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

[PATCH] alsa-tools/hdspmixer: enhance preset save and add hardware ouput loopback buttons
by fassl 27 Feb '21

27 Feb '21

>From 3c887b31fa57ca3004c2eaf0ee5abc0b3f94da70 Mon Sep 17 00:00:00 2001 From: Jasmin Fazlic <superfassl(a)gmail.com> Date: Thu, 4 Feb 2021 15:58:33 +0100 Subject: [PATCH 1/2] alsa-tools/hdspmixer: enhance saving of presets Changing the version in the file header would make a preset file not readable by older versions of the tool. If we just append new data always at the end of the save procedure we should have no problems reading them with different versions, as they all just read to a certain point and ignore the rest of the file. This patch implements the logic to save the presets first to a file called file_name.tmp and appends any extra data that would come after in a possibly present file_name file. Any data written by newer versions would remain in the preset file and from now on no old version should remove data written by newer versions. Also since we write to a temporary file and rename afterwards an extra feature is gained of not corrupting the preset should we crash. Signed-off-by: Jasmin Fazlic <superfassl(a)gmail.com> --- hdspmixer/src/HDSPMixerWindow.cxx | 98 ++++++++++++++++++++----------- 1 file changed, 64 insertions(+), 34 deletions(-) diff --git a/hdspmixer/src/HDSPMixerWindow.cxx b/hdspmixer/src/HDSPMixerWindow.cxx index 9efc25d..3b3d668 100644 --- a/hdspmixer/src/HDSPMixerWindow.cxx +++ b/hdspmixer/src/HDSPMixerWindow.cxx @@ -353,18 +353,25 @@ void HDSPMixerWindow::save() sizeof(inputs->strips[0]->data[0][0][0]->fader_pos) / sizeof(inputs->strips[0]->data[0][0][0]->fader_pos[0])); - - FILE *file; - - if ((file = fopen(file_name, "w")) == NULL) { - fl_alert("Error opening file %s for saving", file_name); + FILE *in,*out; + + /* We want to append any existing extra data that might got written by a + * newer version to this file, therefore write our data to file_name.tmp + * and append the old data. Also this way we would not corrupt the file + * should we crash. + */ + std::string const tmp = file_name + std::string(".tmp"); + char const * const tmpc = tmp.c_str(); + + if ((out = fopen(tmpc, "w")) == NULL) { + fl_alert("Error opening file %s for saving", tmpc); } if (dirty) { inputs->buttons->presets->save_preset(current_preset+1); } /* since hdspmixer 1.11, we also store the meter level settings. Indicate * the new on-disk structure via a small header */ - if (fwrite((void *)&header, sizeof(char), sizeof(header), file) != + if (fwrite((void *)&header, sizeof(char), sizeof(header), out) != sizeof(header)) { goto save_error; } @@ -374,99 +381,122 @@ void HDSPMixerWindow::save() for (int preset = 0; preset < 8; ++preset) { for (int channel = 0; channel < HDSP_MAX_CHANNELS; ++channel) { /* inputs pans and volumes */ - if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->pan_pos[0]), sizeof(int), pan_array_size, file) != pan_array_size) { + if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->pan_pos[0]), sizeof(int), pan_array_size, out) != pan_array_size) { goto save_error; } - if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->fader_pos[0]), sizeof(int), pan_array_size, file) != pan_array_size) { + if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->fader_pos[0]), sizeof(int), pan_array_size, out) != pan_array_size) { goto save_error; } /* playbacks pans and volumes */ - if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->pan_pos[0]), sizeof(int), pan_array_size, file) != pan_array_size) { + if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->pan_pos[0]), sizeof(int), pan_array_size, out) != pan_array_size) { goto save_error; } - if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->fader_pos[0]), sizeof(int), pan_array_size, file) != pan_array_size) { + if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->fader_pos[0]), sizeof(int), pan_array_size, out) != pan_array_size) { goto save_error; } /* inputs mute/solo/dest */ - if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->mute), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->mute), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->solo), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->solo), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->dest), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(inputs->strips[channel]->data[card][speed][preset]->dest), sizeof(int), 1, out) != 1) { goto save_error; } /* playbacks mute/solo/dest */ - if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->mute), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->mute), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->solo), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->solo), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->dest), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(playbacks->strips[channel]->data[card][speed][preset]->dest), sizeof(int), 1, out) != 1) { goto save_error; } /* outputs volumes */ - if (fwrite((void *)&(outputs->strips[channel]->data[card][speed][preset]->fader_pos), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(outputs->strips[channel]->data[card][speed][preset]->fader_pos), sizeof(int), 1, out) != 1) { goto save_error; } } /* Lineouts */ - if (fwrite((void *)&(outputs->strips[HDSP_MAX_CHANNELS]->data[card][speed][preset]->fader_pos), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(outputs->strips[HDSP_MAX_CHANNELS]->data[card][speed][preset]->fader_pos), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(outputs->strips[HDSP_MAX_CHANNELS+1]->data[card][speed][preset]->fader_pos), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(outputs->strips[HDSP_MAX_CHANNELS+1]->data[card][speed][preset]->fader_pos), sizeof(int), 1, out) != 1) { goto save_error; } /* Global settings */ - if (fwrite((void *)&(data[card][speed][preset]->input), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->input), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->output), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->output), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->playback), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->playback), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->submix), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->submix), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->submix_value), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->submix_value), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->solo), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->solo), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->mute), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->mute), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->last_destination), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->last_destination), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->rmsplus3), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->rmsplus3), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->numbers), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->numbers), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->over), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->over), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->level), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->level), sizeof(int), 1, out) != 1) { goto save_error; } - if (fwrite((void *)&(data[card][speed][preset]->rate), sizeof(int), 1, file) != 1) { + if (fwrite((void *)&(data[card][speed][preset]->rate), sizeof(int), 1, out) != 1) { goto save_error; } } } } - fclose(file); + + /* If the file we want to write already exists it could be possible that it + * was saved with a newer version. If that is the case we just append its + * content to the new output file and that way ensure that we don't lose any + * data the new version wrote. + */ + if ((in = fopen(file_name, "r")) != NULL) { + if (!fseek(in, ftell(out), SEEK_SET)) { + char buff[512]; + size_t read; + while ((read = fread(&buff, sizeof(char), sizeof(buff), in)) != 0) + fwrite(buff, sizeof(char), read, out); + if (ferror(in) || ferror(out)) + fl_alert("Error appending %s to %s", file_name, tmpc); + } + fclose(in); + } + + fclose(out); + + if (rename(tmpc, file_name)) + fl_alert("Error renaming %s to %s", tmpc, file_name); + ::remove(tmpc); + return; save_error: - fclose(file); + fclose(out); fl_alert("Error saving presets to file %s", file_name); return; } -- 2.27.0 >From 2dbf462a9126da817db40f4d62207b47e245f3e7 Mon Sep 17 00:00:00 2001 From: Jasmin Fazlic <superfassl(a)gmail.com> Date: Thu, 4 Feb 2021 17:00:22 +0100 Subject: [PATCH 2/2] alsa-tools/hdspmixer: add output loopback buttons This patch adds "LPBK" buttons to the output strip channels allowing to toggle the hardware output loopback state. Efforts were made to hide the buttons for not yet enabled devices but due to difficulties in implementation was cancelled for the time being. Should it be a no-go efforts should be made to fix this and hide them, although I would think fixing the driver for other devices would be an effort with much more merit. This is currently only enabled for HDSP9632 devices with following driver patch: https://github.com/tiwai/sound/commit/da2a040ee7cfe1dd57d5bec7906cb979c5787… Signed-off-by: Jasmin Fazlic <superfassl(a)gmail.com> --- hdspmixer/pixmaps/loopback.xpm | 69 ++++ hdspmixer/pixmaps/output.xpm | 498 +++++++++++++++----------- hdspmixer/pixmaps/output_r.xpm | 498 +++++++++++++++----------- hdspmixer/src/HDSPMixerCard.cxx | 34 ++ hdspmixer/src/HDSPMixerCard.h | 2 + hdspmixer/src/HDSPMixerLoopback.cxx | 133 +++++++ hdspmixer/src/HDSPMixerLoopback.h | 48 +++ hdspmixer/src/HDSPMixerOutput.cxx | 1 + hdspmixer/src/HDSPMixerOutput.h | 3 + hdspmixer/src/HDSPMixerOutputData.h | 1 + hdspmixer/src/HDSPMixerOutputs.cxx | 4 +- hdspmixer/src/HDSPMixerPresetData.cxx | 1 + hdspmixer/src/HDSPMixerPresetData.h | 1 + hdspmixer/src/HDSPMixerPresets.cxx | 2 + hdspmixer/src/HDSPMixerWindow.cxx | 68 ++++ hdspmixer/src/Makefile.am | 2 + hdspmixer/src/defines.h | 2 +- hdspmixer/src/pixmaps.cxx | 1 + hdspmixer/src/pixmaps.h | 1 + 19 files changed, 931 insertions(+), 438 deletions(-) create mode 100644 hdspmixer/pixmaps/loopback.xpm create mode 100644 hdspmixer/src/HDSPMixerLoopback.cxx create mode 100644 hdspmixer/src/HDSPMixerLoopback.h diff --git a/hdspmixer/pixmaps/loopback.xpm b/hdspmixer/pixmaps/loopback.xpm new file mode 100644 index 0000000..a327467 --- /dev/null +++ b/hdspmixer/pixmaps/loopback.xpm @@ -0,0 +1,69 @@ +/* XPM */ +char const * loopback_xpm[] = { +"34 15 51 1", +" c None", +". c #2E3038", +"+ c #FFFB7C", +"@ c #FDF97B", +"# c #2E2D11", +"$ c #000000", +"% c #383716", +"& c #898740", +"* c #F1EE75", +"= c #22210A", +"- c #727034", +"; c #E2DE6D", +"> c #D5D267", +", c #787637", +"' c #FBF77A", +") c #AAA750", +"! c #ABA851", +"~ c #CDC962", +"{ c #83813C", +"] c #C8C560", +"^ c #84823D", +"/ c #F7F378", +"( c #949145", +"_ c #B7B457", +": c #949245", +"< c #353414", +"[ c #87853F", +"} c #F1ED75", +"| c #3B3A17", +"1 c #DFDB6C", +"2 c #161505", +"3 c #4B4A20", +"4 c #E5E16E", +"5 c #FEFA7C", +"6 c #E8E470", +"7 c #535124", +"8 c #FAF679", +"9 c #E7E370", +"0 c #66642E", +"a c #E4E06E", +"b c #F6F277", +"c c #EAE671", +"d c #E0DC6C", +"e c #B9B658", +"f c #5F5D2A", +"g c #EDE973", +"h c #69672F", +"i c #DDD96A", +"j c #333342", +"k c #32323E", +"l c #313136", +"..................................", +".++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++.", +".+++@#++++@$$%&*@$$=-;+@#++>,*+++.", +".+++@#++++@#+')!@#+@~{+@#+]^/++++.", +".+++@#++++@#+'!!@#+@~(+@#_:'+++++.", +".+++@#++++@$$<[}@$$$|1+@234++++++.", +".+++@#++++@#++++@#+5678@#90a+++++.", +".+++@#++++@#++++@#+59#b@#+c0d++++.", +".+++@$$$$e@#++++@$$2f]+@#++ghi+++.", +".