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hosted pcm-alsa improvements
* xduoo x3ii/x20: Better line out support * less granular volume settings (too many steps before) * Better handling of swiching sample rates * Log actual sample rate in debug menu Most credit goes to Roman Stolyarov Additional integration [re]work by myself Change-Id: I63af3740678cf2ed3170f61534e1029c81826bb6
This commit is contained in:
Solomon Peachy
parent
6459fa0765
commit
e43726df2c
15 changed files with 218 additions and 91 deletions
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@ -68,7 +68,7 @@
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* with multple applications running */
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static char device[] = "plughw:0,0"; /* playback device */
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static const snd_pcm_access_t access_ = SND_PCM_ACCESS_RW_INTERLEAVED; /* access mode */
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#ifdef SONY_NWZ_LINUX
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#if defined(SONY_NWZ_LINUX) || defined(HAVE_FIIO_LINUX_CODEC)
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/* Sony NWZ must use 32-bit per sample */
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static const snd_pcm_format_t format = SND_PCM_FORMAT_S32_LE; /* sample format */
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typedef long sample_t;
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@ -77,6 +77,9 @@ static const snd_pcm_format_t format = SND_PCM_FORMAT_S16; /* sample format *
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typedef short sample_t;
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#endif
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static const int channels = 2; /* count of channels */
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static unsigned int sample_rate = 0;
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static unsigned int real_sample_rate = 0;
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static snd_pcm_t *handle = NULL;
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static snd_pcm_sframes_t buffer_size = MIX_FRAME_SAMPLES * 32; /* ~16k */
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static snd_pcm_sframes_t period_size = MIX_FRAME_SAMPLES * 4; /* ~4k */
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@ -93,14 +96,13 @@ static char signal_stack[SIGSTKSZ];
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static int recursion;
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#endif
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static int set_hwparams(snd_pcm_t *handle, unsigned sample_rate)
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static int set_hwparams(snd_pcm_t *handle)
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{
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unsigned int rrate;
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int err;
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unsigned int srate;
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snd_pcm_hw_params_t *params;
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snd_pcm_hw_params_malloc(¶ms);
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/* choose all parameters */
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err = snd_pcm_hw_params_any(handle, params);
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if (err < 0)
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@ -130,16 +132,17 @@ static int set_hwparams(snd_pcm_t *handle, unsigned sample_rate)
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goto error;
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}
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/* set the stream rate */
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rrate = sample_rate;
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err = snd_pcm_hw_params_set_rate_near(handle, params, &rrate, 0);
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sample_rate = srate = pcm_sampr;
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err = snd_pcm_hw_params_set_rate_near(handle, params, &srate, 0);
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if (err < 0)
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{
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printf("Rate %iHz not available for playback: %s\n", sample_rate, snd_strerror(err));
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goto error;
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}
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if (rrate != sample_rate)
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real_sample_rate = srate;
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if (real_sample_rate != sample_rate)
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{
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printf("Rate doesn't match (requested %iHz, get %iHz)\n", sample_rate, rrate);
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printf("Rate doesn't match (requested %iHz, get %iHz)\n", sample_rate, real_sample_rate);
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err = -EINVAL;
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goto error;
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}
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@ -159,8 +162,9 @@ static int set_hwparams(snd_pcm_t *handle, unsigned sample_rate)
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printf("Unable to set period size %ld for playback: %s\n", period_size, snd_strerror(err));
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goto error;
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}
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if (!frames)
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frames = malloc(period_size * channels * sizeof(sample_t));
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free(frames);
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frames = calloc(1, period_size * channels * sizeof(sample_t));
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/* write the parameters to device */
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err = snd_pcm_hw_params(handle, params);
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@ -229,26 +233,37 @@ error:
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* and add 48dB to the input volume. We cannot go lower -43dB because several
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* values between -48dB and -43dB would require a fractional multiplier, which is
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* stupid to implement for such very low volume. */
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static int dig_vol_mult = 2 ^ 16; /* multiplicative factor to apply to each sample */
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static int dig_vol_mult_l = 2 ^ 16; /* multiplicative factor to apply to each sample */
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static int dig_vol_mult_r = 2 ^ 16; /* multiplicative factor to apply to each sample */
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void pcm_alsa_set_digital_volume(int vol_db)
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void pcm_alsa_set_digital_volume(int vol_db_l, int vol_db_r)
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{
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if(vol_db > 0 || vol_db < -43)
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if(vol_db_l > 0 || vol_db_r > 0 || vol_db_l < -43 || vol_db_r < -43)
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panicf("invalid pcm alsa volume");
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if(format != SND_PCM_FORMAT_S32_LE)
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panicf("this function assumes 32-bit sample size");
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vol_db += 48; /* -42dB .. 0dB => 5dB .. 48dB */
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vol_db_l += 48; /* -42dB .. 0dB => 5dB .. 48dB */
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vol_db_r += 48; /* -42dB .. 0dB => 5dB .. 48dB */
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/* NOTE if vol_dB = 5 then vol_shift = 1 but r = 1 so we do vol_shift - 1 >= 0
