[pulseaudio-discuss] lots of retry on failure of hw_params interface
Hi community When doing development for Audio driver, I hit an interesting behaviour of pulseaudio Due to some miss configurations in devicetree, my CPU DAI driver always fail on hw_params() interface, when pulseaudio starts to configure driver for the first time, due to the problem in Driver configuration, it fails on hw_params interface. But instead of just exit from execution, (or just retry a few times) pulseaudio continues to retry SNDRV_PCM_IOCTL_HW_PARAMS ioctl call, with the following log message: ... ... I: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Invalid argument I: [pulseaudio] alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument I: [pulseaudio] (alsa-lib)conf.c: Unknown parameters 0 I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM front:0 I: [pulseaudio] alsa-util.c: Error opening PCM device front:0: Invalid argument I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Invalid argument I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Invalid argument I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) ... ... as pulseaudio continues its retry of hw_params, it stops the whole system, which doesn't make sense for me. my question is: is this behaviour default for pulseaudio? can I change the retry number (for example, I want pulseaudio only retry a few times), or can I disable the retry mechanism at all? Thanks, Jiada ___ pulseaudio-discuss mailing list pulseaudio-discuss@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] [PATCH] module-virtual-surround-sink: Move normalization heuristic to its own function
This patch also adds a description how the heuristic works and mentions that
there is a scaling factor that can be adjusted if there is audible clipping.
---
src/modules/module-virtual-surround-sink.c | 57 +-
1 file changed, 40 insertions(+), 17 deletions(-)
diff --git a/src/modules/module-virtual-surround-sink.c
b/src/modules/module-virtual-surround-sink.c
index 1d6cfc6..34f23fd 100644
--- a/src/modules/module-virtual-surround-sink.c
+++ b/src/modules/module-virtual-surround-sink.c
@@ -519,6 +519,45 @@ static pa_channel_position_t
mirror_channel(pa_channel_position_t channel) {
}
}
+static void normalize_hrir(struct userdata *u) {
+/* normalize hrir to avoid audible clipping
+ *
+ * The following heuristic tries to avoid audible clipping. It cannot avoid
+ * clipping in the worst case though, because the scaling factor would
+ * become too large resulting in a too quiet signal.
+ * The idea of the heuristic is to avoid clipping when a single click is
+ * played back on all channels. The scaling factor describes the additional
+ * factor that is necessary to avoid clipping for normal signals.
+ *
+ * This algorithm doesn't pretend to be perfect, it's just something that
+ * appears to work (not too quiet, no audible clipping) on the material
that
+ * it has been tested on. If you find a real-world example where this
+ * algorithm results in audible clipping, please write a patch that adjusts
+ * the scaling factor constants or improves the algorithm (or if you can't
+ * write a patch, at least report the problem to the PulseAudio mailing
list
+ * or bug tracker). */
+
+const float scaling_factor = 2.5;
+
+float hrir_sum, hrir_max;
+unsigned i, j;
+
+hrir_max = 0;
+for (i = 0; i u-hrir_samples; i++) {
+hrir_sum = 0;
+for (j = 0; j u-hrir_channels; j++)
+hrir_sum += fabs(u-hrir_data[i * u-hrir_channels + j]);
+
+if (hrir_sum hrir_max)
+hrir_max = hrir_sum;
+}
+
+for (i = 0; i u-hrir_samples; i++) {
+for (j = 0; j u-hrir_channels; j++)
+u-hrir_data[i * u-hrir_channels + j] /= hrir_max *
scaling_factor;
+}
+}
+
int pa__init(pa_module*m) {
struct userdata *u;
pa_sample_spec ss, sink_input_ss;
@@ -533,7 +572,6 @@ int pa__init(pa_module*m) {
const char *hrir_file;
unsigned i, j, found_channel_left, found_channel_right;
-float hrir_sum, hrir_max;
float *hrir_data;
pa_sample_spec hrir_ss;
@@ -758,22 +796,7 @@ int pa__init(pa_module*m) {
goto fail;
}
-/* normalize hrir to avoid clipping */
-hrir_max = 0;
-for (i = 0; i u-hrir_samples; i++) {
-hrir_sum = 0;
-for (j = 0; j u-hrir_channels; j++)
-hrir_sum += fabs(u-hrir_data[i * u-hrir_channels + j]);
-
-if (hrir_sum hrir_max)
-hrir_max = hrir_sum;
-}
-if (hrir_max 1) {
-for (i = 0; i u-hrir_samples; i++) {
-for (j = 0; j u-hrir_channels; j++)
-u-hrir_data[i * u-hrir_channels + j] /= hrir_max * 1.2;
-}
-}
+normalize_hrir(u);
/* create mapping between hrir and input */
u-mapping_left = (unsigned *) pa_xnew0(unsigned, u-channels);
--
1.9.1
___
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[pulseaudio-discuss] [PATCH] resampler: Support speex resampler compiled with FIXED_POINT
When Speex is compiled with FIXED_POINT defined, it scales float input
to +/-32768 instead of +/-1. In order to make floating point resampler
work with speex compiled with FIXED_POINT, we need to rescale the input
to speex.
