 0 D INT0_MI2S_RX: ASoC: open playback BE INT0_MI2S_RX 12-21 16:38:32.076 0 0 D INT4_MI2S_RX: ASoC: open playback BE INT4_MI2S_RX 12-21 16:38:32.077 0 0 E soc_compr_set_params_fe: : async schedule hw_params for be INT0_MI2S_RX 12-21 16:38:32.077 0 0 E soc_compr_set_params_fe: : async schedule hw_params for be INT4_MI2S_RX 12-21 16:38:32.078 0 0 E dpcm_be_hw_params_prepare: : BE INT4_MI2S_RX -> dpcm_fe_dai_hw_params_be 12-21 16:38:32.079 0 0 E msm-dai-q6-mi2s soc: qcom,msm-dai-mi2s:qcom,msm-dai-q6-int-mi2s4: msm_dai_q6_mi2s_hw_params: dai id 11 dai_data->channels = 1 12-21 16:38:32.143 0 0 E dpcm_be_hw_params_prepare: : BE INT4_MI2S_RX -> dpcm_fe_dai_hw_params_be 12-21 16:38:32.173 0 0 E msm-dai-q6-mi2s soc: qcom,msm-dai-mi2s:qcom,msm-dai-q6-int-mi2s4: msm_dai_q6_mi2s_hw_params: ERROR dai id 11 dai_data->channels = 1 It is not clear why dpcm_be_hw_params_prepare is happening for same BE twice even though async schedule was done for both. - With async_synchronize call after scheduling of hw_param, the behavior is as expected. hw_param for both BE's are success. diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index ad3cc68..b0a1995 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -535,6 +535,7 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, } } } + async_synchronize_full_domain(&async_domain); - Adding this code fixing the issue, but looks not correct way, please suggest for (i = 0; i < j; i++) { cstream->be = be_list[i]; dpcm_be_hw_params_prepare(cstream); With regards Laxminath Kasam

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix implicit fb endpoint setup by quirk
by manuel.rhdt＠gmail.com 07 Feb '19

07 Feb '19

The commit a60945fd08e4 ("ALSA: usb-audio: move implicit fb quirks to separate function") introduced an error in the handling of quirks for implicit feedback endpoints. This commit fixes this. If a quirk successfully sets up an implicit feedback endpoint, usb-audio no longer tries to find the implicit fb endpoint itself. Signed-off-by: Manuel Reinhardt <manuel.rhdt(a)gmail.com> --- sound/usb/pcm.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 382847154227..db114f3977e0 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -314,6 +314,9 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, return 0; } +/* Setup an implicit feedback endpoint from a quirk. Returns 0 if no quirk + * applies. Returns 1 if a quirk was found. + */ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, struct usb_device *dev, struct usb_interface_descriptor *altsd, @@ -384,7 +387,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; - return 0; + return 1; } static int set_sync_endpoint(struct snd_usb_substream *subs, @@ -423,6 +426,10 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, if (err < 0) return err; + /* endpoint set by quirk */ + if (err > 0) + return 0; + if (altsd->bNumEndpoints < 2) return 0; -- 2.20.1

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[alsa-devel] [PATCH v15 0/8] ASoC: sun4i-codec: Add Line-In, FM-In, Mic 2
by Danny Milosavljevic 07 Feb '19

