[PATCH] cs46xx Phase reversal fixes and some AC3 pass through progress

[PATCH] cs46xx Phase reversal fixes and some AC3 pass through progress

[Benny Sjostrand]

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Hi!

In this patch the phase problems should be gone, if still think that you 
got "phase reversal" problem
please sendme a dump of the /proc/asound/card0/dsp/sample content while 
you are playbacking
whatever giving you "phase reversal" problem.

On the AC3 front I've managed to get my receiver to switch on to 
"Digital Surround" mode (instead of PCM), which
must mean that it finally detected the AC3 stream. But still no sound, 
 got no idea why, but please if you
have time give it a test.

Consider this patch as "experimental", just in the mean, dont be 
surprised if something that worked
OK before is now broken.

Summar of changes:
- Phase reversal fixes.
- Some rewrites on playback code, related to the sample rate setup.
- AC3 stuff ....

/Benny

[-- Attachment #2: cs46xx.patch --]
[-- Type: text/plain, Size: 21412 bytes --]

diff --exclude=CVS -Naur alsa-kernel/include/cs46xx.h ../cvs/alsa-kernel/include/cs46xx.h
--- alsa-kernel/include/cs46xx.h	Mon Nov  4 10:57:10 2002
+++ ../cvs/alsa-kernel/include/cs46xx.h	Sun Dec  8 19:03:27 2002
@@ -1663,6 +1663,8 @@
 	snd_pcm_substream_t *substream;
 
 	pcm_channel_descriptor_t * pcm_channel;
+
+	int pcm_channel_id;    /* Fron Rear, Center Lfe  ... */
 } cs46xx_pcm_t;
 
 typedef struct {
diff --exclude=CVS -Naur alsa-kernel/include/cs46xx_dsp_spos.h ../cvs/alsa-kernel/include/cs46xx_dsp_spos.h
--- alsa-kernel/include/cs46xx_dsp_spos.h	Sun Dec  8 13:21:47 2002
+++ ../cvs/alsa-kernel/include/cs46xx_dsp_spos.h	Sun Dec  8 19:04:51 2002
@@ -140,7 +140,6 @@
 	dsp_scb_descriptor_t * pcm_reader_scb;
 	dsp_scb_descriptor_t * src_scb;
 	dsp_scb_descriptor_t * mixer_scb;
-	int pcm_channel_id;
 
 	void * private_data;
 } pcm_channel_descriptor_t;
diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/cs46xx_lib.c ../cvs/alsa-kernel/pci/cs46xx/cs46xx_lib.c
--- alsa-kernel/pci/cs46xx/cs46xx_lib.c	Sun Dec  8 13:21:55 2002
+++ ../cvs/alsa-kernel/pci/cs46xx/cs46xx_lib.c	Sun Dec  8 20:29:22 2002
@@ -952,7 +952,7 @@
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
 		/* magic value to unmute PCM stream  playback volume */
 		snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address + 
-				       SCBVolumeCtrl) << 2, 0x80007fff);
+				       SCBVolumeCtrl) << 2, 0x80008000);
 
 		if (cpcm->pcm_channel->unlinked)
 			cs46xx_dsp_pcm_link(chip,cpcm->pcm_channel);
@@ -1032,32 +1032,27 @@
 static int _cs46xx_adjust_sample_rate (cs46xx_t *chip, cs46xx_pcm_t *cpcm,
 				       int sample_rate) 
 {
-	/* if this is the only PCMReaderSCB child under current
-	   SrcTask then there no need to re-create pcm-channel */
-	if ( cpcm->pcm_channel->src_scb->ref_count == 1 &&
-	     cpcm->pcm_channel->sample_rate != sample_rate &&
-	     /* never set a 0 khz sample rate */
-	     sample_rate) {
-		/* sample rate not set or we can reuse
-		   the same SRC*/
-		cs46xx_dsp_set_src_sample_rate (chip,cpcm->pcm_channel->src_scb,sample_rate);
-		cpcm->pcm_channel->sample_rate = sample_rate;
-	} 
 
