March 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0_synth.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0_synth.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0_synth.c 2007-03-25 17:16:29.000000000 +0200 @@ -147,47 +147,6 @@ static void vortex_wt_connect(vortex_t * } } -/* Read WT Register */ -#if 0 -static int vortex_wt_GetReg(vortex_t * vortex, char reg, int wt) -{ - //int eax, esi; - - if (reg == 4) { - return hwread(vortex->mmio, WT_PARM(wt, 3)); - } - if (reg == 7) { - return hwread(vortex->mmio, WT_GMODE(wt)); - } - - return 0; -} - -/* WT hardware abstraction layer generic register interface. */ -static int -vortex_wt_SetReg2(vortex_t * vortex, unsigned char reg, int wt, - u16 val) -{ - /* - int eax, edx; - - if (wt >= NR_WT) // 0x40 -> NR_WT - return 0; - - if ((reg - 0x20) > 0) { - if ((reg - 0x21) != 0) - return 0; - eax = ((((b & 0xff) << 0xb) + (edx & 0xff)) << 4) + 0x208; // param 2 - } else { - eax = ((((b & 0xff) << 0xb) + (edx & 0xff)) << 4) + 0x20a; // param 3 - } - hwwrite(vortex->mmio, eax, c); - */ - return 1; -} - -/*public: static void __thiscall CWTHal::SetReg(unsigned char,int,unsigned long) */ -#endif static int vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, u32 val) @@ -315,81 +274,4 @@ static void vortex_wt_init(vortex_t * vo vortex_wt_SetReg(vortex, 0xa, edi, var10); /* ctrl */ } -/* Extract of CAdbTopology::SetVolume(struct _ASPVOLUME *) */ -#if 0 -static void vortex_wt_SetVolume(vortex_t * vortex, int wt, int vol[]) -{ - wt_voice_t *voice = &(vortex->wt_voice[wt]); - int ecx = vol[1], eax = vol[0]; - - /* This is pure guess */ - voice->parm0 &= 0xff00ffff; - voice->parm0 |= (vol[0] & 0xff) << 0x10; - voice->parm1 &= 0xff00ffff; - voice->parm1 |= (vol[1] & 0xff) << 0x10; - - /* This is real */ - hwwrite(vortex, WT_PARM(wt, 0), voice->parm0); - hwwrite(vortex, WT_PARM(wt, 1), voice->parm0); - - if (voice->this_1D0 & 4) { - eax >>= 8; - ecx = eax; - if (ecx < 0x80) - ecx = 0x7f; - voice->parm3 &= 0xFFFFC07F; - voice->parm3 |= (ecx & 0x7f) << 7; - voice->parm3 &= 0xFFFFFF80; - voice->parm3 |= (eax & 0x7f); - } else { - voice->parm3 &= 0xFFE03FFF; - voice->parm3 |= (eax & 0xFE00) << 5; - } - - hwwrite(vortex, WT_PARM(wt, 3), voice->parm3); -} - -/* Extract of CAdbTopology::SetFrequency(unsigned long arg_0) */ -static void vortex_wt_SetFrequency(vortex_t * vortex, int wt, unsigned int sr) -{ - wt_voice_t *voice = &(vortex->wt_voice[wt]); - u32 eax, edx; - - //FIXME: 64 bit operation. - eax = ((sr << 0xf) * 0x57619F1) & 0xffffffff; - edx = (((sr << 0xf) * 0x57619F1)) >> 0x20; - - edx >>= 0xa; - edx <<= 1; - if (edx) { - if (edx & 0x0FFF80000) - eax = 0x7fff; - else { - edx <<= 0xd; - eax = 7; - while ((edx & 0x80000000) == 0) { - edx <<= 1; - eax--; - if (eax == 0) ; - break; - } - if (eax) - edx <<= 1; - eax <<= 0xc; - edx >>= 0x14; - eax |= edx; - } - } else - eax = 0; - voice->parm0 &= 0xffff0001; - voice->parm0 |= (eax & 0x7fff) << 1; - voice->parm1 = voice->parm0 | 1; - // Wt: this_1D4 - //AuWt::WriteReg((ulong)(this_1DC<<4)+0x200, (ulong)this_1E4); - //AuWt::WriteReg((ulong)(this_1DC<<4)+0x204, (ulong)this_1E8); - hwwrite(vortex->mmio, WT_PARM(wt, 0), voice->parm0); - hwwrite(vortex->mmio, WT_PARM(wt, 1), voice->parm1); -} -#endif - /* End of File */

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[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0_wt.h remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> mjander(a)users.sourceforge.org --- linux-work5/sound/pci/au88x0/au88x0_wt.h 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0_wt.h 2007-03-25 16:57:17.000000000 +0200 @@ -33,25 +33,6 @@ #define WT_PARM(x,y) (((WT_BAR(x))+ 0x80 +(((x)&0x1f)<<2)+(y))<<2) /* 0x0200 */ #define WT_DELAY(x,y) (((WT_BAR(x))+ 0x100 +(((x)&0x1f)<<2)+(y))<<2) /* 0x0400 */ -/* Numeric indexes used by SetReg() and GetReg() */ -#if 0 -enum { - run = 0, /* 0 W 1:run 0:stop */ - parm0, /* 1 W filter, samplerate */ - parm1, /* 2 W filter, samplerate */ - parm2, /* 3 W */ - parm3, /* 4 RW volume. This value is calculated using floating point ops. */ - sramp, /* 5 W */ - mute, /* 6 W 1:mute, 0:unmute */ - gmode, /* 7 RO Looks like only bit0 is used. */ - aramp, /* 8 W */ - mramp, /* 9 W */ - ctrl, /* a W */ - delay, /* b W All 4 values are written at once with same value. */ - dsreg, /* c (R)W */ -} wt_reg; -#endif - typedef struct { u32 parm0; /* this_1E4 */ u32 parm1; /* this_1E8 */

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[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0.h remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0.h 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0.h 2007-03-25 16:32:57.000000000 +0200 @@ -253,11 +253,6 @@ static void vortex_wt_init(vortex_t * vo static void vortex_route(vortex_t * vortex, int en, unsigned char channel, unsigned char source, unsigned char dest); -#if 0 -static void vortex_routes(vortex_t * vortex, int en, unsigned char channel, - unsigned char source, unsigned char dest0, - unsigned char dest1); -#endif static void vortex_connection_mixin_mix(vortex_t * vortex, int en, unsigned char mixin, unsigned char mix, int a);