++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++.", +"..jklllllllllkj....jklllllllllkj.."}; diff --git a/hdspmixer/pixmaps/output.xpm b/hdspmixer/pixmaps/output.xpm index 8f7cd65..c827b1c 100644 --- a/hdspmixer/pixmaps/output.xpm +++ b/hdspmixer/pixmaps/output.xpm @@ -1,220 +1,282 @@ /* XPM */ char const * output_xpm[] = { -"36 208 9 1", -" c #595966", -". c #2E3038", -"+ c #000000", -"@ c #474951", -"# c #404044", -"$ c #7A7A8F", -"% c #27272B", -"& c #616176", -"* c #737384", -" ", -" .................................. 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+".+#######$%###+++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++&*++++++++@@@@@@@@@@++++++.", +".++++++++++++++++++@@@@@@@@@@++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++==============================++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +".++++++++++++++++++++++++++++++++++.", +"....................................", +".++++++++++++++++++++++++++++++++++.", +".+================================+.", +".+================================+.", +".+================================+.", +".+===-;====->>,')->>!~{=-;==]^)===+.", +".+===-;====-;=/((-;=-_:=-;=<[}====+.", +".+===-;====-;=|((-;=-_1=-;23/=====+.", +".+===-;====->>,[)->>>45=-678======+.", +".+===-;====-;====-;=90a|-;bcd=====+.", +".+===-;====-;====-;=9b;e-;=fcg====+.", +".+===->>>>h-;====->>6ij=-;==klm===+.", +".+================================+.", +".+================================+.", +".+================================+.", +".++nopppppppppon++++nopppppppppon++.", +"...................................."}; diff --git a/hdspmixer/src/HDSPMixerCard.cxx b/hdspmixer/src/HDSPMixerCard.cxx index ce40ba7..72232c8 100644 --- a/hdspmixer/src/HDSPMixerCard.cxx +++ b/hdspmixer/src/HDSPMixerCard.cxx @@ -231,6 +231,8 @@ void HDSPMixerCard::adjustSettings() { /* should never happen */ break; } + + max_channels = sizeof(channel_map_mf_ss); } if (type == Digiface) { @@ -253,6 +255,8 @@ void HDSPMixerCard::adjustSettings() { /* should never happen */ break; } + + max_channels = sizeof(channel_map_df_ss); } if (type == RPM) { @@ -263,6 +267,8 @@ void HDSPMixerCard::adjustSettings() { channel_map_input = channel_map_playback = channel_map_rpm; dest_map = dest_map_rpm; meter_map_input = meter_map_playback = channel_map_rpm; + + max_channels = sizeof(channel_map_rpm); } @@ -286,6 +292,8 @@ void HDSPMixerCard::adjustSettings() { /* should never happen */ break; } + + max_channels = sizeof(channel_map_df_ss); } if (type == H9632) { @@ -312,6 +320,8 @@ void HDSPMixerCard::adjustSettings() { meter_map_input = meter_map_playback = channel_map_h9632_qs; break; } + + max_channels = sizeof(channel_map_h9632_ss); } if (HDSPeMADI == type) { @@ -341,6 +351,7 @@ void HDSPMixerCard::adjustSettings() { break; } + max_channels = sizeof(channel_map_unity_ss); } if (HDSPeAIO == type) { @@ -379,6 +390,7 @@ void HDSPMixerCard::adjustSettings() { break; } + max_channels = sizeof(channel_map_aio_out_ss); } if (HDSP_AES == type) { @@ -394,6 +406,7 @@ void HDSPMixerCard::adjustSettings() { meter_map_input = channel_map_aes32; meter_map_playback = channel_map_aes32; + max_channels = sizeof(channel_map_aes32); } if (HDSPeRayDAT == type) { @@ -426,6 +439,7 @@ void HDSPMixerCard::adjustSettings() { break; } + max_channels = sizeof(channel_map_raydat_ss); } window_width = (channels_playback+2)*STRIP_WIDTH; @@ -545,3 +559,23 @@ int HDSPMixerCard::initializeCard(HDSPMixerWindow *w) return 0; } +int HDSPMixerCard::supportsLoopback() const +{ + int err = 0; + snd_ctl_elem_value_t *elemval; + snd_ctl_elem_id_t * elemid; + snd_ctl_t *handle; + snd_ctl_elem_value_alloca(&elemval); + snd_ctl_elem_id_alloca(&elemid); + if ((err = snd_ctl_open(&handle, name, SND_CTL_NONBLOCK)) < 0) + return err; + + snd_ctl_elem_id_set_name(elemid, "Output Loopback"); + snd_ctl_elem_id_set_interface(elemid, SND_CTL_ELEM_IFACE_HWDEP); + snd_ctl_elem_id_set_index(elemid, 0); + snd_ctl_elem_value_set_id(elemval, elemid); + err = snd_ctl_elem_read(handle, elemval); + snd_ctl_close(handle); + + return err; +} diff --git a/hdspmixer/src/HDSPMixerCard.h b/hdspmixer/src/HDSPMixerCard.h index faaeefa..eecfca3 100644 --- a/hdspmixer/src/HDSPMixerCard.h +++ b/hdspmixer/src/HDSPMixerCard.h @@ -52,6 +52,7 @@ public: HDSPMixerCard(int cardtype, int id, char *shortname); int channels_input, channels_playback, window_width, window_height, card_id; int channels_output; + int max_channels; int type; int last_preset; /* Last activated preset before switching to another card */ int last_dirty; /* Last dirty flag before switching to another card */ @@ -68,6 +69,7 @@ public: void adjustSettings(); void getAeb(); hdsp_9632_aeb_t h9632_aeb; + int supportsLoopback() const; }; #endif diff --git a/hdspmixer/src/HDSPMixerLoopback.cxx b/hdspmixer/src/HDSPMixerLoopback.cxx new file mode 100644 index 0000000..fcfec2a --- /dev/null +++ b/hdspmixer/src/HDSPMixerLoopback.cxx @@ -0,0 +1,133 @@ +/* + * HDSPMixer + * + * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#pragma implementation +#include "HDSPMixerLoopback.h" + +HDSPMixerLoopback::HDSPMixerLoopback(int x, int y, int idx):Fl_Widget(x, y, 34, 15) +{ + basew = (HDSPMixerWindow *)window(); + index = idx; +} + +void HDSPMixerLoopback::draw() +{ + if (_loopback == 1) + fl_draw_pixmap(loopback_xpm, x(), y()); +} + +int HDSPMixerLoopback::get() +{ + auto const card { basew->cards[basew->current_card] }; + + if (card->supportsLoopback() != 0) + return -1; + + if (index >= card->max_channels) + return -1; + + int err; + snd_ctl_elem_value_t *elemval; + snd_ctl_elem_id_t * elemid; + snd_ctl_t *handle; + snd_ctl_elem_value_alloca(&elemval); + snd_ctl_elem_id_alloca(&elemid); + char const * const name = basew->cards[basew->current_card]->name; + if ((err = snd_ctl_open(&handle, name, SND_CTL_NONBLOCK)) < 0) { + fprintf(stderr, "Error accessing ctl interface on card %s\n.", name); + return -1; + } + + snd_ctl_elem_id_set_name(elemid, "Output Loopback"); + snd_ctl_elem_id_set_interface(elemid, SND_CTL_ELEM_IFACE_HWDEP); + snd_ctl_elem_id_set_index(elemid, index); + snd_ctl_elem_value_set_id(elemval, elemid); + if ((err = snd_ctl_elem_read(handle, elemval)) < 0) + fprintf(stderr, "cannot read loopback: %d\n", err); + else + _loopback = snd_ctl_elem_value_get_integer(elemval, 0); + + snd_ctl_close(handle); + + return _loopback; +} + +void HDSPMixerLoopback::set(int l) +{ + auto const card { basew->cards[basew->current_card] }; + + if (card->supportsLoopback() != 0) + return; + + if (index >= card->max_channels) + return; + + if (l != _loopback) { + int err; + + snd_ctl_elem_id_t *id; + snd_ctl_elem_value_t *ctl; + snd_ctl_t *handle; + + snd_ctl_elem_value_alloca(&ctl); + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_id_set_name(id, "Output Loopback"); + snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_HWDEP); + snd_ctl_elem_id_set_device(id, 0); + snd_ctl_elem_id_set_index(id, index); + snd_ctl_elem_value_set_id(ctl, id); + + if ((err = snd_ctl_open( + &handle, basew->cards[basew->current_card]->name, SND_CTL_NONBLOCK)) < 0) { + fprintf(stderr, "Alsa error 1: %s\n", snd_strerror(err)); + return; + } + + snd_ctl_elem_value_set_integer(ctl, 0, l); + if ((err = snd_ctl_elem_write(handle, ctl)) < 0) { + fprintf(stderr, "Alsa error 2: %s\n", snd_strerror(err)); + snd_ctl_close(handle); + return; + } + + _loopback = l; + + snd_ctl_close(handle); + + redraw(); + } +} + +int HDSPMixerLoopback::handle(int e) +{ + int button3 = Fl::event_button3(); + switch (e) { + case FL_PUSH: + set(!_loopback); + if (button3) + relative->set(_loopback); + basew->checkState(); + redraw(); + return 1; + default: + return Fl_Widget::handle(e); + } +} + diff --git a/hdspmixer/src/HDSPMixerLoopback.h b/hdspmixer/src/HDSPMixerLoopback.h new file mode 100644 index 0000000..07f1f0c --- /dev/null +++ b/hdspmixer/src/HDSPMixerLoopback.h @@ -0,0 +1,48 @@ +/* + * HDSPMixer + * + * Copyright (C) 2003 Thomas Charbonnel (thomas(a)undata.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#pragma interface +#ifndef HDSPMixerLoopback_H +#define HDSPMixerLoopback_H + +#include <FL/Fl.H> +#include <FL/Fl_Widget.H> +#include "HDSPMixerWindow.h" +#include "pixmaps.h" + +class HDSPMixerWindow; + +class HDSPMixerLoopback:public Fl_Widget +{ +private: + HDSPMixerWindow *basew; + int _loopback{-1}; +public: + HDSPMixerLoopback *relative; + int index; + HDSPMixerLoopback(int x, int y, int idx); + void draw(); + int handle(int e); + int get(); + void set(int l); +}; + +#endif + diff --git a/hdspmixer/src/HDSPMixerOutput.cxx b/hdspmixer/src/HDSPMixerOutput.cxx index 0053fe0..7f80a74 100644 --- a/hdspmixer/src/HDSPMixerOutput.cxx +++ b/hdspmixer/src/HDSPMixerOutput.cxx @@ -204,6 +204,7 @@ HDSPMixerOutput::HDSPMixerOutput(int x, int y, int w, int h, int num):Fl_Group(x peak = new HDSPMixerPeak(x+3, y+4, 0); gain = new HDSPMixerGain(x+3, y+175, 0); meter = new HDSPMixerMeter(x+20, y+27, false, peak); + loopback = new HDSPMixerLoopback(x+1, y+208, out_num); end(); } diff --git a/hdspmixer/src/HDSPMixerOutput.h b/hdspmixer/src/HDSPMixerOutput.h index 6278cfd..1e04ae5 100644 --- a/hdspmixer/src/HDSPMixerOutput.h +++ b/hdspmixer/src/HDSPMixerOutput.h @@ -27,6 +27,7 @@ #include <FL/fl_draw.H> #include <alsa/sound/hdsp.h> #include "HDSPMixerFader.h" +#include "HDSPMixerLoopback.h" #include "HDSPMixerPeak.h" #include "HDSPMixerGain.h" #include "HDSPMixerMeter.h" @@ -36,6 +37,7 @@ class HDSPMixerFader; class HDSPMixerGain; +class HDSPMixerLoopback; class HDSPMixerPeak; class HDSPMixerMeter; class HDSPMixerOutputData; @@ -56,6 +58,7 @@ public: HDSPMixerFader *fader; HDSPMixerGain *gain; HDSPMixerMeter *meter; + HDSPMixerLoopback *loopback; HDSPMixerOutput(int x, int y, int w, int h, int out); void draw(); void draw_background(); diff --git a/hdspmixer/src/HDSPMixerOutputData.h b/hdspmixer/src/HDSPMixerOutputData.h index 885047b..75b9109 100644 --- a/hdspmixer/src/HDSPMixerOutputData.h +++ b/hdspmixer/src/HDSPMixerOutputData.h @@ -26,6 +26,7 @@ class HDSPMixerOutputData { public: int fader_pos; + int loopback; HDSPMixerOutputData(); }; diff --git a/hdspmixer/src/HDSPMixerOutputs.cxx b/hdspmixer/src/HDSPMixerOutputs.cxx index 0b4e7f4..342685d 100644 --- a/hdspmixer/src/HDSPMixerOutputs.cxx +++ b/hdspmixer/src/HDSPMixerOutputs.cxx @@ -25,13 +25,15 @@ HDSPMixerOutputs::HDSPMixerOutputs(int x, int y, int w, int h, int nchans):Fl_Gr { int i; for (i = 0; i < HDSP_MAX_CHANNELS+2; i += 2) { - strips[i] = new HDSPMixerOutput((i*STRIP_WIDTH), y, STRIP_WIDTH, SMALLSTRIP_HEIGHT, i); + strips[i] = new HDSPMixerOutput((i*STRIP_WIDTH), y, STRIP_WIDTH, SMALLSTRIP_HEIGHT, i); strips[i+1] = new HDSPMixerOutput(((i+1)*STRIP_WIDTH), y, STRIP_WIDTH, SMALLSTRIP_HEIGHT, i+1); /* Setup linked stereo channels */ strips[i]->fader->relative = strips[i+1]->fader; strips[i+1]->fader->relative = strips[i]->fader; strips[i]->fader->gain = strips[i]->gain; strips[i+1]->fader->gain = strips[i+1]->gain; + strips[i]->loopback->relative = strips[i+1]->loopback; + strips[i+1]->loopback->relative = strips[i]->loopback; } empty_aebo[0] = new HDSPMixerEmpty((nchans-6)*STRIP_WIDTH, y, 2*STRIP_WIDTH, SMALLSTRIP_HEIGHT, 0); diff --git a/hdspmixer/src/HDSPMixerPresetData.cxx b/hdspmixer/src/HDSPMixerPresetData.cxx index 276d101..6359732 100644 --- a/hdspmixer/src/HDSPMixerPresetData.cxx +++ b/hdspmixer/src/HDSPMixerPresetData.cxx @@ -36,5 +36,6 @@ HDSPMixerPresetData::HDSPMixerPresetData() over = 3; level = 0; rate = 1; + loopback = 0; } diff --git a/hdspmixer/src/HDSPMixerPresetData.h b/hdspmixer/src/HDSPMixerPresetData.h index 58536d3..3b46b92 100644 --- a/hdspmixer/src/HDSPMixerPresetData.h +++ b/hdspmixer/src/HDSPMixerPresetData.h @@ -38,6 +38,7 @@ public: int over; int rate; int rmsplus3; + int loopback; HDSPMixerPresetData(); }; diff --git a/hdspmixer/src/HDSPMixerPresets.cxx