-	/* if there is more then 1 PCMReaderSCB child's under current
-	   SrcTask then we must recreate channel */
-	if (cpcm->pcm_channel->sample_rate != sample_rate &&
-	    cpcm->pcm_channel->src_scb->ref_count != 1 &&
-	    /* never set a 0 khz sample rate */
-	    sample_rate) {
+	/* If PCMReaderSCB and SrcTaskSCB not created yet ... */
+	if ( cpcm->pcm_channel == NULL) {
+		cpcm->pcm_channel = cs46xx_dsp_create_pcm_channel (chip, sample_rate, 
+								   cpcm, cpcm->hw_addr,cpcm->pcm_channel_id);
+		if (cpcm->pcm_channel == NULL) {
+			snd_printk(KERN_ERR "cs46xx: failed to create virtual PCM channel\n");
+			return -ENOMEM;
+		}
+		cpcm->pcm_channel->sample_rate = sample_rate;
+	} else
+	/* if sample rate is changed */
+	if (cpcm->pcm_channel->sample_rate != sample_rate) {
 		int unlinked = cpcm->pcm_channel->unlinked;
 		cs46xx_dsp_destroy_pcm_channel (chip,cpcm->pcm_channel);
 
 		if ( (cpcm->pcm_channel = cs46xx_dsp_create_pcm_channel (chip, sample_rate, cpcm, 
 									 cpcm->hw_addr,
-									 cpcm->pcm_channel->pcm_channel_id)) == NULL) {
+									 cpcm->pcm_channel_id)) == NULL) {
 			snd_printk(KERN_ERR "cs46xx: failed to re-create virtual PCM channel\n");
-			return -ENXIO;
+			return -ENOMEM;
 		}
 
 		if (!unlinked) cs46xx_dsp_pcm_link (chip,cpcm->pcm_channel);
@@ -1082,15 +1077,21 @@
 	cpcm = snd_magic_cast(cs46xx_pcm_t, runtime->private_data, return -ENXIO);
 
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
+	snd_assert (sample_rate != 0, return -ENXIO);
+
 	down (&chip->spos_mutex);
 
-	snd_assert (cpcm->pcm_channel != NULL);
-	
 	if (_cs46xx_adjust_sample_rate (chip,cpcm,sample_rate)) {
 		up (&chip->spos_mutex);
 		return -ENXIO;
 	}
 
+	snd_assert (cpcm->pcm_channel != NULL);
+	if (!cpcm->pcm_channel) {
+		up (&chip->spos_mutex);
+		return -ENXIO;
+	}
+
 	if (cs46xx_dsp_pcm_channel_set_period (chip,cpcm->pcm_channel,period_size * 4)) {
 		 up (&chip->spos_mutex);
 		 return -EINVAL;
@@ -1108,11 +1109,11 @@
 
 
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
-		if (cpcm->pcm_channel->pcm_channel_id == DSP_PCM_MAIN_CHANNEL) {
+		if (cpcm->pcm_channel_id == DSP_PCM_MAIN_CHANNEL) {
 			substream->ops = &snd_cs46xx_playback_ops;
-		} else if (cpcm->pcm_channel->pcm_channel_id == DSP_PCM_REAR_CHANNEL) {
+		} else if (cpcm->pcm_channel_id == DSP_PCM_REAR_CHANNEL) {
 			substream->ops = &snd_cs46xx_playback_rear_ops;
-		} else if (cpcm->pcm_channel->pcm_channel_id == DSP_IEC958_CHANNEL) {
+		} else if (cpcm->pcm_channel_id == DSP_IEC958_CHANNEL) {
 			substream->ops = &snd_cs46xx_playback_iec958_ops;
 		} else {
 			snd_assert(0);
@@ -1135,11 +1136,11 @@
 		}
 
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
-		if (cpcm->pcm_channel->pcm_channel_id == DSP_PCM_MAIN_CHANNEL) {
+		if (cpcm->pcm_channel_id == DSP_PCM_MAIN_CHANNEL) {
 			substream->ops = &snd_cs46xx_playback_indirect_ops;
-		} else if (cpcm->pcm_channel->pcm_channel_id == DSP_PCM_REAR_CHANNEL) {
+		} else if (cpcm->pcm_channel_id == DSP_PCM_REAR_CHANNEL) {
 			substream->ops = &snd_cs46xx_playback_indirect_rear_ops;
-		} else if (cpcm->pcm_channel->pcm_channel_id == DSP_IEC958_CHANNEL) {
+		} else if (cpcm->pcm_channel_id == DSP_IEC958_CHANNEL) {
 			substream->ops = &snd_cs46xx_playback_indirect_iec958_ops;
 		} else {
 			snd_assert(0);
@@ -1165,6 +1166,10 @@
 
 	cpcm = snd_magic_cast(cs46xx_pcm_t, runtime->private_data, return -ENXIO);
 
+	/* if play_back open fails, then this function
+	   is called and cpcm can actually be NULL here */
+	if (!cpcm) return -ENXIO;
+
 	if (runtime->dma_area != cpcm->hw_area)
 		snd_pcm_lib_free_pages(substream);
     
