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nwz/alsa: various improvements

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Also audiohw driver to specific device name, rewrite alsa controls code to
cache more data, thus making the code easier and use less stack. Avoid using
short/long in pcm alsa code since it's the wrong size on 64-bit (simulator
for example)

Change-Id: Ibc1ec44396e37b6cbdedbcf37300878638e5d2d3
This commit is contained in:
Amaury Pouly 2018-02-19 16:28:11 +11:00 • committed by Solomon Peachy
parent ce9e7e7122
commit dd82f13fa1
10 changed files with 122 additions and 93 deletions
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18
firmware/target/hosted/pcm-alsa.c
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@ -66,10 +66,10 @@ static const snd_pcm_access_t access_ = SND_PCM_ACCESS_RW_INTERLEAVED; /* access
#if defined(SONY_NWZ_LINUX) || defined(HAVE_FIIO_LINUX_CODEC)
/* Sony NWZ must use 32-bit per sample */
static const snd_pcm_format_t format = SND_PCM_FORMAT_S32_LE; /* sample format */
typedef long sample_t;
typedef int32_t sample_t;
#else
static const snd_pcm_format_t format = SND_PCM_FORMAT_S16; /* sample format */
typedef short sample_t;
typedef int16_t sample_t;
#endif
static const int channels = 2; /* count of channels */
static unsigned int real_sample_rate;
@ -260,10 +260,9 @@ error:
* and add 48dB to the input volume. We cannot go lower -43dB because several
* values between -48dB and -43dB would require a fractional multiplier, which is
* stupid to implement for such very low volume. */
static int dig_vol_mult_l = 2 ^ 16; /* multiplicative factor to apply to each sample */
static int dig_vol_mult_r = 2 ^ 16; /* multiplicative factor to apply to each sample */
void pcm_alsa_set_digital_volume(int vol_db_l, int vol_db_r)
static int dig_vol_mult_l = 2 << 16; /* multiplicative factor to apply to each sample */
static int dig_vol_mult_r = 2 << 16; /* multiplicative factor to apply to each sample */
void pcm_set_mixer_volume(int vol_db_l, int vol_db_r)
{
if(vol_db_l > 0 || vol_db_r > 0 || vol_db_l < -43 || vol_db_r < -43)
panicf("invalid pcm alsa volume");
@ -336,7 +335,7 @@ static bool copy_frames(bool first)
{
/* 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 short *pcm_ptr = pcm_data;
const int16_t *pcm_ptr = pcm_data;
sample_t *sample_ptr = &frames[2*(period_size-frames_left)];
for (int i = 0; i < nframes; i++)
{
@ -757,11 +756,6 @@ void pcm_play_dma_postinit(void)
#endif
}
void pcm_set_mixer_volume(int volume)
{
(void)volume;
}
int pcm_alsa_get_rate(void)
{
return real_sample_rate;