b/hdspmixer/src/HDSPMixerPresets.cxx index aeeb9c9..82654d9 100644 --- a/hdspmixer/src/HDSPMixerPresets.cxx +++ b/hdspmixer/src/HDSPMixerPresets.cxx @@ -143,6 +143,7 @@ void HDSPMixerPresets::save_preset(int prst) { basew->playbacks->strips[i]->data[card][speed][p]->dest = basew->playbacks->strips[i]->targets->selected; basew->outputs->strips[i]->data[card][speed][p]->fader_pos = basew->outputs->strips[i]->fader->pos[0]; + basew->outputs->strips[i]->data[card][speed][p]->loopback = basew->outputs->strips[i]->loopback->get(); } /* Line outs */ basew->outputs->strips[HDSP_MAX_CHANNELS]->data[card][speed][p]->fader_pos = basew->outputs->strips[HDSP_MAX_CHANNELS]->fader->pos[0]; @@ -188,6 +189,7 @@ void HDSPMixerPresets::restore_preset(int prst) { basew->playbacks->strips[i]->targets->selected = basew->playbacks->strips[i]->data[card][speed][p]->dest; basew->outputs->strips[i]->fader->pos[0] = basew->outputs->strips[i]->data[card][speed][p]->fader_pos; + basew->outputs->strips[i]->loopback->set(basew->outputs->strips[i]->data[card][speed][p]->loopback); } /* Line outs */ basew->outputs->strips[HDSP_MAX_CHANNELS]->fader->pos[0] = basew->outputs->strips[HDSP_MAX_CHANNELS+1]->data[card][speed][p]->fader_pos; diff --git a/hdspmixer/src/HDSPMixerWindow.cxx b/hdspmixer/src/HDSPMixerWindow.cxx index 3b3d668..4a911c1 100644 --- a/hdspmixer/src/HDSPMixerWindow.cxx +++ b/hdspmixer/src/HDSPMixerWindow.cxx @@ -471,6 +471,29 @@ void HDSPMixerWindow::save() } } + /* Output loopback data */ + for (int channel = 0; channel < HDSP_MAX_CHANNELS; ++channel) { + auto const strip = outputs->strips[channel]; + + for (int card = 0; card < MAX_CARDS; ++card) { + auto const data = strip->data[card]; + + for (int speed = 0; speed < 3; ++speed) { + auto const spd = data[speed]; + + for (int preset = 0; preset < 8; ++preset) { + auto const data = spd[preset]; + + if (fwrite((void *)&(data->loopback), + sizeof(int), + 1, + out) != 1) + goto save_error; + } + } + } + } + /* If the file we want to write already exists it could be possible that it * was saved with a newer version. If that is the case we just append its * content to the new output file and that way ensure that we don't lose any @@ -519,6 +542,7 @@ void HDSPMixerWindow::load() bool ondisk_v1 = false; int pan_array_size = 14; /* old (pre 1.0.24) HDSP_MAX_DEST */ int channels_per_card = 26; /* old (pre 1.0.24) HDSP_MAX_CHANNELS */ + bool res = true; if (fread(&buffer, sizeof(char), sizeof(buffer), file) != sizeof(buffer)) { goto load_error; @@ -647,6 +671,46 @@ void HDSPMixerWindow::load() } } } + + /* Output loopback data */ + for (int channel = 0; channel < HDSP_MAX_CHANNELS; ++channel) { + auto const strip = outputs->strips[channel]; + + for (int card = 0; card < MAX_CARDS; ++card) { + auto const data = strip->data[card]; + + for (int speed = 0; speed < 3; ++speed) { + auto const spd = data[speed]; + + for (int preset = 0; preset < 8; ++preset) { + auto const data = spd[preset]; + + /* TODO: Somewhere we get a value of 5 from, investigate + * this another day. For now just reset it here and + * continue looping to reset the value. + */ + data->loopback = 0; + + if (feof(file)) { + res = true; + continue; + } + + if (ferror(file)) { + res = false; + continue; + } + + if (fread((void *)&(data->loopback), sizeof(int), 1, file) != 1) + res = false; + } + } + } + } + + if (!res) + goto load_error; + fclose(file); setTitleWithFilename(); resetMixer(); @@ -844,6 +908,8 @@ void HDSPMixerWindow::restoreDefaults(int card) } outputs->strips[i]->data[card][speed][preset]->fader_pos = (preset != 4) ? 137*CF : 0; outputs->strips[i+1]->data[card][speed][preset]->fader_pos = (preset != 4) ? 137*CF : 0; + outputs->strips[i]->data[card][speed][preset]->loopback = 0; + outputs->strips[i+1]->data[card][speed][preset]->loopback = 0; if (preset == 3 || preset == 7) { inputs->strips[i]->data[card][speed][preset]->mute = 1; inputs->strips[i+1]->data[card][speed][preset]->mute = 1; @@ -1051,6 +1117,8 @@ void HDSPMixerWindow::checkState() /* Outputs row */ if (outputs->strips[i]->data[current_card][speed][p]->fader_pos != outputs->strips[i]->fader->pos[0]) corrupt++; + if (outputs->strips[i]->data[current_card][speed][p]->loopback != outputs->strips[i]->loopback->get()) + corrupt++; } /* Global settings */ diff --git a/hdspmixer/src/Makefile.am b/hdspmixer/src/Makefile.am index e80a8ac..ebc4c3c 100644 --- a/hdspmixer/src/Makefile.am +++ b/hdspmixer/src/Makefile.am @@ -15,6 +15,8 @@ hdspmixer_SOURCES = \ HDSPMixerEmpty.h \ HDSPMixerOutput.cxx \ HDSPMixerOutput.h \ + HDSPMixerLoopback.cxx \ + HDSPMixerLoopback.h \ HDSPMixerIOMixer.cxx \ HDSPMixerIOMixer.h \ HDSPMixerSelector.cxx \ diff --git a/hdspmixer/src/defines.h b/hdspmixer/src/defines.h index af5c382..e76141b 100644 --- a/hdspmixer/src/defines.h +++ b/hdspmixer/src/defines.h @@ -34,7 +34,7 @@ #define STRIP_WIDTH 36 #define FULLSTRIP_HEIGHT 253 -#define SMALLSTRIP_HEIGHT 208 +#define SMALLSTRIP_HEIGHT 224 #define MENU_HEIGHT 20 #define MIN_WIDTH 2*STRIP_WIDTH diff --git a/hdspmixer/src/pixmaps.cxx b/hdspmixer/src/pixmaps.cxx index 2f7c589..34f8834 100644 --- a/hdspmixer/src/pixmaps.cxx +++ b/hdspmixer/src/pixmaps.cxx @@ -47,6 +47,7 @@ #include "../pixmaps/over.xpm" #include "../pixmaps/peak.xpm" #include "../pixmaps/solo.xpm" +#include "../pixmaps/loopback.xpm" #include "../pixmaps/iomixer_r.xpm" #include "../pixmaps/output_r.xpm" #include "../pixmaps/matrix_black.xpm" diff --git a/hdspmixer/src/pixmaps.h b/hdspmixer/src/pixmaps.h index b980a62..3548ba3 100644 --- a/hdspmixer/src/pixmaps.h +++ b/hdspmixer/src/pixmaps.h @@ -49,6 +49,7 @@ extern char const * output_xpm[]; extern char const * over_xpm[]; extern char const * peak_xpm[]; extern char const * solo_xpm[]; +extern char const * loopback_xpm[]; extern char const * iomixer_r_xpm[]; extern char const * output_r_xpm[]; extern char const * matrix_white_xpm[]; -- 2.27.0