07 Feb '19

This patchset adds some mixer controls to sun4i-codec for the Allwinner A10 and the Allwinner A20. It also adds the PGA for the MIC2 preamp. Where possible, it uses SOC_DAPM_DOUBLE in order to cut down on the number of distinct controls in alsamixer. v15 changes compared to v14 are: - Instead of adding controls dynamically, add an extra sun7i_codec_controls structure and duplicate the sun4i_codec_controls there, with small changes. - Split Mic Playback Volume into extra patch. - LEFT OFF adding Capture Selects since it's unclear how we want to structure it. - LEFT OFF adding Differential Line Source since it might also change if the Capture Select implementation is different. v14 changes compared to v13 are: - Merged some of the patches together if it made sense to test them together. - Use snd_soc_component_driver. - Moved SUN4I_CODEC_DAC_ACTL_LFMS, SUN4I_CODEC_DAC_ACTL_RFMS to the correct patch in the series. - Kept "Left Mixer Left DAC Playback Switch", "Right Mixer Left DAC Playback Switch" and "Right Mixer Right DAC Playback Switch" unchanged compared to the released version - for backward compatibility. v13 changes compared to v12 are: - Added my "Signed-off-by". - Clarified some commit message text. v12 changes compared to v11 are: - Split up patchset in another way. - Renamed "Mic1 Capture Volume" to "Mic1 Boost Volume". - Renamed "Mic2 Capture Volume" to "Mic2 Boost Volume". - Renamed "Line Capture Volume" to "Line Boost Volume". - Renamed "Differential Line Capture Switch" to "Differential Line Source". v11 changes compared to v10 are: - Split up patchset. - Fixed typo in Differential Line Capture Switch. - Renamed "Non-Differential" value to "Stereo". - Removed duplicate PA Volume mixer control. v10 changes compared to v9 are: - Use SOC_DAPM_DOUBLE where possible and it makes sense in order to cut down on the number of controls. v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (8): ASoC: sun4i-codec: Add MIC2 Pre-Amplifier, Mic2 ASoC: sun4i-codec: Add Mic Playback Volume ASoC: sun4i-codec: Add sun7i_codec_controls, sun7i_codec_codec. ASoC: sun4i-codec: Add Mic1 Boost Volume, Mic2 Boost Volume ASoC: sun4i-codec: Merge sun4i_codec_left_mixer_controls and sun4i_codec_right_mixer_controls into sun4i_codec_mixer_controls ASoC: sun4i-codec: Add Mic1 Playback Switch, Mic2 Playback Switch ASoC: sun4i-codec: Add FM Playback Volume, FM Left, FM Right, FM Playback Switch ASoC: sun4i-codec: Add Line Playback Volume, Line Boost Volume, Line Right, Line Left, Line Playback Switch sound/soc/sunxi/sun4i-codec.c | 149 ++++++++++++++++++++++++++++++---- 1 file changed, 134 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH v10 0/4] Media Device Allocator API
by Shuah Khan 06 Feb '19

06 Feb '19

Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. - This patch series is tested on 5.0-rc3 and addresses comments on v9 series from Hans Verkuil. - v9 was tested on 4.20-rc6. - Tested sharing resources with kaffeine, vlc, xawtv, tvtime, and arecord. When analog is streaming, digital and audio user-space applications detect that the tuner is busy and exit. When digital is streaming, analog and audio applications detect that the tuner is busy and exit. When arecord is owns the tuner, digital and analog detect that the tuner is busy and exit. - Tested media device allocator API with bind/unbind testing on snd-usb-audio and au0828 drivers to make sure /dev/mediaX is released only when the last driver is unbound. - Addressed review comments from Hans on the RFC v8 (rebased on 4.19) - Updated change log to describe the use-case more clearly. - No changes to 0001,0002 code since the v7 referenced below. - 0003 is a new patch to enable ALSA defines that have been disabled for kernel between 4.9 and 4.19. - Minor merge conflict resolution in 0004. - Added SPDX to new files. Changes since v9: - Patch 1: Fix mutex assert warning from find_module() calls. This code was written before the change to find_module() that requires callers to hold module_mutex. I missed this during my testing on 4.20-rc6. Hans Verkuil reported the problem. - Patch 4: sound/usb: Initializes all the entities it can before registering the device based on comments from Hans Verkuil - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - No changes to Patches 2 and 3. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (4): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 41 ++++ drivers/media/Makefile | 4 + drivers/media/media-dev-allocator.c | 144 +++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 53 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 327 +++++++++++++++++++++++++ sound/usb/media.h | 74 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 18 files changed, 723 insertions(+), 22 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.17.1

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[alsa-devel] [PATCH v2] ALSA: hda/tegra: enable clock during probe
by Sameer Pujar 05 Feb '19