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[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0_core.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0_core.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0_core.c 2007-03-25 17:06:31.000000000 +0200 @@ -97,85 +97,6 @@ static void vortex_mixer_dis_sr(vortex_t hwread(vortex->mmio, VORTEX_MIXER_SR) & ~(0x1 << channel)); } -#if 0 -static void -vortex_mix_muteinputgain(vortex_t * vortex, unsigned char mix, - unsigned char channel) -{ - hwwrite(vortex->mmio, VORTEX_MIX_INVOL_A + ((mix << 5) + channel), - 0x80); - hwwrite(vortex->mmio, VORTEX_MIX_INVOL_B + ((mix << 5) + channel), - 0x80); -} - -static int vortex_mix_getvolume(vortex_t * vortex, unsigned char mix) -{ - int a; - a = hwread(vortex->mmio, VORTEX_MIX_VOL_A + (mix << 2)) & 0xff; - //FP2LinearFrac(a); - return (a); -} - -static int -vortex_mix_getinputvolume(vortex_t * vortex, unsigned char mix, - int channel, int *vol) -{ - int a; - if (!(mchannels[mix] & (1 << channel))) - return 0; - a = hwread(vortex->mmio, - VORTEX_MIX_INVOL_A + (((mix << 5) + channel) << 2)); - /* - if (rampchs[mix] == 0) - a = FP2LinearFrac(a); - else - a = FP2LinearFracWT(a); - */ - *vol = a; - return (0); -} - -static unsigned int vortex_mix_boost6db(unsigned char vol) -{ - return (vol + 8); /* WOW! what a complex function! */ -} - -static void vortex_mix_rampvolume(vortex_t * vortex, int mix) -{ - int ch; - char a; - // This function is intended for ramping down only (see vortex_disableinput()). - for (ch = 0; ch < 0x20; ch++) { - if (((1 << ch) & rampchs[mix]) == 0) - continue; - a = hwread(vortex->mmio, - VORTEX_MIX_INVOL_B + (((mix << 5) + ch) << 2)); - if (a > -126) { - a -= 2; - hwwrite(vortex->mmio, - VORTEX_MIX_INVOL_A + - (((mix << 5) + ch) << 2), a); - hwwrite(vortex->mmio, - VORTEX_MIX_INVOL_B + - (((mix << 5) + ch) << 2), a); - } else - vortex_mix_killinput(vortex, mix, ch); - } -} - -static int -vortex_mix_getenablebit(vortex_t * vortex, unsigned char mix, int mixin) -{ - int addr, temp; - if (mixin >= 0) - addr = mixin; - else - addr = mixin + 3; - addr = ((mix << 3) + (addr >> 2)) << 2; - temp = hwread(vortex->mmio, VORTEX_MIX_ENIN + addr); - return ((temp >> (mixin & 3)) & 1); -} -#endif static void vortex_mix_setvolumebyte(vortex_t * vortex, unsigned char mix, unsigned char vol) @@ -500,61 +421,6 @@ vortex_src_persist_convratio(vortex_t * return temp; } -#if 0 -static void vortex_src_slowlock(vortex_t * vortex, unsigned char src) -{ - int temp; - - hwwrite(vortex->mmio, VORTEX_SRC_DRIFT2 + (src << 2), 1); - hwwrite(vortex->mmio, VORTEX_SRC_DRIFT0 + (src << 2), 0); - temp = hwread(vortex->mmio, VORTEX_SRC_U0 + (src << 2)); - if (temp & 0x200) - hwwrite(vortex->mmio, VORTEX_SRC_U0 + (src << 2), - temp & ~0x200L); -} - -static void -vortex_src_change_convratio(vortex_t * vortex, unsigned char src, int ratio) -{ - int temp, a; - - if ((ratio & 0x10000) && (ratio != 0x10000)) { - if (ratio & 0x3fff) - a = (0x11 - ((ratio >> 0xe) & 0x3)) - 1; - else - a = (0x11 - ((ratio >> 0xe) & 0x3)) - 2; - } else - a = 0xc; - temp = hwread(vortex->mmio, VORTEX_SRC_U0 + (src << 2)); - if (((temp >> 4) & 0xf) != a) - hwwrite(vortex->mmio, VORTEX_SRC_U0 + (src << 2), - (temp & 0xf) | ((a & 0xf) << 4)); - - vortex_src_persist_convratio(vortex, src, ratio); -} - -static int -vortex_src_checkratio(vortex_t * vortex, unsigned char src, - unsigned int desired_ratio) -{ - int hw_ratio, lifeboat = 0; - - hw_ratio = hwread(vortex->mmio, VORTEX_SRC_CONVRATIO + (src << 2)); - - while (hw_ratio != desired_ratio) { - hwwrite(vortex->mmio, VORTEX_SRC_CONVRATIO + (src << 2), desired_ratio); - - if ((lifeboat++) > 15) { - printk(KERN_ERR "Vortex: could not set src-%d from %d to %d\n", - src, hw_ratio, desired_ratio); - break; - } - } - - return hw_ratio; -} - -#endif /* Objective: Set samplerate for given SRC module. Arguments: @@ -784,19 +650,6 @@ vortex_fifo_clearadbdata(vortex_t * vort (((fifo << FIFO_SIZE_BITS) + x) << 2), 0); } -#if 0 -static void vortex_fifo_adbinitialize(vortex_t * vortex, int fifo, int j) -{ - vortex_fifo_clearadbdata(vortex, fifo, FIFO_SIZE); -#ifdef CHIP_AU8820 - hwwrite(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2), - (FIFO_U1 | ((j & FIFO_MASK) << 0xb))); -#else - hwwrite(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2), - (FIFO_U1 | ((j & FIFO_MASK) << 0xc))); -#endif -} -#endif static void vortex_fifo_setadbvalid(vortex_t * vortex, int fifo, int en) { hwwrite(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2), @@ -1329,22 +1182,6 @@ static void vortex_adbdma_pausefifo(vort dma->fifo_status = FIFO_PAUSE; } -#if 0 // Using pause instead -static void vortex_adbdma_stopfifo(vortex_t * vortex, int adbdma) -{ - stream_t *dma = &vortex->dma_adb[adbdma]; - - int this_4 = 0, this_8 = 0; - if (dma->fifo_status == FIFO_START) - vortex_fifo_setadbctrl(vortex, adbdma, dma->dma_unknown, - this_4, this_8, 0, 0); - else if (dma->fifo_status == FIFO_STOP) - return; - dma->fifo_status = FIFO_STOP; - dma->fifo_enabled = 0; -} - -#endif /* WTDMA */ #ifndef CHIP_AU8810 @@ -1483,22 +1320,6 @@ static int vortex_wtdma_bufshift(vortex_ return delta; } -#if 0 -static void -vortex_wtdma_getposition(vortex_t * vortex, int wtdma, int *subbuf, int *pos) -{ - int temp; - temp = hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)); - *subbuf = (temp >> WT_SUBBUF_SHIFT) & WT_SUBBUF_MASK; - *pos = temp & POS_MASK; -} - -static int vortex_wtdma_getcursubuffer(vortex_t * vortex, int wtdma) -{ - return ((hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)) >> - POS_SHIFT) & POS_MASK); -} -#endif static int inline vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma) { stream_t *dma = &vortex->dma_wt[wtdma]; @@ -1743,40 +1564,6 @@ vortex_route(vortex_t * vortex, int en, } } -#if 0 -static void -vortex_routes(vortex_t * vortex, int en, unsigned char channel, - unsigned char source, unsigned char dest0, unsigned char dest1) -{ - ADBRamLink route[2]; - - route[0] = ((source & ADB_MASK) << ADB_SHIFT) | (dest0 & ADB_MASK); - route[1] = ((source & ADB_MASK) << ADB_SHIFT) | (dest1 & ADB_MASK); - - if (en) { - vortex_adb_addroutes(vortex, channel, route, 2); - if ((source < (OFFSET_SRCOUT + NR_SRC)) - && (source >= (OFFSET_SRCOUT))) - vortex_src_addWTD(vortex, (source - OFFSET_SRCOUT), - channel); - else if ((source < (OFFSET_MIXOUT + NR_MIXOUT)) - && (source >= (OFFSET_MIXOUT))) - vortex_mixer_addWTD(vortex, - (source - OFFSET_MIXOUT), channel); - } else { - vortex_adb_delroutes(vortex, channel, route[0], route[1]); - if ((source < (OFFSET_SRCOUT + NR_SRC)) - && (source >= (OFFSET_SRCOUT))) - vortex_src_delWTD(vortex, (source - OFFSET_SRCOUT), - channel); - else if ((source < (OFFSET_MIXOUT + NR_MIXOUT)) - && (source >= (OFFSET_MIXOUT))) - vortex_mixer_delWTD(vortex, - (source - OFFSET_MIXOUT), channel); - } -} - -#endif /* Route two sources to same target. Sources must be of same class !!! */ static void vortex_routeLRT(vortex_t * vortex, int en, unsigned char ch, @@ -1899,44 +1686,6 @@ vortex_connection_mix_src(vortex_t * vor vortex_mix_setvolumebyte(vortex, mix, MIX_DEFOGAIN); // added to original code. } -#if 0 -static void -vortex_connection_adbdma_src_src(vortex_t * vortex, int en, - unsigned char channel, - unsigned char adbdma, unsigned char src0, - unsigned char src1) -{ - vortex_routes(vortex, en, channel, ADB_DMA(adbdma), - ADB_SRCIN(src0), ADB_SRCIN(src1)); -} - -// Connect two mix to AdbDma. -static void -vortex_connection_mix_mix_adbdma(vortex_t * vortex, int en, - unsigned char ch, unsigned char mix0, - unsigned char mix1, unsigned char adbdma) -{ - - ADBRamLink routes[2]; - routes[0] = - (((mix0 + - OFFSET_MIXOUT) & ADB_MASK) << ADB_SHIFT) | (adbdma & ADB_MASK); - routes[1] = - (((mix1 + OFFSET_MIXOUT) & ADB_MASK) << ADB_SHIFT) | ((adbdma + - 0x20) & - ADB_MASK); - if (en) { - vortex_adb_addroutes(vortex, ch, routes, 0x2); - vortex_mixer_addWTD(vortex, mix0, ch); - vortex_mixer_addWTD(vortex, mix1, ch); - } else { - vortex_adb_delroutes(vortex, ch, routes[0], routes[1]); - vortex_mixer_delWTD(vortex, mix0, ch); - vortex_mixer_delWTD(vortex, mix1, ch); - } -} -#endif - /* CODEC connect. */ static void @@ -1946,7 +1695,6 @@ vortex_connect_codecplay(vortex_t * vort vortex_connection_mix_adb(vortex, en, 0x11, mixers[0], ADB_CODECOUT(0)); vortex_connection_mix_adb(vortex, en, 0x11, mixers[1], ADB_CODECOUT(1)); #else -#if 1 // Connect front channels through EQ. vortex_connection_mix_adb(vortex, en, 0x11, mixers[0], ADB_EQIN(0)); vortex_connection_mix_adb(vortex, en, 0x11, mixers[1], ADB_EQIN(1)); @@ -1965,11 +1713,6 @@ vortex_connect_codecplay(vortex_t * vort ADB_CODECOUT(1 + 4)); //printk("SDAC detected "); } -#else - // Use plain direct output to codec. - vortex_connection_mix_adb(vortex, en, 0x11, mixers[0], ADB_CODECOUT(0)); - vortex_connection_mix_adb(vortex, en, 0x11, mixers[1], ADB_CODECOUT(1)); -#endif #endif } @@ -2356,20 +2099,6 @@ static void vortex_settimer(vortex_t * v hwwrite(vortex->mmio, VORTEX_IRQ_STAT, period); } -#if 0 -static void vortex_enable_timer_int(vortex_t * card) -{ - hwwrite(card->mmio, VORTEX_IRQ_CTRL, - hwread(card->mmio, VORTEX_IRQ_CTRL) | IRQ_TIMER | 0x60); -} - -static void vortex_disable_timer_int(vortex_t * card) -{ - hwwrite(card->mmio, VORTEX_IRQ_CTRL, - hwread(card->mmio, VORTEX_IRQ_CTRL) & ~IRQ_TIMER); -} - -#endif static void vortex_enable_int(vortex_t * card) { // CAsp4ISR__EnableVortexInt_void_

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[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0_eq.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0_eq.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0_eq.c 2007-03-25 17:13:31.000000000 +0200 @@ -142,34 +142,6 @@ static void vortex_EqHw_SetRightStates(v } } -#if 0 -static void vortex_EqHw_GetTimeConsts(vortex_t * vortex, u16 * a, u16 * b) -{ - *a = hwread(vortex->mmio, 0x2b3c4); - *b = hwread(vortex->mmio, 0x2b3c8); -} - -static void vortex_EqHw_GetLeftCoefs(vortex_t * vortex, u16 a[]) -{ - -} - -static void vortex_EqHw_GetRightCoefs(vortex_t * vortex, u16 a[]) -{ - -} - -static void vortex_EqHw_GetLeftStates(vortex_t * vortex, u16 * a, u16 b[]) -{ - -} - -static void vortex_EqHw_GetRightStates(vortex_t * vortex, u16 * a, u16 b[]) -{ - -} - -#endif /* Mix Gains */ static void vortex_EqHw_SetBypassGain(vortex_t * vortex, u16 a, u16 b) { @@ -190,22 +162,6 @@ static void vortex_EqHw_SetA3DBypassGain hwwrite(vortex->mmio, 0x2b3f8, b); } -#if 0 -static void vortex_EqHw_SetCurrBypassGain(vortex_t * vortex, u16 a, u16 b) -{ - - hwwrite(vortex->mmio, 0x2b3d0, a); - hwwrite(vortex->mmio, 0x2b3e8, b); -} - -static void vortex_EqHw_SetCurrA3DBypassGain(vortex_t * vortex, u16 a, u16 b) -{ - - hwwrite(vortex->mmio, 0x2b3dc, a); - hwwrite(vortex->mmio, 0x2b3f4, b); -} - -#endif static void vortex_EqHw_SetLeftGainsSingleTarget(vortex_t * vortex, u16 index, u16 b) { @@ -258,68 +214,6 @@ static void vortex_EqHw_SetRightGainsCur } } -#if 0 -static void vortex_EqHw_GetLeftGainsTarget(vortex_t * vortex, u16 a[]) -{ - eqhw_t *eqhw = &(vortex->eq.this04); - int ebx = 0; - - if (eqhw->this04 < 0) - return; - - do { - a[ebx] = hwread(vortex->mmio, 0x2b02c + ebx * 0x30); - ebx++; - } - while (ebx < eqhw->this04); -} - -static void vortex_EqHw_GetRightGainsTarget(vortex_t * vortex, u16 a[]) -{ - eqhw_t *eqhw = &(vortex->eq.this04); - int ebx = 0; - - if (eqhw->this04 < 0) - return; - - do { - a[ebx] = hwread(vortex->mmio, 0x2b20c + ebx * 0x30); - ebx++; - } - while (ebx < eqhw->this04); -} - -static void vortex_EqHw_GetLeftGainsCurrent(vortex_t * vortex, u16 a[]) -{ - eqhw_t *eqhw = &(vortex->eq.this04); - int ebx = 0; - - if (eqhw->this04 < 0) - return; - - do { - a[ebx] = hwread(vortex->mmio, 0x2b028 + ebx * 0x30); - ebx++; - } - while (ebx < eqhw->this04); -} - -static void vortex_EqHw_GetRightGainsCurrent(vortex_t * vortex, u16 a[]) -{ - eqhw_t *eqhw = &(vortex->eq.this04); - int ebx = 0; - - if (eqhw->this04 < 0) - return; - - do { - a[ebx] = hwread(vortex->mmio, 0x2b208 + ebx * 0x30); - ebx++; - } - while (ebx < eqhw->this04); -} - -#endif /* EQ band levels settings */ static void vortex_EqHw_SetLevels(vortex_t * vortex, u16 peaks[]) { @@ -344,38 +238,6 @@ static void vortex_EqHw_SetLevels(vortex hwwrite(vortex->mmio, 0x2b3f0, peaks[3 + (eqhw->this04 * 2)]); } -#if 0 -static void vortex_EqHw_GetLevels(vortex_t * vortex, u16 a[]) -{ - eqhw_t *eqhw = &(vortex->eq.this04); - int ebx; - - if (eqhw->this04 < 0) - return; - - ebx = 0; - do { - a[ebx] = hwread(vortex->mmio, 0x2b024 + ebx * 0x30); - ebx++; - } - while (ebx < eqhw->this04); - - a[eqhw->this04] = hwread(vortex->mmio, 0x2b3cc); - a[eqhw->this04 + 1] = hwread(vortex->mmio, 0x2b3d8); - - ebx = 0; - do { - a[ebx + (eqhw->this04 + 2)] = - hwread(vortex->mmio, 0x2b204 + ebx * 0x30); - ebx++; - } - while (ebx < eqhw->this04); - - a[2 + (eqhw->this04 * 2)] = hwread(vortex->mmio, 0x2b3e4); - a[3 + (eqhw->this04 * 2)] = hwread(vortex->mmio, 0x2b3f0); -} - -#endif /* Global Control */ static void vortex_EqHw_SetControlReg(vortex_t * vortex, u32 reg) { @@ -387,18 +249,6 @@ static void vortex_EqHw_SetSampleRate(vo hwwrite(vortex->mmio, 0x2b440, ((sr & 0x1f) << 3) | 0xb800); } -#if 0 -static void vortex_EqHw_GetControlReg(vortex_t * vortex, u32 *reg) -{ - *reg = hwread(vortex->mmio, 0x2b440); -} - -static void vortex_EqHw_GetSampleRate(vortex_t * vortex, u32 *sr) -{ - *sr = (hwread(vortex->mmio, 0x2b440) >> 3) & 0x1f; -} - -#endif static void vortex_EqHw_Enable(vortex_t * vortex) { hwwrite(vortex->mmio, VORTEX_EQ_CTRL, 0xf001); @@ -552,29 +402,6 @@ static void vortex_Eqlzr_SetRightGain(vo vortex_EqHw_SetRightGainsSingleTarget(vortex, index, gain); } -#if 0 -static int -vortex_Eqlzr_GetAllBands(vortex_t * vortex, u16 * gains, s32 *cnt) -{ - eqlzr_t *eq = &(vortex->eq); - int si = 0; - - if (eq->this10 == 0) - return 1; - - { - if (vortex_Eqlzr_GetLeftGain(vortex, si, &gains[si])) - return 1; - if (vortex_Eqlzr_GetRightGain - (vortex, si, &gains[si + eq->this10])) - return 1; - si++; - } - while (eq->this10 > si) ; - *cnt = si * 2; - return 0; -} -#endif static int vortex_Eqlzr_SetAllBandsFromActiveCoeffSet(vortex_t * vortex) { eqlzr_t *eq = &(vortex->eq); @@ -678,14 +505,6 @@ static int vortex_Eqlzr_GetAllPeaks(vort return 0; } -#if 0 -static auxxEqCoeffSet_t *vortex_Eqlzr_GetActiveCoefSet(vortex_t * vortex) -{ - eqlzr_t *eq = &(vortex->eq); - - return (&(eq->coefset)); -} -#endif static void vortex_Eqlzr_init(vortex_t * vortex) { eqlzr_t *eq = &(vortex->eq);