@@ -1482,15 +1487,8 @@
 	cpcm->substream = substream;
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
 	down (&chip->spos_mutex);
-	cpcm->pcm_channel = cs46xx_dsp_create_pcm_channel (chip, runtime->rate, cpcm, cpcm->hw_addr,pcm_channel_id);
-
-	if (cpcm->pcm_channel == NULL) {
-		snd_printk(KERN_ERR "cs46xx: failed to create virtual PCM channel\n");
-		snd_free_pci_pages(chip->pci, cpcm->hw_size, cpcm->hw_area, cpcm->hw_addr);
-		snd_magic_kfree(cpcm);
-		up (&chip->spos_mutex);
-		return -ENOMEM;
-	}
+	cpcm->pcm_channel = NULL; 
+	cpcm->pcm_channel_id = pcm_channel_id;
 
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 
 				   &hw_constraints_period_sizes);
@@ -1583,6 +1581,9 @@
 
 	cpcm = snd_magic_cast(cs46xx_pcm_t, runtime->private_data, return -ENXIO);
 
+	/* when playback_open fails, then cpcm can be NULL */
+	if (!cpcm) return -ENXIO;
+
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
 	down (&chip->spos_mutex);
 	if (cpcm->pcm_channel) {
@@ -1887,8 +1888,8 @@
 {
 	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
 
-	ucontrol->value.integer.value[0] = chip->dsp_spos_instance->dac_volume_right;
-	ucontrol->value.integer.value[1] = chip->dsp_spos_instance->dac_volume_left;
+	ucontrol->value.integer.value[0] = chip->dsp_spos_instance->dac_volume_left;
+	ucontrol->value.integer.value[1] = chip->dsp_spos_instance->dac_volume_right;
 
 	return 0;
 }
@@ -3331,8 +3332,8 @@
 	
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
 	/* set the attenuation to 0dB */ 
-	snd_cs46xx_poke(chip, (MASTERMIX_SCB_ADDR + 0xE) << 2, 0x80008000); 
-	snd_cs46xx_poke(chip, (VARIDECIMATE_SCB_ADDR + 0xE) << 2, 0x80008000);
+	/* snd_cs46xx_poke(chip, (MASTERMIX_SCB_ADDR + 0xE) << 2, 0x80008000); 
+	   snd_cs46xx_poke(chip, (VARIDECIMATE_SCB_ADDR + 0xE) << 2, 0x80008000); */
 
 	/*
 	 * Initialize cs46xx SPDIF controller
diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/cs46xx_lib.h ../cvs/alsa-kernel/pci/cs46xx/cs46xx_lib.h
--- alsa-kernel/pci/cs46xx/cs46xx_lib.h	Sun Dec  8 13:21:55 2002
+++ ../cvs/alsa-kernel/pci/cs46xx/cs46xx_lib.h	Sun Dec  8 18:55:22 2002
@@ -152,6 +152,7 @@
                                                         dsp_scb_descriptor_t * parent_scb,
                                                         int scb_child_type);
 dsp_scb_descriptor_t *  cs46xx_dsp_create_src_task_scb(cs46xx_t * chip,char * scb_name,
+						       int sample_rate,
                                                        u16 src_buffer_addr,
                                                        u16 src_delay_buffer_addr,u32 dest,
                                                        dsp_scb_descriptor_t * parent_scb,
@@ -202,8 +203,6 @@
                                                           int pcm_channel_id);
 void                       cs46xx_dsp_destroy_pcm_channel (cs46xx_t * chip,
                                                            pcm_channel_descriptor_t * pcm_channel);
-void                       cs46xx_dsp_set_src_sample_rate(cs46xx_t * chip,dsp_scb_descriptor_t * src, 
-                                                          u32 rate);
 int                        cs46xx_dsp_pcm_unlink (cs46xx_t * chip,pcm_channel_descriptor_t * pcm_channel);
 int                        cs46xx_dsp_pcm_link (cs46xx_t * chip,pcm_channel_descriptor_t * pcm_channel);
 dsp_scb_descriptor_t *     cs46xx_add_record_source (cs46xx_t *chip,dsp_scb_descriptor_t * source,
diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/dsp_spos.c ../cvs/alsa-kernel/pci/cs46xx/dsp_spos.c
--- alsa-kernel/pci/cs46xx/dsp_spos.c	Sun Dec  8 13:21:55 2002
+++ ../cvs/alsa-kernel/pci/cs46xx/dsp_spos.c	Sun Dec  8 19:08:46 2002
@@ -615,7 +615,7 @@
 	snd_iprintf(buffer,"\nMIX_SAMPLE_BUF1:\n");
 
 	col = 0;
-	for (i = MIX_SAMPLE_BUF1;i < MIX_SAMPLE_BUF1 + 0x30; i += sizeof(u32),col ++) {
+	for (i = MIX_SAMPLE_BUF1;i < MIX_SAMPLE_BUF1 + 0x40; i += sizeof(u32),col ++) {
 		if (col == 4) {
 			snd_iprintf(buffer,"\n");
 			col = 0;
@@ -628,14 +628,14 @@
 		snd_iprintf(buffer,"%08X ",readl(dst + i));
 	}
 