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* otherwise vol_dB >= 0 implies vol_shift >= 2 so vol_shift - 2 >= 0 */
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int vol_shift = vol_db / 3;
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int r = vol_db % 3;
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if(r == 0)
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dig_vol_mult = 1 << vol_shift;
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else if(r == 1)
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dig_vol_mult = 1 << vol_shift | 1 << (vol_shift - 2);
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int vol_shift_l = vol_db_l / 3;
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int vol_shift_r = vol_db_r / 3;
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int r_l = vol_db_l % 3;
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int r_r = vol_db_r % 3;
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if(r_l == 0)
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dig_vol_mult_l = 1 << vol_shift_l;
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else if(r_l == 1)
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dig_vol_mult_l = 1 << vol_shift_l | 1 << (vol_shift_l - 2);
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else
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dig_vol_mult = 1 << vol_shift | 1 << (vol_shift - 1);
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printf("%d dB -> factor = %d\n", vol_db - 48, dig_vol_mult);
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dig_vol_mult_l = 1 << vol_shift_l | 1 << (vol_shift_l - 1);
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printf("l: %d dB -> factor = %d\n", vol_db_l - 48, dig_vol_mult_l);
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if(r_r == 0)
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dig_vol_mult_r = 1 << vol_shift_r;
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else if(r_r == 1)
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dig_vol_mult_r = 1 << vol_shift_r | 1 << (vol_shift_r - 2);
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else
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dig_vol_mult_r = 1 << vol_shift_r | 1 << (vol_shift_r - 1);
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printf("r: %d dB -> factor = %d\n", vol_db_r - 48, dig_vol_mult_r);
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}
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/* copy pcm samples to a spare buffer, suitable for snd_pcm_writei() */
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@ -279,8 +294,11 @@ static bool fill_frames(void)
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* sample by some value so the sound is not too low */
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const short *pcm_ptr = pcm_data;
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sample_t *sample_ptr = &frames[2*(period_size-frames_left)];
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for (int i = 0; i < copy_n*2; i++)
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*sample_ptr++ = *pcm_ptr++ * dig_vol_mult;
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for (int i = 0; i < copy_n; i++)
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{
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*sample_ptr++ = *pcm_ptr++ * dig_vol_mult_l;
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*sample_ptr++ = *pcm_ptr++ * dig_vol_mult_r;
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}
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}
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else
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{
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@ -378,7 +396,7 @@ static int async_rw(snd_pcm_t *handle)
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/* fill buffer with silence to initiate playback without noisy click */
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sample_size = buffer_size;
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samples = malloc(sample_size * channels * sizeof(sample_t));
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samples = calloc(1, sample_size * channels * sizeof(sample_t));
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snd_pcm_format_set_silence(format, samples, sample_size);
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err = snd_pcm_writei(handle, samples, sample_size);
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@ -428,7 +446,7 @@ void pcm_play_dma_init(void)
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if ((err = snd_pcm_nonblock(handle, 1)))
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panicf("Could not set non-block mode: %s\n", snd_strerror(err));
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if ((err = set_hwparams(handle, pcm_sampr)) < 0)
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if ((err = set_hwparams(handle)) < 0)
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{
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panicf("Setting of hwparams failed: %s\n", snd_strerror(err));
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}
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@ -473,15 +491,28 @@ void pcm_play_unlock(void)
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#endif
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}
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#if defined(HAVE_XDUOO_LINUX_CODEC) || defined(HAVE_FIIO_LINUX_CODEC) || defined(HAVE_ROCKER_CODEC)
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static void pcm_dma_apply_settings_nolock(void)
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{
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if (sample_rate != pcm_sampr)
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{
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audiohw_mute(true);
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snd_pcm_drop(handle);
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set_hwparams(handle);
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audiohw_mute(false);
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}
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}
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#else
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static void pcm_dma_apply_settings_nolock(void)
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{
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snd_pcm_drop(handle);
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set_hwparams(handle, pcm_sampr);
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set_hwparams(handle);
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#if defined(HAVE_NWZ_LINUX_CODEC)
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/* Sony NWZ linux driver uses a nonstandard mecanism to set the sampling rate */
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audiohw_set_frequency(pcm_sampr);
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#endif
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}
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#endif
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void pcm_dma_apply_settings(void)
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{
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@ -571,11 +602,16 @@ void pcm_play_dma_postinit(void)
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audiohw_postinit();
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}
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void pcm_set_mixer_volume(int volume)
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{
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(void)volume;
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}
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int pcm_alsa_get_rate(void)
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{
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return real_sample_rate;
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}
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#ifdef HAVE_RECORDING
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void pcm_rec_lock(void)
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{
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