1 0

[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0.c 2007-03-25 17:10:42.000000000 +0200 @@ -313,23 +313,6 @@ snd_vortex_probe(struct pci_dev *pci, co vortex_gameport_register(chip); -#if 0 - if (snd_seq_device_new(card, 1, SNDRV_SEQ_DEV_ID_VORTEX_SYNTH, - sizeof(snd_vortex_synth_arg_t), &wave) < 0 - || wave == NULL) { - snd_printk(KERN_ERR "Can't initialize Aureal wavetable synth\n"); - } else { - snd_vortex_synth_arg_t *arg; - - arg = SNDRV_SEQ_DEVICE_ARGPTR(wave); - strcpy(wave->name, "Aureal Synth"); - arg->hwptr = vortex; - arg->index = 1; - arg->seq_ports = seq_ports[dev]; - arg->max_voices = max_synth_voices[dev]; - } -#endif - // (5) if ((err = pci_read_config_word(pci, PCI_DEVICE_ID, &(chip->device))) < 0) {

1 0

[alsa-devel] ALSA initialization still seems to be too late?
by Adrian Bunk 23 Mar '07

23 Mar '07

The commit below from 2005 (sic) seems to be an example that workarounds often have quite a long lifetime. Can we get this sorted out properly for 2.6.22? TIA Adrian commit c7ac6b42ffba28c350cbcd48268f46689f6eb1cc Author: Linus Torvalds <torvalds(a)g5.osdl.org> Date: Wed Dec 21 14:52:32 2005 -0800 Initialize drivers/media/video/saa7134 late When compiled-in, make sure the sound system has initialized before these drivers do. Reported by Adrian Bunk <bunk(a)stusta.de> (The right fix would be to make the sound core use "subsys_initcall()" and thus initialize before all normal drivers, but this is the quick and limited safe fix for 2.6.15). Signed-off-by: Linus Torvalds <torvalds(a)osdl.org> diff --git a/drivers/media/video/saa7134/saa7134-alsa.c b/drivers/media/video/saa7134/saa7134-alsa.c index 953d5fe..6752dd1 100644 --- a/drivers/media/video/saa7134/saa7134-alsa.c +++ b/drivers/media/video/saa7134/saa7134-alsa.c @@ -1028,7 +1028,8 @@ static void saa7134_alsa_exit(void) return; } -module_init(saa7134_alsa_init); +/* We initialize this late, to make sure the sound system is up and running */ +late_initcall(saa7134_alsa_init); module_exit(saa7134_alsa_exit); MODULE_LICENSE("GPL"); MODULE_AUTHOR("Ricardo Cerqueira"); diff --git a/drivers/media/video/saa7134/saa7134-oss.c b/drivers/media/video/saa7134/saa7134-oss.c index 513a699..c450d57 100644 --- a/drivers/media/video/saa7134/saa7134-oss.c +++ b/drivers/media/video/saa7134/saa7134-oss.c @@ -1002,7 +1002,8 @@ static void saa7134_oss_exit(void) return; } -module_init(saa7134_oss_init); +/* We initialize this late, to make sure the sound system is up and running */ +late_initcall(saa7134_oss_init); module_exit(saa7134_oss_exit); MODULE_LICENSE("GPL"); MODULE_AUTHOR("Gerd Knorr <kraxel(a)bytesex.org> [SuSE Labs]");

1 0

[alsa-devel] [5/5] 2.6.21-rc4: known regressions (v2)
by Adrian Bunk 23 Mar '07

23 Mar '07

This email lists some known regressions in Linus' tree compared to 2.6.20. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : Oops when changing DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 Submitter : CIJOML <cijoml(a)volny.cz> Status : unknown Subject : USB: iPod doesn't work References : http://lkml.org/lkml/2007/3/21/320 Submitter : Tino Keitel <tino.keitel(a)gmx.de> Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debuggged Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Status : problem is being debugged Subject : forcedeth: skb_over_panic References : http://bugzilla.kernel.org/show_bug.cgi?id=8058 Submitter : Albert Hopkins <kernel(a)marduk.letterboxes.org> Handled-By : Ayaz Abdulla <aabdulla(a)nvidia.com> Patch : http://bugzilla.kernel.org/show_bug.cgi?id=8058 Status : patch available

1 0

[alsa-devel] snd_pcm_writei() error: Write error: -32,Broken pipe
by John Utz 23 Mar '07

23 Mar '07

Hello; I am running into some trouble with writing out to the device, i get some sound and then i get the Write error. Heres the code fragment: if (-EAGAIN == (iErr = snd_pcm_writei(handle, pu8Frames, iSizPer))) continue; if (0 > iErr) { printf("Write error: %d,%s\n", iErr, snd_strerror(iErr)); Is the problem that my buffer (pointed to by pu8frames) isnt big enuf? or is this a different failure entirely. If so, what are the possibilities? tnx!

1 0

[alsa-devel] PCM -- no need to release params mem nor close handle?
by intellij 23 Mar '07

23 Mar '07

I am looking at PCM interface manual http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html It tells that there are an open and set_params functions but nothing about close(). Furthermore, the description of open() http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#g8340c7dc0… neither tells that. Ok, I have realized from the examples. However, even the examples demonstrate no params deallocation. You call the snd_pcm_hw_params_alloca(&hwparams) but never release the allocated memory. Why? Thank you.

3 2

[alsa-devel] Supported architecture: m68knommu?
by Sven Johnsson 23 Mar '07

23 Mar '07

Hello, I am interested in building ALSA for the m68knommu architecture on uClinux. Is this possible? Best Regards, Sven

2 1

Re: [alsa-devel] Réf. : Where is _snd_pcm defined?
by intellij 23 Mar '07

23 Mar '07

Thank you for your answer. However, you information is not correct. I do not have the snd_pcm structure in the three copies of pcm.h I have in alsa-lib/include, arm-linux/alsa nor arm-linux/include. Peahaps, this is because my lib is slightly obsolete. Fortunately, today I have accidentally discovered the requested _snd_pcm structure in the alsa-lib/src/pcm/pcm_local.h. Excuse me for the fuss. BTW, we have fallen out of the mailing list. Forums are efficient only when there are many participants for two reasons: neighbors can provide more information and the same questions will be prevented from asking repeatedly. Do you think this case is different? > >Hello, > >If you get the alsa-driver sources, you can have everything you need (hope :-). >Note that in my typedefs.h, I don't have any _ at the beginin, so "struct snd_pcm" is in alsa-kernel/include/pcm.h. > >Regards, >JM > > >********************************* >This message and any attachments (the "message") are confidential and intended solely for the addressees. Any unauthorised >use or dissemination is prohibited. >Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or >falsified. >If you are not the intended addressee of this message, please cancel it immediately and inform the sender. >******************************** -- Яндекс.Почта: объем почтового ящика не ограничен! http://mail.yandex.ru/monitoring/

2 1

[alsa-devel] Where is _snd_pcm defined?
by intellij 22 Mar '07

22 Mar '07

Hello, I have gone through all the source codes to wee the definition of snd_pcm_t and see that it is defined typedef struct _snd_pcm snd_pcm_t; But is has many fields inside! At least, name and fast_ops pointers should be there. I do not understand where are they hiding. Actually, I am an alsa user. However, the system occasionaly hangs on writing into the buffer (because the consumption stops). For this reason, I'm looking for the features for hacking the system. I have an idea of turning the printfs on/of in runtime by a shared flag. I could put the pointer to the flag into pcm, so that my application could modify it and the library functions would check it before printing the traces. Thank you.

1 0

[alsa-devel] (no subject)
by ODINOT Jean-Marc ETRALI 22 Mar '07

22 Mar '07

Hi Alsa gurus, I've to write an alsa driver for a custom card. I readed "writing an alsa driver" and other paper about alsa and drivers. Well I got a result, but not as good as I think it's possible with alsa. My card have 30 output slots (and 7 input slots), can be 8Khz or 16Khz (same rate for every slots), 16bits little endian by sample, can have one or more sample for each slot (same for every slots) in a frame, an hardware IT every frame (a frame is 30 slots :-). The purpose is to get, for each IT, a frame with all the datas. IT -> frame [channel1][channel2][channel3]...[channel30] channelX [sample1][sample2]...[sampleN] sampleX [16bitsle] At this time, I've something like a frame if I use channel30 with speaker-test -c30, but if I use channel1, alsa give me a buffer with many samples for channel1. And, just after oppening, I get an extra word. BTW I use .copy to get the buffer. My questions: -why the extra word (can I avoid it?)? -how set the buffer len to always have all my channels (used or not)? -is it possible to have more than a sample by channel contiguous? Think's it's all for today, but maybe your answers will give me more ideas :-) Thanx for replies (and please, forgive my awful english :-(, JM PS: sorry for my previous (and my first) post which was in html only ********************************* This message and any attachments (the "message") are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ********************************

1 0

[alsa-devel] pcm low level
by ODINOT Jean-Marc ETRALI 22 Mar '07

22 Mar '07

********************************* This message and any attachments (the "message") are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ********************************

1 0

Re: [alsa-devel] [Alsa-devel] Static build of alsa-lib
by Takashi Iwai 22 Mar '07

