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alsa: instead of per-target tests, use HAVE_ALSA_32BIT

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Affects all Sony NWZ (linux) and the fiio m3k linux targets.

Change-Id: I2fcf121bd026103d2b72332a5a52cc2b5e93949f
This commit is contained in:
Solomon Peachy 2021-04-08 23:02:15 -04:00
parent e17337c9aa
commit 54fcb907c1
4 changed files with 34 additions and 28 deletions
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1
firmware/export/config/fiiom3klinux.h
View file

@ -117,6 +117,7 @@
/* Audio codec */
#define HAVE_FIIO_LINUX_CODEC
#define HAVE_ALSA_32BIT
/* We don't have hardware controls */
#define HAVE_SW_TONE_CONTROLS

1
firmware/export/config/sonynwzlinux.h
View file

@ -82,6 +82,7 @@
/* Audio codec */
#define HAVE_NWZ_LINUX_CODEC
#define HAVE_ALSA_32BIT
#endif /* SIMULATOR */

58
firmware/target/hosted/pcm-alsa.c
View file

@ -69,8 +69,7 @@
#endif
static const snd_pcm_access_t access_ = SND_PCM_ACCESS_RW_INTERLEAVED; /* access mode */
#if defined(SONY_NWZ_LINUX) || defined(HAVE_FIIO_LINUX_CODEC)
/* Sony NWZ must use 32-bit per sample */
#if defined(HAE_ALSA_32BIT)
static const snd_pcm_format_t format = SND_PCM_FORMAT_S32_LE; /* sample format */
typedef int32_t sample_t;
#else
@ -262,6 +261,7 @@ error:
return err;
}
#if defined(HAVE_ALSA_32BIT)
/* Digital volume explanation:
* with very good approximation (<0.1dB) the convertion from dB to multiplicative
* factor, for dB>=0, is 2^(dB/3). We can then notice that if we write dB=3*k+r
@ -303,6 +303,7 @@ void pcm_set_mixer_volume(int vol_db_l, int vol_db_r)
dig_vol_mult_r = 1 << vol_shift_r | 1 << (vol_shift_r - 1);
logf("r: %d dB -> factor = %d", vol_db_r - 48, dig_vol_mult_r);
}
#endif
/* copy pcm samples to a spare buffer, suitable for snd_pcm_writei() */
static bool copy_frames(bool first)
@ -343,37 +344,40 @@ static bool copy_frames(bool first)
panicf("Wrong pcm_size");
/* the compiler will optimize this test away */
nframes = MIN((ssize_t)pcm_size/4, frames_left);
if (format == SND_PCM_FORMAT_S32_LE)
{
/* We have to convert 16-bit to 32-bit, the need to multiply the
* sample by some value so the sound is not too low */
const int16_t *pcm_ptr = pcm_data;
sample_t *sample_ptr = &frames[2*(period_size-frames_left)];
for (int i = 0; i < nframes; i++)
{
*sample_ptr++ = *pcm_ptr++ * dig_vol_mult_l;
*sample_ptr++ = *pcm_ptr++ * dig_vol_mult_r;
}
}
else
{
#ifdef HAVE_RECORDING
switch (current_alsa_mode)
{
case SND_PCM_STREAM_PLAYBACK:
switch (current_alsa_mode)
{
case SND_PCM_STREAM_PLAYBACK:
#endif
#if defined(HAVE_ALSA_32BIT)
if (format == SND_PCM_FORMAT_S32_LE)
{
/* We have to convert 16-bit to 32-bit, the need to multiply the
* sample by some value so the sound is not too low */
const int16_t *pcm_ptr = pcm_data;
sample_t *sample_ptr = &frames[2*(period_size-frames_left)];
for (int i = 0; i < nframes; i++)
{
*sample_ptr++ = *pcm_ptr++ * dig_vol_mult_l;
*sample_ptr++ = *pcm_ptr++ * dig_vol_mult_r;
}
}
else
#endif
{
/* Rockbox and PCM have same format: memcopy */
memcpy(&frames[2*(period_size-frames_left)], pcm_data, nframes * 4);
}
#ifdef HAVE_RECORDING
break;
case SND_PCM_STREAM_CAPTURE:
memcpy(pcm_data_rec, &frames[2*(period_size-frames_left)], nframes * 4);
break;
default:
break;
}
#endif
break;
case SND_PCM_STREAM_CAPTURE:
memcpy(pcm_data_rec, &frames[2*(period_size-frames_left)], nframes * 4);
break;
default:
break;
}
#endif
pcm_data += nframes*4;
pcm_size -= nframes*4;
frames_left -= nframes;

2
firmware/target/hosted/pcm-alsa.h
View file

@ -22,7 +22,7 @@
#include <config.h>
#if defined(SONY_NWZ_LINUX) || defined(HAVE_FIIO_LINUX_CODEC)
#if defined(HAVE_ALSA_32BIT)
/* Set the PCM volume in dB: each sample with have this volume applied digitally
* before being sent to ALSA. Volume must satisfy -43 <= dB <= 0 */
void pcm_set_mixer_volume(int vol_db_l, int vol_db_r);