-	snd_iprintf(buffer,"\n\n");
+	snd_iprintf(buffer,"\nSRC_TASK_SCB1:\n");
 	col = 0;
-	for (i = SPDIFI_IP_OUTPUT_BUFFER1;i < SPDIFI_IP_OUTPUT_BUFFER1 + 0x40; i += sizeof(u32),col ++) {
+	for (i = 0x2580 ; i < 0x2580 + 0x40 ; i += sizeof(u32),col ++) {
 		if (col == 4) {
 			snd_iprintf(buffer,"\n");
 			col = 0;
 		}
-
+		
 		if (col == 0) {
 			snd_iprintf(buffer, "%04X ",i);
 		}
@@ -646,7 +646,7 @@
 
 	snd_iprintf(buffer,"\nSPDIFO_BUFFER:\n");
 	col = 0;
-	for (i = SPDIFO_IP_OUTPUT_BUFFER1;i < SPDIFO_IP_OUTPUT_BUFFER1 + 0x40; i += sizeof(u32),col ++) {
+	for (i = SPDIFO_IP_OUTPUT_BUFFER1;i < SPDIFO_IP_OUTPUT_BUFFER1 + 0x30; i += sizeof(u32),col ++) {
 		if (col == 4) {
 			snd_iprintf(buffer,"\n");
 			col = 0;
@@ -1356,6 +1356,7 @@
 
 	/* SPDIF input sampel rate converter */
 	src_task_scb = cs46xx_dsp_create_src_task_scb(chip,"SrcTaskSCB_SPDIFI",
+						      48000,
 						      SRC_OUTPUT_BUF1,
 						      SRC_DELAY_BUF1,SRCTASK_SCB_ADDR,
 						      master_mix_scb,
@@ -1601,7 +1602,7 @@
 
 	/* set SPDIF input sample rate and unmute
 	   NOTE: only 48khz support for SPDIF input this time */
-	cs46xx_dsp_set_src_sample_rate(chip,ins->spdif_in_src,48000);
+	/* cs46xx_dsp_set_src_sample_rate(chip,ins->spdif_in_src,48000); */
 
 	/* monitor state */
 	ins->spdif_status_in = 1;
diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/dsp_spos_scb_lib.c ../cvs/alsa-kernel/pci/cs46xx/dsp_spos_scb_lib.c
--- alsa-kernel/pci/cs46xx/dsp_spos_scb_lib.c	Sun Dec  8 13:21:55 2002
+++ ../cvs/alsa-kernel/pci/cs46xx/dsp_spos_scb_lib.c	Mon Dec  9 00:49:07 2002
@@ -576,8 +576,11 @@
 	return scb;
 }
 
+#define GOF_PER_SEC 200
+
 dsp_scb_descriptor_t * 
 cs46xx_dsp_create_src_task_scb(cs46xx_t * chip,char * scb_name,
+			       int rate,
                                u16 src_buffer_addr,
                                u16 src_delay_buffer_addr,u32 dest,
                                dsp_scb_descriptor_t * parent_scb,
@@ -586,45 +589,81 @@
 
 	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
 	dsp_scb_descriptor_t * scb;
-    
-	src_task_scb_t src_task_scb = {
-		0x0028,0x00c8,
-		0x5555,0x0000,
-		0x0000,0x0000,
-		src_buffer_addr,1,
-		0x0028,0x00c8,
-		RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_32,  
-		0x0000,src_delay_buffer_addr,                  
-		0x0,                                            
-		0x80,(src_delay_buffer_addr + (24 * 4)),
-		0,0, /* next_scb, sub_list_ptr */
-		0,0, /* entry, this_spb */
-		RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_8,
-		src_buffer_addr << 0x10,
-		0x04000000,
-		{ 
-			0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left,
-			0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left
-		}
-	};
+	unsigned int tmp1, tmp2;
+	unsigned int phiIncr;
+	unsigned int correctionPerGOF, correctionPerSec;
 
-	if (ins->s16_up == NULL) {
-		ins->s16_up =  cs46xx_dsp_lookup_symbol (chip,"S16_UPSRC",
-							 SYMBOL_CODE);
-    
-		if (ins->s16_up == NULL) {
-			snd_printk (KERN_ERR "dsp_spos: symbol S16_UPSRC not found\n");
-			return NULL;
-		}    
-	}
+	snd_printdd( "dsp_spos: setting %s rate to %u\n",scb_name,rate);
 