22 Mar '07

At Tue, 20 Mar 2007 08:13:41 -0700 (PDT), Ciaccia wrote: > > Hi Takashi, > It's clear that the concept of dynamic loading doesn't > conflict with static library, I think my original > question was not completely clear. > > I would like to port a sound application (just PCM, no > midi or other stuff) to an embedded system. The core > of this embedded system is a Cirrus EP9302 ARM > processor, which supports hardware floating points. > > ARM processors have several ABIs (EABI, OABI, ???) and > programs compiled using one ABI are not linkable > (neither at compile time, nor at run time) with other > binaries, because of the different format. This is a > problem for binary distributions (such as Debian), > since the same binary does not work on all ARM > architectures. > > For a reason I really don't understand, EP9302 > hardware floating point binaries only work when > compiled with -static (don't ask me why...), and > therefore I wanted to have a "static" (=without shared > libraries) ALSA application. If this would not be > possible I can always use OSS (which does not requires > shared objects to be loaded at run time), but I still > think it should be possible to develop ALSA > applications for architectures where dynamic loading > is not available... > > Is there a way to achieve this? Sure, what I meant is that the patch had no function to disable the libdl and libpthread explicitly but only checked. The new patch below, for example, can give you options --with-libdl and --with-pthread. For disabling libdl, pass --with-libdl=no. If this works for you, I'll apply it to the upstream. Takashi diff -r 6d0a999aef24 Makefile.am --- a/Makefile.am Fri Mar 16 15:22:27 2007 +0100 +++ b/Makefile.am Tue Mar 20 16:55:23 2007 +0100 @@ -1,4 +1,7 @@ SUBDIRS=doc include src modules -SUBDIRS=doc include src modules +SUBDIRS=doc include src +if BUILD_MODULES +SUBDIRS += modules +endif if BUILD_PCM_PLUGIN_SHM SUBDIRS += aserver endif diff -r 6d0a999aef24 configure.in --- a/configure.in Fri Mar 16 15:22:27 2007 +0100 +++ b/configure.in Tue Mar 20 16:55:00 2007 +0100 @@ -148,6 +148,44 @@ else else AC_MSG_RESULT(no) fi + +ALSA_DEPLIBS="" +if test "$softfloat" != "yes"; then + ALSA_DEPLIBS="-lm" +fi + +dnl Check for libdl +AC_MSG_CHECKING(for libdl) +AC_ARG_WITH(libdl, + [ --with-libdl Use libdl for plugins (default = yes)], + [ have_libdl="$withval" ], [ have_libdl="yes" ]) +if test "$have_libdl" = "yes"; then + AC_CHECK_LIB([dl], [dlsym], [HAVE_LIBDL="yes"]) + if test "$HAVE_LIBDL" = "yes" ; then + ALSA_DEPLIBS="$ALSA_DEPLIBS -ldl" + AC_DEFINE([HAVE_LIBDL], 1, [Have libdl]) + fi +else + AC_MSG_RESULT(no) +fi +AM_CONDITIONAL(BUILD_MODULES, test "$HAVE_LIBDL"="yes") + +dnl Check for pthread +AC_MSG_CHECKING(for pthread) +AC_ARG_WITH(pthread, + [ --with-pthread Use pthread (default = yes)], + [ have_pthread="$withval" ], [ have_pthread="yes" ]) +if test "$have_pthread" = "yes"; then + AC_CHECK_LIB([pthread], [pthread_join], [HAVE_LIBPTHREAD="yes"]) + if test "$HAVE_LIBPTHREAD" = "yes"; then + ALSA_DEPLIBS="$ALSA_DEPLIBS -lpthread" + AC_DEFINE([HAVE_LIBPTHREAD], 1, [Have libpthread]) + fi +else + AC_MSG_RESULT(no) +fi + +AC_SUBST(ALSA_DEPLIBS) dnl Check for architecture AC_MSG_CHECKING(for architecture) @@ -318,6 +356,21 @@ fi if test "$build_pcm_ioplug" = "yes"; then build_pcm_extplug="yes" +fi + +if test "$HAVE_LIBDL" != "yes"; then + build_pcm_meter="no" + build_pcm_ladspa="no" + build_pcm_pcm_ioplug="no" + build_pcm_pcm_extplug="no" +fi + +if test "$HAVE_LIBPTHREAD" != "yes"; then + build_pcm_share="no" +fi + +if test "$softfloat" != "yes"; then + build_pcm_lfloat="no" fi AM_CONDITIONAL(BUILD_PCM_PLUGIN, test x$build_pcm_plugin = xyes) diff -r 6d0a999aef24 include/local.h --- a/include/local.h Fri Mar 16 15:22:27 2007 +0100 +++ b/include/local.h Tue Mar 20 16:16:36 2007 +0100 @@ -36,6 +36,11 @@ #include "config.h" #ifdef SUPPORT_RESMGR #include <resmgr.h> +#endif +#ifdef HAVE_LIBDL +#include <dlfcn.h> +#else +#define RTLD_NOW 0 #endif #define _snd_config_iterator list_head diff -r 6d0a999aef24 src/Makefile.am --- a/src/Makefile.am Fri Mar 16 15:22:27 2007 +0100 +++ b/src/Makefile.am Tue Mar 20 16:16:36 2007 +0100 @@ -41,7 +41,7 @@ libasound_la_LIBADD += alisp/libalisp.la libasound_la_LIBADD += alisp/libalisp.la endif SUBDIRS += compat conf -libasound_la_LIBADD += compat/libcompat.la -lm -ldl -lpthread +libasound_la_LIBADD += compat/libcompat.la @ALSA_DEPLIBS@ libasound_la_LDFLAGS = -version-info $(COMPATNUM) $(VSYMS) diff -r 6d0a999aef24 src/async.c --- a/src/async.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/async.c Tue Mar 20 16:16:36 2007 +0100 @@ -151,9 +151,11 @@ int snd_async_del_handler(snd_async_hand if (!list_empty(&handler->hlist)) goto _end; switch (handler->type) { +#ifdef BUILD_PCM case SND_ASYNC_HANDLER_PCM: err = snd_pcm_async(handler->u.pcm, -1, 1); break; +#endif case SND_ASYNC_HANDLER_CTL: err = snd_ctl_async(handler->u.ctl, -1, 1); break; diff -r 6d0a999aef24 src/conf.c --- a/src/conf.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/conf.c Tue Mar 20 16:16:36 2007 +0100 @@ -415,12 +415,13 @@ beginning:</P> #include <stdarg.h> -#include <dlfcn.h> #include <limits.h> #include <sys/stat.h> -#include <pthread.h> #include <locale.h> #include "local.h" +#ifdef HAVE_LIBPTHREAD +#include <pthread.h> +#endif #ifndef DOC_HIDDEN @@ -3080,7 +3081,9 @@ int snd_config_update_r(snd_config_t **_ return 1; } +#ifdef HAVE_LIBPTHREAD static pthread_mutex_t snd_config_update_mutex = PTHREAD_MUTEX_INITIALIZER; +#endif /** * \brief Updates #snd_config by rereading the global configuration files (if needed). @@ -3099,9 +3102,13 @@ int snd_config_update(void) { int err; +#ifdef HAVE_LIBPTHREAD pthread_mutex_lock(&snd_config_update_mutex); +#endif err = snd_config_update_r(&snd_config, &snd_config_global_update, NULL); +#ifdef HAVE_LIBPTHREAD pthread_mutex_unlock(&snd_config_update_mutex); +#endif return err; } @@ -3128,15 +3135,18 @@ int snd_config_update_free(snd_config_up */ int snd_config_update_free_global(void) { +#ifdef HAVE_LIBPTHREAD pthread_mutex_lock(&snd_config_update_mutex); +#endif if (snd_config) snd_config_delete(snd_config); snd_config = NULL; if (snd_config_global_update) snd_config_update_free(snd_config_global_update); snd_config_global_update = NULL; +#ifdef HAVE_LIBPTHREAD pthread_mutex_unlock(&snd_config_update_mutex); - +#endif /* FIXME: better to place this in another place... */ snd_dlobj_cache_cleanup(); diff -r 6d0a999aef24 src/confmisc.c --- a/src/confmisc.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/confmisc.c Tue Mar 20 16:16:36 2007 +0100 @@ -946,6 +946,8 @@ SND_DLSYM_BUILD_VERSION(snd_func_card_na SND_DLSYM_BUILD_VERSION(snd_func_card_name, SND_CONFIG_DLSYM_VERSION_EVALUATE); #endif +#ifdef BUILD_PCM + /** * \brief Returns the pcm identification of a device. * \param dst The function puts the handle to the result configuration node @@ -1198,6 +1200,8 @@ int snd_func_private_pcm_subdevice(snd_c #ifndef DOC_HIDDEN SND_DLSYM_BUILD_VERSION(snd_func_private_pcm_subdevice, SND_CONFIG_DLSYM_VERSION_EVALUATE); #endif + +#endif /* BUILD_PCM */ /** * \brief Copies the specified configuration node. diff -r 6d0a999aef24 src/control/control.c --- a/src/control/control.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/control/control.c Tue Mar 20 16:16:36 2007 +0100 @@ -47,7 +47,6 @@ and IEC958 structure. #include <string.h> #include <fcntl.h> #include <signal.h> -#include <dlfcn.h> #include <sys/poll.h> #include "control_local.h" diff -r 6d0a999aef24 src/control/hcontrol.c --- a/src/control/hcontrol.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/control/hcontrol.c Tue Mar 20 16:16:36 2007 +0100 @@ -48,11 +48,13 @@ to reduce overhead accessing the real co #include <string.h> #include <fcntl.h> #include <sys/ioctl.h> -#include <pthread.h> #ifndef DOC_HIDDEN #define __USE_GNU #endif #include "control_local.h" +#ifdef HAVE_LIBPTHREAD +#include <pthread.h> +#endif #ifndef DOC_HIDDEN #define NOT_FOUND 1000000000 @@ -420,17 +422,22 @@ static void snd_hctl_sort(snd_hctl_t *hc static void snd_hctl_sort(snd_hctl_t *hctl) { unsigned int k; +#ifdef HAVE_LIBPTHREAD static pthread_mutex_t sync_lock = PTHREAD_MUTEX_INITIALIZER; +#endif assert(hctl); assert(hctl->compare); INIT_LIST_HEAD(&hctl->elems); +#ifdef HAVE_LIBPTHREAD pthread_mutex_lock(&sync_lock); +#endif compare_hctl = hctl; qsort(hctl->pelems, hctl->count, sizeof(*hctl->pelems), hctl_compare); +#ifdef HAVE_LIBPTHREAD pthread_mutex_unlock(&sync_lock); - +#endif for (k = 0; k < hctl->count; k++) list_add_tail(&hctl->pelems[k]->list, &hctl->elems); } diff -r 6d0a999aef24 src/dlmisc.c --- a/src/dlmisc.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/dlmisc.c Tue Mar 20 16:51:13 2007 +0100 @@ -28,7 +28,6 @@ */ #define _GNU_SOURCE -#include <dlfcn.h> #include "list.h" #include "local.h" @@ -53,13 +52,19 @@ void *snd_dlopen(const char *name, int m if (name == NULL) return &snd_dlsym_start; #else +#ifdef HAVE_LIBDL if (name == NULL) { Dl_info dlinfo; if (dladdr(snd_dlopen, &dlinfo) > 0) name = dlinfo.dli_fname; } #endif +#endif +#ifdef HAVE_LIBDL return dlopen(name, mode); +#else + return NULL; +#endif } /** @@ -76,7 +81,11 @@ int snd_dlclose(void *handle) if (handle == &snd_dlsym_start) return 0; #endif +#ifdef HAVE_LIBDL return dlclose(handle); +#else + return 0; +#endif } /** @@ -91,6 +100,7 @@ int snd_dlclose(void *handle) */ static int snd_dlsym_verify(void *handle, const char *name, const char *version) { +#ifdef HAVE_LIBDL int res; char *vname; @@ -107,6 +117,9 @@ static int snd_dlsym_verify(void *handle if (res < 0) SNDERR("unable to verify version for symbol %s", name); return res; +#else + return 0; +#endif } /** @@ -139,10 +152,16 @@ void *snd_dlsym(void *handle, const char return NULL; } #endif - err = snd_dlsym_verify(handle, name, version); - if (err < 0) - return NULL; +#ifdef HAVE_LIBDL + if (version) { + err = snd_dlsym_verify(handle, name, version); + if (err < 0) + return NULL; + } return dlsym(handle, name); +#else + return NULL; +#endif } /* diff -r 6d0a999aef24 src/hwdep/hwdep.c --- a/src/hwdep/hwdep.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/hwdep/hwdep.c Tue Mar 20 16:16:36 2007 +0100 @@ -33,7 +33,6 @@ #include <unistd.h> #include <string.h> #include <fcntl.h> -#include <dlfcn.h> #include <sys/ioctl.h> #include "hwdep_local.h" diff -r 6d0a999aef24 src/mixer/simple_abst.c --- a/src/mixer/simple_abst.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/mixer/simple_abst.c Tue Mar 20 16:52:35 2007 +0100 @@ -82,14 +82,14 @@ static int try_open(snd_mixer_class_t *c free(xlib); return -ENXIO; } - event_func = dlsym(h, "alsa_mixer_simple_event"); + event_func = snd_dlsym(h, "alsa_mixer_simple_event", NULL); if (event_func == NULL) { SNDERR("Symbol 'alsa_mixer_simple_event' was not found in '%s'", xlib); snd_dlclose(h); free(xlib); return -ENXIO; } - init_func = dlsym(h, "alsa_mixer_simple_init"); + init_func = snd_dlsym(h, "alsa_mixer_simple_init", NULL); if (init_func == NULL) { SNDERR("Symbol 'alsa_mixer_simple_init' was not found in '%s'", xlib); snd_dlclose(h); diff -r 6d0a999aef24 src/pcm/pcm.c --- a/src/pcm/pcm.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/pcm/pcm.c Tue Mar 20 16:16:36 2007 +0100 @@ -634,7 +634,6 @@ playback devices. #include <malloc.h> #include <stdarg.h> #include <signal.h> -#include <dlfcn.h> #include <sys/poll.h> #include <sys/shm.h> #include <sys/mman.h> diff -r 6d0a999aef24 src/pcm/pcm_hooks.c --- a/src/pcm/pcm_hooks.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/pcm/pcm_hooks.c Tue Mar 20 16:16:36 2007 +0100 @@ -27,7 +27,6 @@ * */ -#include <dlfcn.h> #include "pcm_local.h" #include "pcm_generic.h" diff -r 6d0a999aef24 src/pcm/pcm_ladspa.c --- a/src/pcm/pcm_ladspa.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/pcm/pcm_ladspa.c Tue Mar 20 16:16:36 2007 +0100 @@ -33,7 +33,6 @@ */ #include <dirent.h> -#include <dlfcn.h> #include <locale.h> #include <math.h> #include "pcm_local.h" diff -r 6d0a999aef24 src/pcm/pcm_rate.c --- a/src/pcm/pcm_rate.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/pcm/pcm_rate.c Tue Mar 20 16:51:41 2007 +0100 @@ -29,7 +29,6 @@ */ #include <inttypes.h> #include <byteswap.h> -#include <dlfcn.h> #include "pcm_local.h" #include "pcm_plugin.h" #include "pcm_rate.h" @@ -1326,7 +1325,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, free(rate); return -ENOENT; } - open_func = dlsym(h, open_name); + open_func = snd_dlsym(h, open_name, NULL); if (! open_func) { SNDERR("Cannot find function %s", open_name); snd_dlclose(h); diff -r 6d0a999aef24 src/rawmidi/rawmidi.c --- a/src/rawmidi/rawmidi.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/rawmidi/rawmidi.c Tue Mar 20 16:16:36 2007 +0100 @@ -139,7 +139,6 @@ This example shows open and read/write r #include <stdarg.h> #include <unistd.h> #include <string.h> -#include <dlfcn.h> #include "rawmidi_local.h" /** diff -r 6d0a999aef24 src/seq/seq.c --- a/src/seq/seq.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/seq/seq.c Tue Mar 20 16:16:36 2007 +0100 @@ -777,7 +777,6 @@ void event_filter(snd_seq_t *seq, snd_se */ -#include <dlfcn.h> #include <sys/poll.h> #include "seq_local.h" diff -r 6d0a999aef24 src/timer/timer.c --- a/src/timer/timer.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/timer/timer.c Tue Mar 20 16:16:36 2007 +0100 @@ -72,7 +72,6 @@ This example shows opening a timer devic #include <unistd.h> #include <string.h> #include <fcntl.h> -#include <dlfcn.h> #include <signal.h> #include <sys/ioctl.h> #include "timer_local.h" diff -r 6d0a999aef24 src/timer/timer_query.c --- a/src/timer/timer_query.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/timer/timer_query.c Tue Mar 20 16:16:36 2007 +0100 @@ -31,7 +31,6 @@ #include <unistd.h> #include <string.h> #include <fcntl.h> -#include <dlfcn.h> #include <sys/ioctl.h> #include "timer_local.h" diff -r 6d0a999aef24 utils/alsa.pc.in --- a/utils/alsa.pc.in Fri Mar 16 15:22:27 2007 +0100 +++ b/utils/alsa.pc.in Tue Mar 20 16:16:36 2007 +0100 @@ -8,7 +8,7 @@ Version: @VERSION@ Version: @VERSION@ Requires: Libs: -L${libdir} -lasound -Libs.private: -lm -ldl -lpthread +Libs.private: @ALSA_DEPLIBS@ # -I${includedir}/alsa below is just for backward compatibility # (it was set so mistakely in the older version) Cflags: -I${includedir} -I${includedir}/alsa