Unfortunately, there is no easy way to tell how speex has been built,
so we probe at runtime.
A patch to fix speex was rejected saying that speex API is stable.
Further discussion is here
http://lists.openembedded.org/pipermail/openembedded-core/2014-January/087886.html
Signed-off-by: Peter Meerwald pme...@pmeerw.net
Reported-by: Fahad Arslan fahad_ars...@mentor.com
Cc: Damir Jelić poljari...@gmail.com
---
src/pulsecore/resampler.c | 81 ++-
1 file changed, 80 insertions(+), 1 deletion(-)
diff --git a/src/pulsecore/resampler.c b/src/pulsecore/resampler.c
index 38389f3..4a9c1f5 100644
--- a/src/pulsecore/resampler.c
+++ b/src/pulsecore/resampler.c
@@ -39,6 +39,7 @@
#include pulsecore/macro.h
#include pulsecore/strbuf.h
#include pulsecore/remap.h
+#include pulsecore/once.h
#include pulsecore/core-util.h
#include ffmpeg/avcodec.h
@@ -1490,6 +1491,82 @@ static unsigned speex_resample_float(pa_resampler *r,
const pa_memchunk *input,
return 0;
}
+/* float resampler when speexdsp is built with FIXED_POINT */
+static unsigned speex_resample_float_fixed(pa_resampler *r, const pa_memchunk
*input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
+pa_memblock *b;
+float *in, *out, *t;
+uint32_t inf = in_n_frames, outf = *out_n_frames, i;
+unsigned nsamples = r-work_channels * in_n_frames;
+SpeexResamplerState *state;
+
+pa_assert(r);
+pa_assert(input);
+pa_assert(output);
+pa_assert(out_n_frames);
+
+state = r-impl.data;
+
+b = pa_memblock_new(r-mempool, nsamples * sizeof(float));
+t = pa_memblock_acquire(b);
+in = pa_memblock_acquire_chunk(input);
+
+/* speexdsp float API sample range is [-32768,32768] instead of [-1,1]
+ * when compiled with FIXED_POINT, rescale input before passing to speex.
*/
+for (i = 0; i nsamples; i++)
+t[i] = 32768.0f * in[i];
+
+pa_memblock_release(input-memblock);
+
+out = pa_memblock_acquire_chunk(output);
+
+pa_assert_se(speex_resampler_process_interleaved_float(state, t, inf,
out, outf) == 0);
+
+pa_memblock_release(b);
+pa_memblock_unref(b);
+
+pa_memblock_release(output-memblock);
+
+pa_assert(inf == in_n_frames);
+*out_n_frames = outf;
+
+return 0;
+}
+
+/* speexdsp has a nasty issue: when built with FIXED_POINT, the float resampler
+ * expects samples in [-32768,32768] instead of [-1,1]; there is no easy way
+ * to figure out how speexdsp has been compiled. */
+
+static unsigned (*speex_resampler_float)(pa_resampler *r, const pa_memchunk
*input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) =
speex_resample_float;
+
+static void speex_probe_fixed_point() {
+PA_ONCE_BEGIN {
+float f_out = -1.0f, f_in = 1.0f;
+spx_uint32_t in_len = 1, out_len = 1;
+
+SpeexResamplerState *s = speex_resampler_init(1, 1, 1,
+SPEEX_RESAMPLER_QUALITY_MIN, NULL);
+if (!s)
+goto done;
+
+/* feed one sample */
+if (speex_resampler_process_float(s, 0, f_in, in_len,
+f_out, out_len) != RESAMPLER_ERR_SUCCESS)
+goto done;
+
+/* expecting sample has been processed, one sample output */
+if (in_len != 1 out_len != 1)
+goto done;
+
+/* FIXED_POINT implementation will output 0.0 */
+if (fabsf(f_out) 0.1f)
+speex_resampler_float = speex_resample_float_fixed;
+
+done:
+if (s)
+speex_resampler_destroy(s);
+} PA_ONCE_END;
+}
+
static unsigned speex_resample_int(pa_resampler *r, const pa_memchunk *input,
unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
int16_t *in, *out;
uint32_t inf = in_n_frames, outf = *out_n_frames;
@@ -1563,8 +1640,10 @@ static int speex_init(pa_resampler *r) {
} else {
pa_assert(r-method = PA_RESAMPLER_SPEEX_FLOAT_BASE r-method =
PA_RESAMPLER_SPEEX_FLOAT_MAX);
+speex_probe_fixed_point();
+
q = r-method - PA_RESAMPLER_SPEEX_FLOAT_BASE;
-r-impl.resample = speex_resample_float;
+r-impl.resample = speex_resampler_float;
}
pa_log_info(Choosing speex quality setting %i., q);
--
1.9.1
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[pulseaudio-discuss] PulseAudio Python Bindings
Hi PulseAudio developers and Python fans, I am the author of PulseAudio Python Bindings https://github.com/linuxdeepin/pypulseaudio And right now it is public repos, if someone want to maintain it, please let me know, thanks a lot! Regards, Leslie Zhai ___ pulseaudio-discuss mailing list pulseaudio-discuss@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