-	/* clear buffers */
-	_dsp_clear_sample_buffer (chip,src_buffer_addr,8);
-	_dsp_clear_sample_buffer (chip,src_delay_buffer_addr,32);
+	/*
+	 *  Compute the values used to drive the actual sample rate conversion.
+	 *  The following formulas are being computed, using inline assembly
+	 *  since we need to use 64 bit arithmetic to compute the values:
+	 *
+	 *  phiIncr = floor((Fs,in * 2^26) / Fs,out)
+	 *  correctionPerGOF = floor((Fs,in * 2^26 - Fs,out * phiIncr) /
+	 *                                   GOF_PER_SEC)
+	 *  ulCorrectionPerSec = Fs,in * 2^26 - Fs,out * phiIncr -M
+	 *                       GOF_PER_SEC * correctionPerGOF
+	 *
+	 *  i.e.
+	 *
+	 *  phiIncr:other = dividend:remainder((Fs,in * 2^26) / Fs,out)
+	 *  correctionPerGOF:correctionPerSec =
+	 *      dividend:remainder(ulOther / GOF_PER_SEC)
+	 */
+	tmp1 = rate << 16;
+	phiIncr = tmp1 / 48000;
+	tmp1 -= phiIncr * 48000;
+	tmp1 <<= 10;
+	phiIncr <<= 10;
+	tmp2 = tmp1 / 48000;
+	phiIncr += tmp2;
+	tmp1 -= tmp2 * 48000;
+	correctionPerGOF = tmp1 / GOF_PER_SEC;
+	tmp1 -= correctionPerGOF * GOF_PER_SEC;
+	correctionPerSec = tmp1;
 
-	scb = _dsp_create_generic_scb(chip,scb_name,(u32 *)&src_task_scb,
-				      dest,ins->s16_up,parent_scb,
-				      scb_child_type);
+	{
+		src_task_scb_t src_task_scb = {
+			0x0028,0x00c8,
+			0x5555,0x0000,
+			0x0000,0x0000,
+			src_buffer_addr,1,
+			correctionPerGOF,correctionPerSec,
+			RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_32,  
+			0x0000,src_delay_buffer_addr,                  
+			0x0,                                            
+			0x080,(src_delay_buffer_addr + (24 * 4)),
+			0,0, /* next_scb, sub_list_ptr */
+			0,0, /* entry, this_spb */
+			RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_8,
+			src_buffer_addr << 0x10,
+			phiIncr,
+			{ 
+				0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left,
+				0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left
+			}
+		};
+		
+		if (ins->s16_up == NULL) {
+			ins->s16_up =  cs46xx_dsp_lookup_symbol (chip,"S16_UPSRC",
+								 SYMBOL_CODE);
+			
+			if (ins->s16_up == NULL) {
+				snd_printk (KERN_ERR "dsp_spos: symbol S16_UPSRC not found\n");
+				return NULL;
+			}    
+		}
+		
+		/* clear buffers */
+		_dsp_clear_sample_buffer (chip,src_buffer_addr,8);
+		_dsp_clear_sample_buffer (chip,src_delay_buffer_addr,32);
+		
+		scb = _dsp_create_generic_scb(chip,scb_name,(u32 *)&src_task_scb,
+					      dest,ins->s16_up,parent_scb,
+					      scb_child_type);
+	}
 
 	return scb;
 }
@@ -796,7 +835,7 @@
 
 		RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_16,
 		0,
-		0,input_scb->address, 
+		0,input_scb->address,
 		{
 			0x8000,0x8000,
 			0x8000,0x8000
@@ -1007,7 +1046,7 @@
 		/* C */ snoop_buffer_address  << 0x10,
 		/* D */ 0,
 		/* E */ { 0x8000,0x8000,
-			  /* F */   0xffff,0xffff
+	        /* F */   0xffff,0xffff
 		}
 	};
 
@@ -1093,6 +1132,8 @@
 {
 	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
 	dsp_scb_descriptor_t * src_scb = NULL,* pcm_scb, * mixer_scb = NULL;
+	dsp_scb_descriptor_t * src_parent_scb = NULL;
+
 	/*dsp_scb_descriptor_t * pcm_parent_scb;*/
 	char scb_name[DSP_MAX_SCB_NAME];
 	int i,pcm_index = -1, insert_point, src_index = -1;
@@ -1116,13 +1157,14 @@
 	case DSP_IEC958_CHANNEL:
 		snd_assert (ins->asynch_tx_scb != NULL, return NULL);
 		mixer_scb = ins->asynch_tx_scb;
-#if 0
-		if (ins->spdif_status_out & DSP_SPDIF_STATUS_AC3_MODE) {
-			snd_printdd ("IEC958 opened in AC3 mode\n");
-			/*src_scb = ins->asynch_tx_scb;
-			  ins->asynch_tx_scb->ref_count ++;*/
+
+		/* if sample rate is set to 48khz we pass
+		   the Sample Rate Converted (which could
+		   alter the raw data stream ...) */
+		if (sample_rate == 48000) {
+			snd_printdd ("IEC958 pass through\n");
+			src_parent_scb = ins->asynch_tx_scb;
 		}
-#endif
 		break;
 	default:
 		snd_assert (0);
@@ -1158,8 +1200,6 @@
 	}
 