3 5

[PATCH] ALSA: usb-audio: Don't abort even if the clock rate differs
by Takashi Iwai 27 Feb '21

27 Feb '21

The commit 93db51d06b32 ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3") changed the behavior of the function set_sample_rate_v2v3() slightly to treat the inconsistent sample rate as an error. It was done by assumption that the sample rate validation should have been done at the parser phase as implemented in that patch. But the validation is later selectively enabled only for certain devices as it causes a regression (the commit fe773b8711e3 "ALSA: usb-audio: workaround for iface reset issue"), and now the inconsistency surfaced as a fatal error while it worked in the past as is, as reported for FiiO M3K DAC. For recovering from the regression, change set_sample_rate_v2v3() again to ignore the sample rate difference as non-error. BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1182633 Fixes: 93db51d06b32 ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 8243652d5604..a746802d0ac3 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -652,10 +652,10 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, cur_rate = prev_rate; if (cur_rate != rate) { - usb_audio_warn(chip, - "%d:%d: freq mismatch (RO clock): req %d, clock runs @%d\n", - fmt->iface, fmt->altsetting, rate, cur_rate); - return -ENXIO; + usb_audio_dbg(chip, + "%d:%d: freq mismatch: req %d, clock runs @%d\n", + fmt->iface, fmt->altsetting, rate, cur_rate); + /* continue processing */ } validation: -- 2.26.2