05 Feb '19

If CONFIG_PM is disabled or runtime PM calls are forbidden, the clocks will not be ON. This could cause issue during probe, where hda init setup is done. This patch enables clocks unconditionally during probe. Along with above, follwoing changes are done. * enable runtime PM before exiting from probe work. This helps to avoid usage of pm_runtime_get_sync/pm_runtime_put() in probe work. * hda_tegra_disable_clocks() is moved out of CONFIG_PM_SLEEP check. * runtime PM callbacks moved out of CONFIG_PM check Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Reviewed-by: Ravindra Lokhande <rlokhande(a)nvidia.com> Reviewed-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 26 +++++++++++++++++--------- 1 file changed, 17 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index c8d18dc..ba6175f 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -219,7 +219,6 @@ static int hda_tegra_enable_clocks(struct hda_tegra *data) return rc; } -#ifdef CONFIG_PM_SLEEP static void hda_tegra_disable_clocks(struct hda_tegra *data) { clk_disable_unprepare(data->hda2hdmi_clk); @@ -227,6 +226,7 @@ static void hda_tegra_disable_clocks(struct hda_tegra *data) clk_disable_unprepare(data->hda_clk); } +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -257,7 +257,6 @@ static int hda_tegra_resume(struct device *dev) } #endif /* CONFIG_PM_SLEEP */ -#ifdef CONFIG_PM static int hda_tegra_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -283,7 +282,7 @@ static int hda_tegra_runtime_resume(struct device *dev) int rc; rc = hda_tegra_enable_clocks(hda); - if (rc != 0) + if (rc) return rc; if (chip && chip->running) { hda_tegra_init(hda); @@ -292,7 +291,6 @@ static int hda_tegra_runtime_resume(struct device *dev) return 0; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops hda_tegra_pm = { SET_SYSTEM_SLEEP_PM_OPS(hda_tegra_suspend, hda_tegra_resume) @@ -551,9 +549,9 @@ static int hda_tegra_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, card); - pm_runtime_enable(hda->dev); - if (!azx_has_pm_runtime(chip)) - pm_runtime_forbid(hda->dev); + err = hda_tegra_enable_clocks(hda); + if (err) + goto out_free; schedule_work(&hda->probe_work); @@ -571,7 +569,6 @@ static void hda_tegra_probe_work(struct work_struct *work) struct platform_device *pdev = to_platform_device(hda->dev); int err; - pm_runtime_get_sync(hda->dev); err = hda_tegra_first_init(chip, pdev); if (err < 0) goto out_free; @@ -592,8 +589,15 @@ static void hda_tegra_probe_work(struct work_struct *work) chip->running = 1; snd_hda_set_power_save(&chip->bus, power_save * 1000); + /* set device state as active */ + if (pm_runtime_set_active(hda->dev) < 0) + goto out_free; + /* enable runtime PM */ + pm_runtime_enable(hda->dev); + if (!azx_has_pm_runtime(chip)) + pm_runtime_forbid(hda->dev); + out_free: - pm_runtime_put(hda->dev); return; /* no error return from async probe */ } @@ -603,6 +607,10 @@ static int hda_tegra_remove(struct platform_device *pdev) ret = snd_card_free(dev_get_drvdata(&pdev->dev)); pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) { + hda_tegra_runtime_suspend(&pdev->dev); + pm_runtime_set_suspended(&pdev->dev); + } return ret; } -- 2.7.4

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[alsa-devel] [PATCH] ASoC: soc-core: clear platform pointers on error
by Curtis Malainey 04 Feb '19

04 Feb '19

Originally snd_soc_init_platform was not cleaning up its pointers, this was fixed to always reallocate dynamic memory but created a memory leak when snd_soc_init_platform was called multiple times during the same probe attempt and also threw away any changes made to the struct between calls. In order to avoid reallocating memory that is still valid, the behaviour will be changed to clear the dynamically set pointers on a probe error and a unregister event and snd_soc_init_platform will go back to its original behaviour of only allocating null pointers so it will stop throwing away valid changes. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/soc-core.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d6b5edba8d2d..9431dd605103 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1053,7 +1053,7 @@ static int snd_soc_init_platform(struct snd_soc_card *card, * soc.h :: struct snd_soc_dai_link */ /* convert Legacy platform link */ - if (!platform || dai_link->legacy_platform) { + if (!platform) { platform = devm_kzalloc(card->dev, sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); @@ -1076,6 +1076,24 @@ static int snd_soc_init_platform(struct snd_soc_card *card, return 0; } +static void soc_cleanup_platform(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *link; + int i; + /* + * FIXME + * + * this function should be removed with snd_soc_init_platform + */ + + for_each_card_prelinks(card, i, link) { + if (link->legacy_platform) { + link->legacy_platform = 0; + link->platforms = NULL; + } + } +} + static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -2020,6 +2038,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) /* remove and free each DAI */ soc_remove_dai_links(card); soc_remove_pcm_runtimes(card); + soc_cleanup_platform(card); /* remove auxiliary devices */ soc_remove_aux_devices(card); -- 2.20.1.495.gaa96b0ce6b-goog

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