 	if (src_scb == NULL) {
-		dsp_scb_descriptor_t * src_parent_scb;
-
 		if (ins->nsrc_scb >= DSP_MAX_SRC_NR) {
 			snd_printk(KERN_ERR "dsp_spos: to many SRC instances\n!");
 			return NULL;
@@ -1176,18 +1216,21 @@
 		snd_assert (src_index != -1,return NULL);
 
 		/* we need to create a new SRC SCB */
-		if (mixer_scb->sub_list_ptr == ins->the_null_scb) {
-			src_parent_scb = mixer_scb;
-			insert_point = SCB_ON_PARENT_SUBLIST_SCB;
-		} else {
-			src_parent_scb = find_next_free_scb(chip,mixer_scb->sub_list_ptr);
-			insert_point = SCB_ON_PARENT_NEXT_SCB;
-		}
+		if (src_parent_scb == NULL) {
+			if (mixer_scb->sub_list_ptr == ins->the_null_scb) {
+				src_parent_scb = mixer_scb;
+				insert_point = SCB_ON_PARENT_SUBLIST_SCB;
+			} else {
+				src_parent_scb = find_next_free_scb(chip,mixer_scb->sub_list_ptr);
+				insert_point = SCB_ON_PARENT_NEXT_SCB;
+			}
+		} else insert_point = SCB_ON_PARENT_NEXT_SCB;
 
 		snprintf (scb_name,DSP_MAX_SCB_NAME,"SrcTask_SCB%d",src_index);
 
 		snd_printdd( "dsp_spos: creating SRC \"%s\"\n",scb_name);
 		src_scb = cs46xx_dsp_create_src_task_scb(chip,scb_name,
+							 sample_rate,
 							 src_output_buffer_addr[src_index],
 							 src_delay_buffer_addr[src_index],
 							 /* 0x400 - 0x600 source SCBs */
@@ -1200,7 +1243,7 @@
 			return NULL;
 		}
 
-		cs46xx_dsp_set_src_sample_rate(chip,src_scb,sample_rate);
+		/* cs46xx_dsp_set_src_sample_rate(chip,src_scb,sample_rate); */
 
 		ins->nsrc_scb ++;
 	} 
@@ -1226,6 +1269,19 @@
 		return NULL;
 	}
 
+	if (pcm_channel_id == DSP_IEC958_CHANNEL && sample_rate == 48000) {
+		snd_assert (ins->spdif_pcm_input_scb = NULL);
+
+		/* a hack to make the skip the SRC and pass the stream 
+		   directly to the SPDIF task */
+		ins->spdif_pcm_input_scb = 
+			cs46xx_dsp_create_pcm_serial_input_scb(chip,"PCMSerialInput_PCM",
+							       PCMSERIALINII_SCB_ADDR,
+							       pcm_scb,
+							       ins->asynch_tx_scb,
+							       SCB_ON_PARENT_SUBLIST_SCB);		
+	}
+
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	ins->pcm_channels[pcm_index].sample_rate = sample_rate;
 	ins->pcm_channels[pcm_index].pcm_reader_scb = pcm_scb;
@@ -1236,7 +1292,6 @@
 	ins->pcm_channels[pcm_index].active = 1;
 	ins->pcm_channels[pcm_index].pcm_slot = pcm_index;
 	ins->pcm_channels[pcm_index].mixer_scb = mixer_scb;
-	ins->pcm_channels[pcm_index].pcm_channel_id = pcm_channel_id;
 	ins->npcm_channels ++;
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 
@@ -1416,59 +1471,6 @@
 	return 0;
 }
 