1 0

[PATCH v5 0/9] ALSA: add virtio sound driver
by Anton Yakovlev 27 Feb '21

27 Feb '21

This series implements a driver part of the virtio sound device specification v8 [1]. The driver supports PCM playback and capture substreams, jack and channel map controls. A message-based transport is used to write/read PCM frames to/from a device. As a device part was used OpenSynergy proprietary implementation. v5 changes: - Fixed another bunch of sparse warnings (replaced virtio_cread() -> virtio_cread_le()), no functional changes. (Sorry, I didn't know how to run sparse locally, now everything should be fixed) [1] https://lists.oasis-open.org/archives/virtio-dev/202003/msg00185.html Anton Yakovlev (9): uapi: virtio_ids: add a sound device type ID from OASIS spec ALSA: virtio: add virtio sound driver ALSA: virtio: handling control messages ALSA: virtio: build PCM devices and substream hardware descriptors ALSA: virtio: handling control and I/O messages for the PCM device ALSA: virtio: PCM substream operators ALSA: virtio: introduce jack support ALSA: virtio: introduce PCM channel map support ALSA: virtio: introduce device suspend/resume support MAINTAINERS | 9 + include/uapi/linux/virtio_ids.h | 1 + include/uapi/linux/virtio_snd.h | 334 +++++++++++++++++++++ sound/Kconfig | 2 + sound/Makefile | 3 +- sound/virtio/Kconfig | 10 + sound/virtio/Makefile | 13 + sound/virtio/virtio_card.c | 462 +++++++++++++++++++++++++++++ sound/virtio/virtio_card.h | 113 ++++++++ sound/virtio/virtio_chmap.c | 219 ++++++++++++++ sound/virtio/virtio_ctl_msg.c | 310 ++++++++++++++++++++ sound/virtio/virtio_ctl_msg.h | 78 +++++ sound/virtio/virtio_jack.c | 233 +++++++++++++++ sound/virtio/virtio_pcm.c | 498 ++++++++++++++++++++++++++++++++ sound/virtio/virtio_pcm.h | 120 ++++++++ sound/virtio/virtio_pcm_msg.c | 392 +++++++++++++++++++++++++ sound/virtio/virtio_pcm_ops.c | 491 +++++++++++++++++++++++++++++++ 17 files changed, 3287 insertions(+), 1 deletion(-) create mode 100644 include/uapi/linux/virtio_snd.h create mode 100644 sound/virtio/Kconfig create mode 100644 sound/virtio/Makefile create mode 100644 sound/virtio/virtio_card.c create mode 100644 sound/virtio/virtio_card.h create mode 100644 sound/virtio/virtio_chmap.c create mode 100644 sound/virtio/virtio_ctl_msg.c create mode 100644 sound/virtio/virtio_ctl_msg.h create mode 100644 sound/virtio/virtio_jack.c create mode 100644 sound/virtio/virtio_pcm.c create mode 100644 sound/virtio/virtio_pcm.h create mode 100644 sound/virtio/virtio_pcm_msg.c create mode 100644 sound/virtio/virtio_pcm_ops.c -- 2.30.0

3 27

ASoc: soc_core.c stream direction from snd_soc_dai
by Matt Flax 26 Feb '21

26 Feb '21

Hi there, A large number of audio codecs allow different formats for playback and capture. This becomes very useful when there are different latencies between playback and capture hardware data lines. For example digital isolation chips typically have a 1 bit delay in propagation as the bit clock rate gets faster for higher sample rates. By setting the capture and playback formats to differ by one or two bit clock cycles, the delay problem is solved. There doesn't seem to be a simple way to detect stream direction in the codec driver's set_fmt function. The snd_soc_runtime_set_dai_fmt function : https://github.com/torvalds/linux/blob/master/sound/soc/soc-core.c#L1480 calls the snd_soc_dai_set_fmt function : https://github.com/torvalds/linux/blob/master/sound/soc/soc-dai.c#L101 which calls the set_fmt function : https://github.com/torvalds/linux/blob/master/include/sound/soc-dai.h#L189 The snd_soc_dai_ops set_fmt function is defined as : int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); Is there a simple way to find the stream direction from a snd_soc_dai ? If the stream direction can be detected then the playback and capture formats can be set independently for the codec. It there a different way to set the playback and capture formats for the codec independently at runtime, depending on the sample rate ? Matt

5 8

[PATCH alsa-ucm-conf 0/8] codecs/rt5640: Cleanup + HW volume control support
by Hans de Goede 26 Feb '21

26 Feb '21

Hi All, Here is a patch series consisting of some codecs/rt5640 cleanups and addition of HW volume control support to the rt5640 .conf snippets. Note patch 7 and 8 depend on a new "aif:%d" part being added to the components string, the kernel patches for this are pending upstream in the "[PATCH 0/5] AsoC: rt5640/rt5651: Volume control fixes" patch series. Patches 7 and 8 should probably not be merged until the kernel patches have landed, because theoretically the new component string part could change as part of the review of the kernel patches. Regards, Hans Hans de Goede (8): codecs/rt5640: Cleanup: unify Stereo / Mono ADC handling codecs/rt5640: Cleanup: Initially disable all inputs and outputs codecs/rt5640: Cleanup: Move 'SPK MIX' setup to main EnableSequene codecs/rt5640: Drop bogus 'DIG MIXL DAC L2 Switch' setting codecs/rt5640: Unify capture volume for AIF1 and AIF2 recording paths codecs/rt5640: Add hardware volume-control support codecs/rt5640: Use the new "aif:%d" part of the components string codecs/rt5640: Specify Playback/CaptureMasterElem for HW volume-control ucm2/bytcr-rt5640/HiFi.conf | 46 +++++++++++++++++ ucm2/codecs/rt5640/DigitalMics.conf | 18 ++++++- ucm2/codecs/rt5640/EnableSeq-AIF-unknown.conf | 7 +++ ucm2/codecs/rt5640/EnableSeq-AIF1.conf | 7 +++ ucm2/codecs/rt5640/EnableSeq-AIF2.conf | 7 +++ ucm2/codecs/rt5640/EnableSeq.conf | 50 ++++++++++++++----- ucm2/codecs/rt5640/HeadPhones.conf | 15 +++++- ucm2/codecs/rt5640/HeadsetMic.conf | 24 ++++++--- ucm2/codecs/rt5640/IN1-InternalMic.conf | 27 ++++++---- ucm2/codecs/rt5640/IN3-InternalMic.conf | 27 ++++++---- ucm2/codecs/rt5640/MonoSpeaker.conf | 17 +++++-- ucm2/codecs/rt5640/Speaker.conf | 17 +++++-- 12 files changed, 211 insertions(+), 51 deletions(-) create mode 100644 ucm2/codecs/rt5640/EnableSeq-AIF-unknown.conf create mode 100644 ucm2/codecs/rt5640/EnableSeq-AIF1.conf create mode 100644 ucm2/codecs/rt5640/EnableSeq-AIF2.conf -- 2.30.1