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[alsa-devel] During ADPCM playback, when is the reference byte set?
by Rask Ingemann Lambertsen 21 Mar '07

21 Mar '07

Hi. When an app is playing ADPCM samples, how does the driver know how to find the reference byte, other than assuming it is the first byte of sample data? Could there be more than one? Etc. -- Rask Ingemann Lambertsen

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[alsa-devel] s/sf.net/alsa-devel.org/
by Rene Herman 21 Mar '07

21 Mar '07

Hi Takashi. This replaces all occurences of alsa-devel(a)lists.s[ource]f[orge].net that a simple recursive grep found in the current HG ALSA repos by alsa-devel(a)alsa-project.org. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Rene. diff -urN alsa.orig/alsa-driver/pci/bt87x.patch alsa/alsa-driver/pci/bt87x.patch --- alsa.orig/alsa-driver/pci/bt87x.patch 2006-06-29 23:22:39.000000000 +0200 +++ alsa/alsa-driver/pci/bt87x.patch 2007-03-20 17:52:59.000000000 +0100 @@ -39,7 +39,7 @@ + pci->device, subsystem_vendor, subsystem_device); snd_printk(KERN_DEBUG "please mail id, board name, and, " "if it works, the correct digital_rate option to " - "<alsa-devel(a)lists.sf.net>\n"); + "<alsa-devel(a)alsa-project.org>\n"); @@ -935,3 +948,5 @@ module_init(alsa_card_bt87x_init) diff -urN alsa.orig/alsa-kernel/Documentation/ALSA-Configuration.txt alsa/alsa-kernel/Documentation/ALSA-Configuration.txt --- alsa.orig/alsa-kernel/Documentation/ALSA-Configuration.txt 2007-02-26 17:00:05.000000000 +0100 +++ alsa/alsa-kernel/Documentation/ALSA-Configuration.txt 2007-03-20 17:54:24.000000000 +0100 @@ -2049,4 +2049,4 @@ https://bugtrack.alsa-project.org/bugs/ ALSA Developers ML - mailto:alsa-devel@lists.sourceforge.net + mailto:alsa-devel@alsa-project.org diff -urN alsa.orig/alsa-kernel/Documentation/Bt87x.txt alsa/alsa-kernel/Documentation/Bt87x.txt --- alsa.orig/alsa-kernel/Documentation/Bt87x.txt 2006-06-29 23:25:22.000000000 +0200 +++ alsa/alsa-kernel/Documentation/Bt87x.txt 2007-03-20 17:55:06.000000000 +0100 @@ -36,8 +36,8 @@ other values than the default 32000 (often it's 44100 or 64000). If you have an unknown card, please mail the ID and board name to -<alsa-devel(a)lists.sf.net>, regardless of whether audio capture works or -not, so that future versions of this driver know about your card. +<alsa-devel(a)alsa-project.org>, regardless of whether audio capture works +or not, so that future versions of this driver know about your card. Audio modes diff -urN alsa.orig/alsa-kernel/pci/bt87x.c alsa/alsa-kernel/pci/bt87x.c --- alsa.orig/alsa-kernel/pci/bt87x.c 2007-03-06 11:30:32.000000000 +0100 +++ alsa/alsa-kernel/pci/bt87x.c 2007-03-20 17:56:40.000000000 +0100 @@ -835,7 +835,7 @@ pci->device, pci->subsystem_vendor, pci->subsystem_device); snd_printk(KERN_DEBUG "please mail id, board name, and, " "if it works, the correct digital_rate option to " - "<alsa-devel(a)lists.sf.net>\n"); + "<alsa-devel(a)alsa-project.org>\n"); return 32000; /* default rate */ } diff -urN alsa.orig/alsa-kernel/pci/cs46xx/cs46xx_lib.c alsa/alsa-kernel/pci/cs46xx/cs46xx_lib.c --- alsa.orig/alsa-kernel/pci/cs46xx/cs46xx_lib.c 2006-11-21 18:00:06.000000000 +0100 +++ alsa/alsa-kernel/pci/cs46xx/cs46xx_lib.c 2007-03-20 17:55:54.000000000 +0100 @@ -3107,7 +3107,7 @@ snd_printk(KERN_ERR "ERROR: snd-cs46xx: never read ISV3 & ISV4 from AC'97\n"); snd_printk(KERN_ERR " Try reloading the ALSA driver, if you find something\n"); snd_printk(KERN_ERR " broken or not working on your soundcard upon\n"); - snd_printk(KERN_ERR " this message please report to alsa-devel(a)lists.sourceforge.net\n"); + snd_printk(KERN_ERR " this message please report to alsa-devel(a)alsa-project.org\n"); return -EIO; #endif diff -urN alsa.orig/alsa-lib/src/conf/cards/USB-Audio.conf alsa/alsa-lib/src/conf/cards/USB-Audio.conf --- alsa.orig/alsa-lib/src/conf/cards/USB-Audio.conf 2007-02-13 10:00:05.000000000 +0100 +++ alsa/alsa-lib/src/conf/cards/USB-Audio.conf 2007-03-20 17:57:20.000000000 +0100 @@ -9,7 +9,7 @@ # cards.USB-Audio.pcm.use_dmix."NoiseBlaster 3000" no # # If your device requires such a definition to work correctly, please report it -# to <alsa-devel(a)lists.sourceforge.net>. +# to <alsa-devel(a)alsa-project.org>. # If a device has sample formats not supported by dmix, dmix can be disabled

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[alsa-devel] Problem in dummy driver
by Himanshu Chauhan 21 Mar '07

21 Mar '07

Dear all, I am working on the dummy driver provided with the ALSA. I took it from the linux kernel 2.6.20.1 I build the module and load it. The XMMS seems to play ok (doesn't hang and all) but none of the recording application seem to record from the driver. Can the driver in current state work as the loop back cable between applications? Thanks in advance... Warm Regards --Himanshu

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[alsa-devel] RESUBSCRIBE - New alsa-devel mailing list
by Jaroslav Kysela 21 Mar '07