-
-#define GOF_PER_SEC 200
-  
-void cs46xx_dsp_set_src_sample_rate(cs46xx_t *chip,dsp_scb_descriptor_t * src, u32 rate)
-{
-	unsigned long flags;
-	unsigned int tmp1, tmp2;
-	unsigned int phiIncr;
-	unsigned int correctionPerGOF, correctionPerSec;
-
-	snd_printdd( "dsp_spos: setting SRC rate to %u\n",rate);
-	/*
-	 *  Compute the values used to drive the actual sample rate conversion.
-	 *  The following formulas are being computed, using inline assembly
-	 *  since we need to use 64 bit arithmetic to compute the values:
-	 *
-	 *  phiIncr = floor((Fs,in * 2^26) / Fs,out)
-	 *  correctionPerGOF = floor((Fs,in * 2^26 - Fs,out * phiIncr) /
-	 *                                   GOF_PER_SEC)
-	 *  ulCorrectionPerSec = Fs,in * 2^26 - Fs,out * phiIncr -M
-	 *                       GOF_PER_SEC * correctionPerGOF
-	 *
-	 *  i.e.
-	 *
-	 *  phiIncr:other = dividend:remainder((Fs,in * 2^26) / Fs,out)
-	 *  correctionPerGOF:correctionPerSec =
-	 *      dividend:remainder(ulOther / GOF_PER_SEC)
-	 */
-	tmp1 = rate << 16;
-	phiIncr = tmp1 / 48000;
-	tmp1 -= phiIncr * 48000;
-	tmp1 <<= 10;
-	phiIncr <<= 10;
-	tmp2 = tmp1 / 48000;
-	phiIncr += tmp2;
-	tmp1 -= tmp2 * 48000;
-	correctionPerGOF = tmp1 / GOF_PER_SEC;
-	tmp1 -= correctionPerGOF * GOF_PER_SEC;
-	correctionPerSec = tmp1;
-
-	/*
-	 *  Fill in the SampleRateConverter control block.
-	 */
-	spin_lock_irqsave(&chip->reg_lock, flags);
-
-	snd_cs46xx_poke(chip, (src->address + SRCCorPerGof) << 2,
-	  ((correctionPerSec << 16) & 0xFFFF0000) | (correctionPerGOF & 0xFFFF));
-
-	snd_cs46xx_poke(chip, (src->address + SRCPhiIncr6Int26Frac) << 2, phiIncr);
-
-	spin_unlock_irqrestore(&chip->reg_lock, flags);
-}
-
 dsp_scb_descriptor_t * cs46xx_add_record_source (cs46xx_t *chip,dsp_scb_descriptor_t * source,
                                                  u16 addr,char * scb_name)
 {
@@ -1655,10 +1657,16 @@
 	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_default);
 	
 	/* deallocate stuff */
+	if (ins->spdif_pcm_input_scb != NULL) {
+		cs46xx_dsp_remove_scb (chip,ins->spdif_pcm_input_scb);
+		ins->spdif_pcm_input_scb = NULL;
+	}
+
 	cs46xx_dsp_remove_scb (chip,ins->asynch_tx_scb);
 	ins->asynch_tx_scb = NULL;
 
 	/* restore state */
+
 	if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) {
 		cs46xx_dsp_enable_spdif_out (chip);
 	}

Re: [PATCH] cs46xx Phase reversal fixes and some AC3 pass through progress

[Takashi Iwai]

Hi Benny,

At Mon, 09 Dec 2002 01:12:09 +0100,
Benny Sjostrand wrote:
> 
> Hi!
> 
> In this patch the phase problems should be gone, if still think that you 
> got "phase reversal" problem
> please sendme a dump of the /proc/asound/card0/dsp/sample content while 
> you are playbacking
> whatever giving you "phase reversal" problem.
> 
> On the AC3 front I've managed to get my receiver to switch on to 
> "Digital Surround" mode (instead of PCM), which
> must mean that it finally detected the AC3 stream. But still no sound, 
>  got no idea why, but please if you
> have time give it a test.
> 
> Consider this patch as "experimental", just in the mean, dont be 
> surprised if something that worked
> OK before is now broken.
> 
> Summar of changes:
> - Phase reversal fixes.
> - Some rewrites on playback code, related to the sample rate setup.
> - AC3 stuff ....

thanks!  now committed on cvs.


the spdif output on my terratec x-fire seems working fine now, at
least for the normal playback.  however, if i stop the playback in the
middle, it results in a constant beep.  does happen on another card?

also, ac3dec seems not working yet on the card.
this could be due to another reason, though.


Takashi


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Re: [PATCH] cs46xx Phase reversal fixes and some AC3 pass through progress

[Friedrich Ewaldt]

Hi Benny,

I tested your patch and phase reversals are gone now. That's fine so 
far. But to apply your patch I had to install a version newer than rc6. 
I downloaded the cvs snapshot 2002-12-09.tar.bz2 and installed it 
successfully. After that the SPDIF ac3 mode switch is gone (like with 
rc6 + your previous patch). That seems to be ok.(?)
But now I have the following problems with my terratec xfire card that I 
didn't have with the unpatched rc6 version:

* SPDIF input doesn't work anymore: no sound. Even reinstalling 
unpatched rc6 only helps if I do a cold reboot afterwards. (restarting 
alsa after installing rc6 doesn't help)

* (probably related) SPDIF volume control can't be changed individually 
for left and right channel. If you change the volume of one channel, the 
other channel follows more or less. I get a max. volume difference betw. 
right and left channel of 4 (on the scale 0..100).