1 8

Possible alsa bug around USB drivers
by Steven Skeard 25 Feb '21

25 Feb '21

Hey all, The last few weeks I've been plagued by inconsistent audio on my PopOS 20.10 desktop. Sometimes on boot everything would be fine, but other times I would have no working audio and only dummy devices in the audio menus. I eventually found that if I specify the modprobe order so that the USB driver (eg. for my webcam) loads after the onboard motherboard driver, that audio will work consistently. Here are the related links: - Reddit thread where I posted the original problem and the solution: https://www.reddit.com/r/pop_os/comments/lmvbj3/audio_not_working_after_upg… - A related thread where it sounds like an arch user has the same problem: https://bbs.archlinux.org/viewtopic.php?id=260373 - The alsa info I uploaded while under the effects of the problem: http://alsa-project.org/db/?f=a23ccbb818fec6eaf08b032504eac5d5d8a30589 I'm not really sure where this problem originated but I figured you guys might be able to point me in the right direction if it is not an alsa problem. I don't think I messed with the original alsa-base.conf but I suppose my memory could be failing me here. I'm not sure where this file comes from but I'm guessing it could be either a pop os, ubuntu, or alsa default configuration? Thanks, Steven Skeard

1 0

[PATCH] ASoC: rt1015p: add acpi device id for rt1015p
by jack.yu＠realtek.com 25 Feb '21

25 Feb '21

From: Jack Yu <jack.yu(a)realtek.com> Add acpi device id for rt1015p. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015p.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 671f2a2130fe..4a00e7a4ef6f 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -4,6 +4,7 @@ // // Copyright 2020 The Linux Foundation. All rights reserved. +#include <linux/acpi.h> #include <linux/delay.h> #include <linux/device.h> #include <linux/err.h> @@ -130,10 +131,21 @@ static const struct of_device_id rt1015p_device_id[] = { MODULE_DEVICE_TABLE(of, rt1015p_device_id); #endif +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt1015p_acpi_match[] = { + { "RT1015", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt1015p_acpi_match); +#endif + static struct platform_driver rt1015p_platform_driver = { .driver = { .name = "rt1015p", .of_match_table = of_match_ptr(rt1015p_device_id), +#ifdef CONFIG_ACPI + .acpi_match_table = ACPI_PTR(rt1015p_acpi_match), +#endif }, .probe = rt1015p_platform_probe, }; -- 2.29.0

3 2

[PATCH] ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 10-p0XX OVCD current threshold
by Hans de Goede 25 Feb '21

25 Feb '21

When I added the quirk for the "HP Pavilion x2 10-p0XX" I copied the byt_rt5640_quirk_table[] entry for the HP Pavilion x2 10-k0XX / 10-n0XX models since these use almost the same settings. While doing this I accidentally also copied and kept the non-standard OVCD_TH_1500UA setting used on those models. This too low threshold is causing headsets to often be seen as headphones (without a headset-mic) and when correctly identified it is causing ghost play/pause button-presses to get detected. Correct the HP Pavilion x2 10-p0XX quirk to use the default OVCD_TH_2000UA setting, fixing these problems. Fixes: fbdae7d6d04d ("ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 Detachable quirks") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 24ced8e4a574..1f6a636571c2 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -581,7 +581,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | BYT_RT5640_JD_SRC_JD1_IN4P | - BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MCLK_EN), }, -- 2.30.1

3 2

[PATCH v2] ALSA: hda/realtek: Enable headset mic of Acer SWIFT with ALC256
by chris.chiu＠canonical.com 25 Feb '21

25 Feb '21

From: Chris Chiu <chris.chiu(a)canonical.com> The Acer SWIFT Swift SF314-54/55 laptops with ALC256 cannot detect both the headset mic and the internal mic. Introduce new fixup to enable the jack sense and the headset mic. However, the internal mic actually connects to Intel SST audio. It still needs Intel SST support to make internal mic capture work. Signed-off-by: Chris Chiu <chris.chiu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1927605f0f7e..4871507cd4bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6406,6 +6406,7 @@ enum { ALC236_FIXUP_DELL_AIO_HEADSET_MIC, ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, + ALC256_FIXUP_ACER_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -7853,6 +7854,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE, }, + [ALC256_FIXUP_ACER_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a1113c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x90a1092f }, /* use as internal mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7879,9 +7890,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK), SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x1269, "Acer SWIFT SF314-54", ALC256_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), -- 2.20.1

1 0

[PATCH] ALSA: hda/realtek: Enable MICs of Acer SWIFT with ALC256
by Chris Chiu 25 Feb '21

25 Feb '21

The Acer SWIFT Swift SF314-54/55 with ALC256 cannot detect the headset microphone and suffers the noise problem in audio capture. This patch enables the headset jack sense and fixes the noise problem with aamix fixup. Signed-off-by: Chris Chiu <chris.chiu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..57e4dbcd76a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6372,6 +6372,8 @@ enum { ALC236_FIXUP_DELL_AIO_HEADSET_MIC, ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, + ALC256_FIXUP_ACER_SWIFT_NO_MIC_PRESENCE, + ALC256_FIXUP_ACER_SWIFT_DISABLE_AAMIX, }; static const struct hda_fixup alc269_fixups[] = { @@ -7815,6 +7817,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE, }, + [ALC256_FIXUP_ACER_SWIFT_NO_MIC_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a1113c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x90a1092f }, /* use as internal mic */ + { }, + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, + [ALC256_FIXUP_ACER_SWIFT_DISABLE_AAMIX] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC256_FIXUP_ACER_SWIFT_NO_MIC_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7841,9 +7859,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK), SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x1269, "Acer SWIFT SF314-54", ALC256_FIXUP_ACER_SWIFT_DISABLE_AAMIX), SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_SWIFT_DISABLE_AAMIX), SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), -- 2.20.1

2 2