21 Mar '07

Hello everyone, because never ending story with SF services, I quickly setup new mailman alsa-devel list on new ALSA server (we'll get working list archive as bonus). Because I don't want to move automatically all subscribers, please, subscribe yourself to new list: http://mailman.alsa-project.org/mailman/listinfo/alsa-devel New list is available as alsa-devel(a)alsa-project.org while old on SF is available as alsa-devel(a)lists.sourceforge.net . The archive for new alsa-devel mailing list is here: http://mailman.alsa-project.org/pipermail/alsa-devel/ The posts from non-members are moderated now! Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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Re: [alsa-devel] Development Question
by M. Gagnon 21 Mar '07

21 Mar '07

> > >What is the output of the program? > > > > pending events size: 28 > > pending events size: 56 > > pending events size: 84 > > ... > > BEFORE DRAIN: pending events size: 476 > > AFTER DRAIN: pending events size: 0 > >Looks OK. In theory, all events should be sent somewhere. > >Please run aseqdump and send the events to its port. What does it >output? > % aseqdump Waiting for data at port 128:0. Press Ctrl+C to end. Source_ Event_________________ Ch _Data__ 0:1 Port subscribed 129:0 -> 128:0 129:0 Note on 2 60 127 0:1 Port unsubscribed 129:0 -> 128:0 Seems like only one event gets sent - just like what i hear. >Run "cat /proc/asound/seq/queues" and >"cat /proc/asound/seq/timer". > % cat /proc/asound/seq/queues queue 0: [Queue-0] owned by client : 128 lock status : Locked queued time events : 0 queued tick events : 0 timer state : Running timer PPQ : 96 current tempo : 500000 current BPM : 120 current time : 1.536000000 s current tick : 294 queue 1: [Queue-1] owned by client : 129 lock status : Locked queued time events : 0 queued tick events : 0 timer state : Running timer PPQ : 48 current tempo : 1000000 current BPM : 60 current time : 1.488000000 s current tick : 71 % cat /proc/asound/seq/timer Timer for queue 0 : system timer Period time : 0.004000000 Skew : 65536 / 65536 Timer for queue 1 : system timer Period time : 0.004000000 Skew : 65536 / 65536 thanks for your help just as reminder (so the output can be interpreted) this is my latest code /* all: g++ midi_receiver.cpp -o AlsaNoteTest -lasound */ #include <stdio.h> #include <alsa/asoundlib.h> #include "midi_receiver.h" #include <vector> #include <iostream> #include <unistd.h> #include <pthread.h> pthread_t thread1; char *message1 = "Thread 1"; int iret1; void *play_func( void *ptr ) { AlsaNotePlayer::play(); } int main() { AlsaNotePlayer::findDevices(); AlsaNotePlayer::openDevice(128, 0); AlsaNotePlayer::setTempo(60, 48); int time=0; int duration = 50; AlsaNotePlayer::noteOn_seq(1, 60, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 60, 0, time); AlsaNotePlayer::noteOn_seq(1, 62, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 62, 0, time); AlsaNotePlayer::noteOn_seq(1, 60, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 60, 0, time); AlsaNotePlayer::noteOn_seq(1, 59, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 59, 0, time); // ---------- AlsaNotePlayer::noteOn_seq(1, 63, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 63, 0, time); AlsaNotePlayer::noteOn_seq(1, 65, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 65, 0, time); AlsaNotePlayer::noteOn_seq(1, 63, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 63, 0, time); AlsaNotePlayer::noteOn_seq(1, 61, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 61, 0, time); iret1 = pthread_create( &thread1, NULL, play_func, (void*) message1); pthread_join( thread1, NULL); while(1) { usleep(10000); } } namespace AlsaNotePlayer { class MidiDevice { public: int client, port; char* name; MidiDevice(int client_arg, int port_arg, const char* name_arg) { client = client_arg; port = port_arg; name = (char*)name_arg; } }; std::vector<MidiDevice> devices; struct MidiContext { snd_seq_t* sequencer; int queue; snd_seq_addr_t address; int port; snd_seq_port_subscribe_t *subs; }; struct MidiContext MidiContext; snd_seq_addr_t address; void findDevices() { snd_seq_client_info_t* clientInfo; if (snd_seq_open(&MidiContext.sequencer, "default", SND_SEQ_OPEN_OUTPUT, 0) < 0) { return; } snd_seq_client_info_alloca(&clientInfo); snd_seq_client_info_set_client(clientInfo, -1); // iterate through clients while (snd_seq_query_next_client(MidiContext.sequencer, clientInfo) >= 0) { snd_seq_port_info_t* pinfo; snd_seq_port_info_alloca(&pinfo); snd_seq_port_info_set_client(pinfo, snd_seq_client_info_get_client(clientInfo)); snd_seq_port_info_set_port(pinfo, -1); // and now through ports while (snd_seq_query_next_port(MidiContext.sequencer, pinfo) >= 0) { unsigned int capability = snd_seq_port_info_get_capability(pinfo); if ((capability & SND_SEQ_PORT_CAP_SUBS_WRITE) == 0) { continue; } int client = (snd_seq_port_info_get_addr(pinfo))->client; int port = (snd_seq_port_info_get_addr(pinfo))->port; std::cout << client << ":" << port << " --> " << snd_seq_port_info_get_name(pinfo) << std::endl; devices.push_back( MidiDevice(client, port, snd_seq_port_info_get_name(pinfo)) ); } } if (snd_seq_set_client_name(MidiContext.sequencer, "Aria") < 0) { return; } //MidiContext.address.port = snd_seq_create_simple_port(MidiContext.sequencer, "Aria Port 0", SND_SEQ_PORT_CAP_SUBS_WRITE, SND_SEQ_PORT_TYPE_APPLICATION); MidiContext.address.port = snd_seq_create_simple_port(MidiContext.sequencer, "Aria Port 0", SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_WRITE | SND_SEQ_PORT_CAP_READ, SND_SEQ_PORT_TYPE_APPLICATION); //SND_SEQ_PORT_TYPE_MIDI_GENERIC); MidiContext.address.client = snd_seq_client_id (MidiContext.sequencer); MidiContext.queue = snd_seq_alloc_queue (MidiContext.sequencer); snd_seq_set_client_pool_output (MidiContext.sequencer, 1024); } void setTempo(int tempo_arg, int resolution) { // argument is not yet considered snd_seq_queue_tempo_t *tempo; snd_seq_queue_tempo_alloca(&tempo); // snd_seq_queue_tempo_set_tempo(tempo, 1000000); // 60 BPM // snd_seq_queue_tempo_set_ppq(tempo, 48); // 48 PPQ snd_seq_queue_tempo_set_ppq(tempo, resolution); snd_seq_queue_tempo_set_tempo(tempo, 60*1000000/tempo_arg); snd_seq_set_queue_tempo(MidiContext.sequencer, MidiContext.queue, tempo); } void openDevice(int client, int port) { address.client = client; address.port = port; snd_seq_port_subscribe_alloca(&MidiContext.subs); snd_seq_port_subscribe_set_sender(MidiContext.subs, &MidiContext.address); snd_seq_port_subscribe_set_dest(MidiContext.subs, &address); snd_seq_subscribe_port(MidiContext.sequencer, MidiContext.subs); } void closeDevice() { snd_seq_port_subscribe_alloca(&MidiContext.subs); snd_seq_port_subscribe_set_sender(MidiContext.subs, &MidiContext.address); snd_seq_port_subscribe_set_dest(MidiContext.subs, &address); if (snd_seq_unsubscribe_port(MidiContext.sequencer, MidiContext.subs) < 0) { return; } //snd_seq_port_subscribe_free(MidiContext.subs); } //************************************************************************************// //************************************MIDI EVENTS*************************************// //************************************************************************************// void noteOn_seq(int channel, int note, int velocity, int tick) { snd_seq_event_t event; snd_seq_ev_clear(&event); event.queue = MidiContext.queue; event.source = MidiContext.address; snd_seq_ev_set_subs(&event); snd_seq_ev_set_source(&event, MidiContext.port); snd_seq_ev_set_noteon(&event, channel, note, velocity); snd_seq_ev_schedule_tick(&event, MidiContext.queue, 0, tick ); snd_seq_event_output(MidiContext.sequencer, &event); std::cout << "pending events size: " << snd_seq_event_output_pending(MidiContext.sequencer) << std::endl; } void noteOff_seq(int channel, int note, int velocity, int tick) { snd_seq_event_t event; snd_seq_ev_clear(&event); event.queue = MidiContext.queue; event.source = MidiContext.address; snd_seq_ev_set_subs(&event); snd_seq_ev_set_noteoff(&event, channel, note, velocity); snd_seq_ev_schedule_tick(&event, MidiContext.queue, 0 /*absolute*/, tick ); snd_seq_event_output(MidiContext.sequencer, &event); std::cout << "pending events size: " << snd_seq_event_output_pending(MidiContext.sequencer) << std::endl; } void play() { snd_seq_start_queue(MidiContext.sequencer, MidiContext.queue, 0); std::cout << "BEFORE DRAIN: pending events size: " << snd_seq_event_output_pending(MidiContext.sequencer) << std::endl; int output = snd_seq_drain_output(MidiContext.sequencer); std::cout << "AFTER DRAIN: pending events size: " << snd_seq_event_output_pending(MidiContext.sequencer) << std::endl; if(output == 0) std::cout << "all events drained" << std::endl; if(output > 0) std::cout << output << "bytes remaining" << std::endl; if(output < 0) { fprintf (stderr, "Error: (%s)\n", snd_strerror (output)); //std::cout << "error" << std::endl; } } } _________________________________________________________________ Ne perdez pas de temps dans les files dattente magasinez en ligne. http://magasiner.sympatico.msn.ca

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Re: [alsa-devel] [Alsa-devel] [RFC][RFT] Adding support for Jazz16 based sound cards
by Rene Herman 20 Mar '07