* full duplex operation causes much more xruns compared to unpatched rc6:
My application sets input and output buffer size to 1024 frames, chunk 
size to 512 and the avail_min (for the poll/wait function) to 512. Then 
I link playback and capture device and basically do the following:

// using buffer size = 2 * chunk size
// therefore, fill output buffer with 2*chunkSize frames before starting
err = snd_pcm_wait(pHandle, 1000);
    if(err < 0){fprintf(stderr, "output wait error(init0.5): %s\n", 
snd_strerror(err));return -1;}
err = snd_pcm_writei(pHandle, outbuf, chunkSize);
  if(err < 0){fprintf(stderr, "audio output error(init1): %s\n", 
snd_strerror(err));return -1;}
err = snd_pcm_wait(pHandle, 1000);
    if(err < 0){fprintf(stderr, "output wait error(init1.5): %s\n", 
snd_strerror(err));return -1;}
err = snd_pcm_writei(pHandle, outbuf, chunkSize);
    if(err < 0){fprintf(stderr, "audio output error(init2): %s\n", 
snd_strerror(err));return -1;}
// start audio device
err = snd_pcm_start(cHandle);
      if(err < 0){fprintf(stderr, "error starting streams: %s\n", 
snd_strerror(err));return -1;}

Then a loop of waits for the capture device, readi, wait for playback 
device, writei, ... follows.
This code worked with (almost) no xruns with rc6 driver. With cvs from 
2002-12-09 (patched for phase reversal or not does no difference), this 
code doesn't work anymore, the first writei fails with 'broken pipe' 
error. This shouldn't be the case since the snd_pcm_wait returned 
without error and there must be space left in the buffer. If I change 
the buffer size to 2048 frames (still chunkSize = 512), the program 
works in about 2 of 3 cases and fails after various chunks 
played/recorded otherwise with xruns. I don't know if I could have 
programmed something wrong, but it worked with rc6 without problems and 
stable. I don't see a reason why the first writei could fail if the 
buffer is empty and snd_pcm_wait has returned without error before.
Do you have an idea if the changes of the driver code could cause these 
problems?
  fe

PS. regarding the phase reversal problems: switching the 'pcm out 
path&mute' control from pre3d to post3d (or back) also reverses the 
phase of the pcm output. That's no problem for stereo sound because left 
and rigth channel always have same phase but it might cause problems 
with > 2 channels/surround sound.

Benny Sjostrand schrieb:

> Hi!
>
> In this patch the phase problems should be gone, if still think that 
> you got "phase reversal" problem
> please sendme a dump of the /proc/asound/card0/dsp/sample content 
> while you are playbacking
> whatever giving you "phase reversal" problem.
>
> On the AC3 front I've managed to get my receiver to switch on to 
> "Digital Surround" mode (instead of PCM), which
> must mean that it finally detected the AC3 stream. But still no sound, 
> got no idea why, but please if you
> have time give it a test.
>
> Consider this patch as "experimental", just in the mean, dont be 
> surprised if something that worked
> OK before is now broken.
>
> Summar of changes:
> - Phase reversal fixes.
> - Some rewrites on playback code, related to the sample rate setup.
> - AC3 stuff ....
>
> /Benny
>
>  
>



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Re: [PATCH] cs46xx Phase reversal fixes and some AC3 pass through progress

[Benny Sjostrand]

>
>
>the spdif output on my terratec x-fire seems working fine now, at
>least for the normal playback.  however, if i stop the playback in the
>  
>
>middle, it results in a constant beep.  does happen on another card?
>

It has not happen for me (yet)

Sounds like there some garbage left in some sample buffer.
Did you use the IEC958 PCM device (aplay -Dhw:0,2)  or just turn on the
SPDIF output switch in mixer ???

/Benny




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Re: [PATCH] cs46xx Phase reversal fixes and some AC3 pass through progress

[Takashi Iwai]

At Mon, 09 Dec 2002 22:06:21 +0100,
Benny Sjostrand wrote:
> 
> >
> >
> >the spdif output on my terratec x-fire seems working fine now, at
> >least for the normal playback.  however, if i stop the playback in the
> >  
> >
> >middle, it results in a constant beep.  does happen on another card?
> >
> 
> It has not happen for me (yet)
> 
> Sounds like there some garbage left in some sample buffer.
> Did you use the IEC958 PCM device (aplay -Dhw:0,2)  or just turn on the
> SPDIF output switch in mixer ???

i've used -Dhw:0,2.  well, it seems fixed on the latest version.

thanks!


Takashi


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