20 Mar '07

On 03/19/2007 11:46 PM, Rask Ingemann Lambertsen wrote: >> This setup also shares that same problem with the BIOS generally >> not having reserved/routed the resources -- on a general PCI/ISA >> system the user needs to go into the BIOS setup and reserve the >> resources the card is (or will be, in this case) set to. So now the >> user really needs to know what resources are going to be assigned, >> destroying the plug and play idea... > > Not really. A standards compliant PnPBIOS set up with "PnP aware OS = > YES" will use only enough resources to boot the system, which should > leave enough irqs and dmas for the Linux PnP layer to do its work. There's also the "routed" bit though. I actually ran into this recently when I was playing with snd-sgalaxy (which is why I'm harping on it). An sgalaxy is also a (partly) software configurable card and when not given explicit parameters the driver just searches for an unrequested IRQ and DMA and then programs the card for it. Over here it grabbed IRQ 10 but then didn't work anyway, and only started working when I set IRQ 10 to "Legacy ISA" in the BIOS setup. It seems the IRQ simply was not available to ISA without having done that (no, no PCI cards were using IRQ 10 or anything). I must say though that my comment above was partly based on a somewhat foggy understanding of how things get assigned. Yes, _if_ this "routing" problem I seemed to be experiencing is not a serious problem in practice (Adam? Are you able to comment?) then the fact that the PnP layer will be handing out the resources taking into account the other (known) cards is indeed a practical advantage of using that layer. > I think there are three good reasons to use the PnP layer: > > 1) The PnP layer informs the PCI layer of irqs to try to avoid - please > see pnp_register_irq_resource() in drivers/pnp/resource.c. How does that help? On my systems (one AMI, all others Award BIOses) all PCI cards get assigned their IRQ before the OS ever boots. That "PnP OS: Yes/No" switch only effects whether or not the BIOS will fully configure all ISA-PnP cards. > 2) The PnP layer knows which resources are used by active devices, even > if they have not been allocated using e.g. request_region(). This > happens in practice. For example: Yes, this is an advantage. It does miss all the resources used by old ISA cards for which no (normal or pnp protocol as in this one) drivers have been loaded but I guess that's not much of a difference with a regular ISA driver... > 3) Knowing which resources a card can utilize, the PnP layer can try to > stay away from those resources when configuring other cards. The PnP layer > doesn't do that yet in a stock kernel, but I've patched it to do so. I'll > try to get those patches into the kernel. I guess that could be an advantage. When I was looking at pnp/isa integration a while ago myself I was looking at it from the angle of generic old ISA (ie, hardwired non settable resources) and concluded that for these there wasn't any point. Having looked a little closer at the resource assigment being done in the PnP layer now, I now feel there's some point, but still little for those. But with software settable devices such as this Jazz16, there ofcourse is rather more point. I believe the routing issue I refferred to above is the most important thing to have answered. I also still don't really like "splitting the one driver into the two pieces" like that but well, we don't actually have all that many software settable ISA devices and I guess it's not _too_ bad to have a few of these "front-ends" run as a protocol driver at boot as long as it stays "a few". Hey, I already said I like the concept... By the way; in the "straightforward version" I posted yesterday I neglected to reserve the SB_JAZZ16_CONFIG_PORT and the SB range before poking at it during setup. Attached version corrects that. Updated patch including your SB changes still at http://members.home.nl/rene.herman/jazz16.diff Looks more involved now, but posted mainly for completeness; correcting yesterday's post. Has been tested and found working again. Rene. diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 9fef210..b60690a 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1120,6 +1120,13 @@ Prior to version 0.9.0rc4 options had a This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ----------------- + + Module for sound cards based Media Vision Jazz16 chip + + This module supports multiple cards. + Module snd-korg1212 ------------------- diff --git a/include/sound/sb.h b/include/sound/sb.h index 2dd5c8e..9594f89 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -38,6 +38,7 @@ enum sb_hw_type { SB_HW_ALS100, /* Avance Logic ALS100 chip */ SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ + SB_HW_JAZZ16, /* Media Vision Jazz16 chip */ }; #define SB_OPEN_PCM 0x01 @@ -140,8 +141,11 @@ #define SB_DSP_LO_OUTPUT_AUTO 0x1c #define SB_DSP_LO_INPUT_AUTO 0x2c #define SB_DSP_HI_OUTPUT_AUTO 0x90 #define SB_DSP_HI_INPUT_AUTO 0x98 +#define SB_DSP_ULAW_OUTPUT_AUTO 0x7f +#define SB_DSP_ULAW_INPUT_AUTO 0x8f #define SB_DSP_IMMED_INT 0xf2 #define SB_DSP_GET_VERSION 0xe1 +#define SB_DSP_GET_JAZZ_VERSION 0xfa #define SB_DSP_SPEAKER_ON 0xd1 #define SB_DSP_SPEAKER_OFF 0xd3 #define SB_DSP_DMA8_OFF 0xd0 @@ -265,6 +269,14 @@ #define SB_DMASETUP_DMA5 0x20 #define SB_DMASETUP_DMA6 0x40 #define SB_DMASETUP_DMA7 0x80 +/* Jazz16 configuration. */ +#define SB_JAZZ16_CONFIG_PORT 0x201 +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 + +/* Check the output of SB_DSP_GET_JAZZ_VERSION. */ +#define SB_VERSION_IS_JAZZ16(x) ((x) == 0x12) + /* * */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index a629b52..450255f 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -258,6 +258,14 @@ config SND_SW60XG To compile this driver as a module, choose M here: the module will be called snd-sw60xg. +config SND_JAZZ16 + tristate "Media Vision Jazz16" + depends on SND + select SND_OPL3_LIB + select SND_PCM + help + Say 'Y' or 'M' to include support for Media Vision Jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" depends on SND diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index fd9d9c5..745e1c1 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # # this function returns: @@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SB16) += snd-sb16.o snd obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o snd-sb16-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ES968) += snd-es968.o snd-sb8-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ALS4000) += snd-sb-common.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o snd-sb8-dsp.o snd-sb-common.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c new file mode 100644 index 0000000..988a5e9 --- /dev/null +++ b/sound/isa/sb/jazz16.c @@ -0,0 +1,334 @@ +/* Driver for Media Vision Jazz16 based boards. + * + * Copyright (c) 2007 by Rask Ingemann Lambertsen + * + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * The Jazz16 is an SB Pro compatible chip with a 16-bit mode and higher + * playback and capture rates added to it. It is not SB 16 compatible. + * The chip has an MPU-401 interface which is handled by the MPU-401 driver. + * + * The IBM PPS Model 6015 has a Jazz16 chip on board. Please tell me if it + * works with this driver or not. + */ + +#include <sound/driver.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/isa.h> +#include <linux/ioport.h> +#include <asm/dma.h> +#include <asm/io.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> + +#define CRD_NAME "Media Vision Jazz16" +#define DEV_NAME "jazz16" + +MODULE_DESCRIPTION(CRD_NAME); +MODULE_AUTHOR("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_LICENSE("GPL"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ +static long port = SNDRV_DEFAULT_PORT1; /* 0x220,0x240,0x260 */ +static long mpu_port = SNDRV_DEFAULT_PORT1; /* 0x310,0x320,0x330 */ +static int irq = SNDRV_DEFAULT_IRQ1; /* 5,9,3,7,10,15 */ +static int mpu_irq = SNDRV_DEFAULT_IRQ1; /* 5,9,3,7,10,15 */ +static int dma8 = SNDRV_DEFAULT_DMA1; /* 1,3 */ +static int dma16 = SNDRV_DEFAULT_DMA1; /* 5,7 */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); +module_param(enable, bool, 0444); +MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); +module_param(port, long, 0444); +MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); +module_param(mpu_port, long, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); +module_param(irq, int, 0444); +MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); +module_param(mpu_irq, int, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); +module_param(dma8, int, 0444); +MODULE_PARM_DESC(dma8, "8-bit DMA # for " CRD_NAME " driver."); +module_param(dma16, int, 0444); +MODULE_PARM_DESC(dma16, "16-bit DMA # for " CRD_NAME " driver."); + +static int __devinit snd_jazz16_match(struct device *dev, unsigned int n) +{ + if (!enable) + return 0; + + if (port == SNDRV_AUTO_PORT) { + snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + return 0; + } + if (irq == SNDRV_AUTO_IRQ) { + snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id); + return 0; + } + if (dma8 == SNDRV_AUTO_DMA) { + snd_printk(KERN_ERR "%s: please specify dma8\n", dev->bus_id); + return 0; + } + if (dma16 == SNDRV_AUTO_DMA) { + snd_printk(KERN_ERR "%s: please specify dma16\n", dev->bus_id); + return 0; + } + return 1; +} + +#define BUSY_LOOPS 100000 + +static int __devinit jazz16_sbdsp_command(u8 value) +{ + int i; + + for (i = BUSY_LOOPS; i; i--) + if ((inb(SBP1(port, STATUS)) & 0x80) == 0) { + outb(value, SBP1(port, COMMAND)); + return 0; + } + return -EIO; +} + +static int __devinit jazz16_sbdsp_get_byte(void) +{ + int i; + + for (i = BUSY_LOOPS; i; i--) + if (inb(SBP1(port, DATA_AVAIL)) & 0x80) + return inb(SBP1(port, READ)); + + return -EIO; +} + +static int __devinit jazz16_sbdsp_reset(void) +{ + int error; + + outb(1, SBP1(port, RESET)); + udelay(10); + outb(0, SBP1(port, RESET)); + udelay(30); + + error = jazz16_sbdsp_get_byte(); + if (error < 0) + return error; + + return error == 0xaa ? 0 : -ENODEV; +} + +static void jazz16_set_ports(u8 value) +{ + outb(SB_JAZZ16_WAKEUP, SB_JAZZ16_CONFIG_PORT); + outb(SB_JAZZ16_SET_PORTS, SB_JAZZ16_CONFIG_PORT); + outb(value, SB_JAZZ16_CONFIG_PORT); +} + +#define JAZZ16_SET_DMAINTR 0xfb + +static int __devinit jazz16_enable(struct device *dev) +{ + static u8 irq_config[] = {0,0,2,3,0,1,0,4,0,2,5,0,0,0,0,6}; + static u8 dma_config[] = {0,1,0,2,0,3,0,4}; + u8 config = 0; + + int error; + + if (port != 0x220 && port != 0x240 && port != 0x260) + port = 0; + + if (mpu_port != 0x310 && mpu_port != 0x320 && mpu_port != 0x330) + mpu_port = 0; + + config = (port & 0x70) | ((mpu_port >> 4) & 0x07); + + snd_printd("%s: setting SB port = %#lx, MPU port = %#lx.\n", + dev->bus_id, port, mpu_port); + + jazz16_set_ports(config); + + snd_printd("%s: setting SB irq %d, dma %d&%d, MPU irq %d.\n", + dev->bus_id, irq, dma8, dma16, mpu_irq); + + if (!request_region(port, 16, "Jazz16")) + return -EBUSY; + + error = jazz16_sbdsp_command(JAZZ16_SET_DMAINTR); + if (error < 0) + goto out; + + config = dma_config[dma8] | dma_config[dma16] << 4; + + error = jazz16_sbdsp_command(config); + if (error < 0) + goto out; + + config = irq_config[irq] | irq_config[mpu_irq] << 4; + + error = jazz16_sbdsp_command(config); + if (error < 0) + goto out; + + error = jazz16_sbdsp_reset(); + if (error < 0) + goto out; + + error = jazz16_sbdsp_command(SB_DSP_GET_JAZZ_VERSION); + if (error < 0) + goto out; + + error = jazz16_sbdsp_get_byte(); + if (error < 0) + goto out; + + error = SB_VERSION_IS_JAZZ16(error) ? 0 : -ENODEV; + +out: release_region(port, 16); + return error; +} + +static irqreturn_t snd_jazz16_interrupt(int irq, void *dev_id) +{ + return snd_sb8dsp_interrupt(dev_id); +} + +static void snd_jazz16_free(struct snd_card *card) +{ + jazz16_set_ports(0); + release_and_free_resource(card->private_data); +} + +static int __devinit snd_jazz16_probe(struct device *dev, unsigned int n) +{ + struct snd_sb *chip; + struct snd_card *card; + struct snd_opl3 *opl3; + int error; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (!card) + return -EINVAL; + + card->private_data = request_region(SB_JAZZ16_CONFIG_PORT, 1, + "Jazz16 Config"); + if (!card->private_data) { + error = -EBUSY; + goto out; + } + card->private_free = snd_jazz16_free; + + if (mpu_port == SNDRV_AUTO_PORT) + mpu_port = 0; + + if (mpu_irq == SNDRV_AUTO_IRQ) + mpu_irq = 0; + + error = jazz16_enable(dev); + if (error < 0) + goto out; + + error = snd_sbdsp_create(card, port, irq, snd_jazz16_interrupt, + dma8, dma16, SB_HW_AUTO, &chip); + if (error < 0) + goto out; + + if (chip->hardware != SB_HW_JAZZ16) { + snd_printk(KERN_ERR "%s: not a Jazz16 chip at %#lx.\n", + dev->bus_id, chip->port); + error = -ENODEV; + goto out; + } + + strcpy(card->driver, "Jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, "%s at %#lx, irq %d, dma8 %d, dma16 %d", + chip->name, chip->port, chip->irq, chip->dma8, chip->dma16); + + error = snd_sb8dsp_pcm(chip, 0, NULL); + if (error < 0) + goto out; + + error = snd_sbmixer_new(chip); + if (error < 0) + goto out; + + error = snd_opl3_create(card, chip->port, chip->port + 2, OPL3_HW_AUTO, + 1, &opl3); + if (error < 0) + snd_printk(KERN_WARNING "%s: no OPL device found, skipping.\n", + dev->bus_id); + else { + error = snd_opl3_timer_new(opl3, 1, 2); + if (error < 0) + goto out; + + error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (error < 0) + goto out; + } + + snd_card_set_dev(card, dev); + + error = snd_card_register(card); + if (error < 0) + goto out; + + dev_set_drvdata(dev, card); + return 0; + +out: snd_card_free(card); + return error; +} + +static int __devexit snd_jazz16_remove(struct device *dev, unsigned int n) +{ + snd_card_free(dev_get_drvdata(dev)); + dev_set_drvdata(dev, NULL); + return 0; +} + +static struct isa_driver snd_jazz16_driver = { + .match = snd_jazz16_match, + .probe = snd_jazz16_probe, + .remove = __devexit_p(snd_jazz16_remove), + + .driver = { + .name = DEV_NAME + } +}; + +static int __devinit alsa_card_jazz16_init(void) +{ + return isa_register_driver(&snd_jazz16_driver, 1); +} + +static void __devexit alsa_card_jazz16_exit(void) +{ + isa_unregister_driver(&snd_jazz16_driver); +} + +module_init (alsa_card_jazz16_init); +module_exit (alsa_card_jazz16_exit); diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index aea9e5e..70cd031 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -28,6 +28,9 @@ * * Wed Jul 12 22:02:55 CEST 2000 Uros Bizjak <uros(a)kss-loka.si> * Cleaned up and rewrote lowlevel routines. + * + * Sun Mar 11 18:45:38 CEST 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * Added Jazz16 enhancements. */ #include <sound/driver.h> @@ -73,6 +76,15 @@ static struct snd_ratnum stereo_clocks[] } }; +/* For stereo playback and capture, the denominator is divided by two, so it + * must be even to get the intended sample rate. */ +static struct snd_ratnum jazz16_stereo_clock = { + .num = SB8_CLOCK, + .den_min = 2, + .den_max = 512, + .den_step = 2, +}; + static int snd_sb8_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -101,14 +113,54 @@ static int snd_sb8_hw_constraint_channel return 0; } +static int snd_jazz16_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + unsigned int num = 0, den = 0; + int err = 0; + + if (c->max > 1) + err = snd_interval_ratnum(r, 1, &jazz16_stereo_clock, &num, &den); + else + err = snd_interval_ratnum(r, 1, &clock, &num, &den); + if (err >= 0 && den) { + params->rate_num = num; + params->rate_den = den; + } + return err; +} + +static int snd_jazz16_hw_constraint_channels_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + /* Force mono mode if the sample rate interval doesn't allow stereo. */ + if (SB8_DEN(r->min) == SB8_DEN(r->max) + && SB8_DEN(r->min) & 1) + { + struct snd_interval t = { .min = 1, .max = 1 }; + struct snd_interval *c; + + c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + err = snd_interval_refine(c, &t); + } + return err; +} + static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) { unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int mixreg, rate, size, count; + unsigned int mixreg, rate, size, count, dma; + unsigned char stereo, format; rate = runtime->rate; + stereo = runtime->channels > 1; switch (chip->hardware) { case SB_HW_PRO: if (runtime->channels > 1) { @@ -117,6 +169,12 @@ static int snd_sb8_playback_prepare(stru break; } /* fallthru */ + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_MU_LAW) { + chip->playback_format = SB_DSP_ULAW_OUTPUT_AUTO; + break; + } + /* fall through */ case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; @@ -134,8 +192,26 @@ static int snd_sb8_playback_prepare(stru } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); + switch (runtime->format) + { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_MU_LAW: + dma = chip->dma8; + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + dma = chip->dma16; + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + break; + + default: + return -EINVAL; + } spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); + if (chip->hardware >= SB_HW_PRO) + snd_sbdsp_command(chip, format); if (runtime->channels > 1) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); @@ -173,7 +249,7 @@ static int snd_sb8_playback_prepare(stru snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); return 0; } @@ -232,9 +308,11 @@ static int snd_sb8_capture_prepare(struc unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int mixreg, rate, size, count; + unsigned int mixreg, rate, size, count, dma; + unsigned char stereo, format; rate = runtime->rate; + stereo = runtime->channels > 1; switch (chip->hardware) { case SB_HW_PRO: if (runtime->channels > 1) { @@ -242,6 +320,12 @@ static int snd_sb8_capture_prepare(struc chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } + /* fall through */ + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_MU_LAW) { + chip->capture_format = SB_DSP_ULAW_OUTPUT_AUTO; + break; + } chip->capture_format = (rate > 23000) ? SB_DSP_HI_INPUT_AUTO : SB_DSP_LO_INPUT_AUTO; break; case SB_HW_201: @@ -261,10 +345,26 @@ static int snd_sb8_capture_prepare(struc } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); + switch (runtime->format) + { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_MU_LAW: + dma = chip->dma8; + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + dma = chip->dma16; + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + break; + + default: + return -EINVAL; + } spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) - snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); + if (chip->hardware >= SB_HW_PRO) + snd_sbdsp_command(chip, format); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (runtime->channels > 1) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); @@ -278,14 +378,14 @@ static int snd_sb8_capture_prepare(struc } else { snd_sbdsp_command(chip, 256 - runtime->rate_den); } - if (chip->capture_format != SB_DSP_OUTPUT) { + if (chip->capture_format != SB_DSP_INPUT) { count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); return 0; } @@ -358,10 +458,12 @@ static snd_pcm_uframes_t snd_sb8_playbac { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; if (chip->mode != SB_MODE_PLAYBACK_8) return 0; - ptr = snd_dma_pointer(chip->dma8, chip->p_dma_size); + dma = (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? chip->dma16 : chip->dma8; + ptr = snd_dma_pointer(dma, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -369,10 +471,12 @@ static snd_pcm_uframes_t snd_sb8_capture { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; if (chip->mode != SB_MODE_CAPTURE_8) return 0; - ptr = snd_dma_pointer(chip->dma8, chip->c_dma_size); + dma = (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? chip->dma16 : chip->dma8; + ptr = snd_dma_pointer(dma, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -443,6 +547,20 @@ static int snd_sb8_open(struct snd_pcm_s runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_MU_LAW; + runtime->hw.rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_44100; + runtime->hw.rate_max = 45455; + runtime->hw.channels_max = 2; + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_jazz16_hw_constraint_rate_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_jazz16_hw_constraint_channels_rate, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; @@ -463,9 +581,11 @@ static int snd_sb8_open(struct snd_pcm_s default: break; } + if (chip->hardware == SB_HW_JAZZ16) + return 0; snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clock); - return 0; + return 0; } static int snd_sb8_close(struct snd_pcm_substream *substream) diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 3094f38..69e72b5 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -146,8 +146,16 @@ static int snd_sbdsp_probe(struct snd_sb } break; case 3: - chip->hardware = SB_HW_PRO; - str = "Pro"; + spin_lock_irqsave(&chip->reg_lock, flags); + if (snd_sbdsp_command(chip, SB_DSP_GET_JAZZ_VERSION) && + SB_VERSION_IS_JAZZ16(snd_sbdsp_get_byte (chip))) { + chip->hardware = SB_HW_JAZZ16; + str = "Pro (Jazz16)"; + } else { + chip->hardware = SB_HW_PRO; + str = "Pro"; + } + spin_unlock_irqrestore (&chip->reg_lock, flags); break; case 4: chip->hardware = SB_HW_16; diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 490b1ca..b16b592 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -811,6 +811,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -946,6 +947,7 @@ void snd_sbmixer_suspend(struct snd_sb * save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -971,6 +973,7 @@ void snd_sbmixer_resume(struct snd_sb *c restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16:

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[alsa-devel] [Fwd: Re: [Alsa-devel] [RFC][RFT] Adding support for Jazz16 based sound cards]
by Rene Herman 20 Mar '07

20 Mar '07

I didn't see this one make the old list at all, so forward to the new one... -------- Original Message -------- Subject: Re: [Alsa-devel] [RFC][RFT] Adding support for Jazz16 based sound cards Date: Mon, 19 Mar 2007 23:46:07 +0100 From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> To: Rene Herman <rene.herman(a)gmail.com> CC: alsa-devel(a)lists.sourceforge.net, Adam Belay <ambx1(a)neo.rr.com> References: <20070311232436.GB9778(a)sygehus.dk> <45FDDA0F.7050503(a)gmail.com> On Mon, Mar 19, 2007 at 01:32:15AM +0100, Rene Herman wrote: > > I just now tested this on a standalone Jazz16 and it seems to be working > fine for me. I once had ogg123 segfault on me at the end of playback but > was unsuccesful in reproducing -- may have been a fluke. Thanks. I'll try out ogg123. > As to the comments... > > The changes to the SB code to support the Jazz16 I ofcourse agree with. > Supporting a new card is great. I am however not a huge fan of the new > PnP protocol. It's cute, but I feel it's a little over-engineered and > could just as well live directly in snd-jazz16. See below. > This setup also shares that same problem with the BIOS generally not > having reserved/routed the resources -- on a general PCI/ISA system the > user needs to go into the BIOS setup and reserve the resources the card > is (or will be, in this case) set to. So now the user really needs to > know what resources are going to be assigned, destroying the plug and > play idea... Not really. A standards compliant PnPBIOS set up with "PnP aware OS = YES" will use only enough resources to boot the system, which should leave enough irqs and dmas for the Linux PnP layer to do its work. I think there are three good reasons to use the PnP layer: 1) The PnP layer informs the PCI layer of irqs to try to avoid - please see pnp_register_irq_resource() in drivers/pnp/resource.c. 2) The PnP layer knows which resources are used by active devices, even if they have not been allocated using e.g. request_region(). This happens in practice. For example: $ cat /sys/bus/pnp/devices/01\:02.00/resources state = active io 0x2a0-0x2af io 0x390-0x393 io 0x500-0x50f irq 5 dma 0 dma 1 $ cat /proc/ioports [...] 0200-0200 : ns558-pnp 0201-0201 : ns558-pnp 0208-020f : ns558-pnp 0213-0213 : ISAPnP 0280-028f : ES18xx 0356-0356 : ide2 0360-0361 : MPU401 UART 0376-0376 : 0000:00:07.1 0376-0376 : ide1 0390-0391 : OPL2/3 (left) 0392-0393 : OPL2/3 (right) 03c0-03df : vga+ 03e8-03ef : serial 03f6-03f6 : 0000:00:07.1 03f6-03f6 : ide0 03f8-03ff : serial 0500-050f : AD1816A 0800-0807 : ES18xx - CTRL [...] Notice that 0x2a0-0x2af is missing from /proc/ioports. The PnP layer will avoid using that port range because it knows a device is using it. 3) Knowing which resources a card can utilize, the PnP layer can try to stay away from those resources when configuring other cards. The PnP layer doesn't do that yet in a stock kernel, but I've patched it to do so. I'll try to get those patches into the kernel. An advantage of autoprobing in a PnP protocol driver (as opposed to doing so in a sound card driver) is that it happens before any file systems have been mounted, so if the autoprobe happens to find one of those #%!$@ NE2000 cards and lock up the system, it won't corrupt your disks. -- Rask Ingemann Lambertsen

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[alsa-devel] test1
by Jaroslav Kysela 20 Mar '07

20 Mar '07

test1 ----- Jaroslav Kysela <perex(a)perex.cz>

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[alsa-devel] test
by Jaroslav Kysela 20 Mar '07

20 Mar '07

test ----- Jaroslav Kysela <perex(a)perex.cz>

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