[igt-dev] [PATCH i-g-t 1/5] lib/igt_alsa: support all alsa formats
Ser, Simon
simon.ser at intel.com
Mon May 20 10:38:34 UTC 2019
On Mon, 2019-05-20 at 13:11 +0300, Martin Peres wrote:
> On 17/05/2019 19:02, Simon Ser wrote:
> > This commit adds a new format argument to alsa_configure_output. This allows
> > the caller to decide the format used for the audio signal.
> >
> > This removes the assumption that samples are S16_LE items in igt_alsa. The
> > alsa function snd_pcm_format_physical_width is used instead of sizeof(short).
> > This also removes the theorically incorrect assumption that sizeof(short) ==
> > sizeof(int16_t).
> >
> > Right now S16_LE is hardcoded in the test. Tests with other formats will be
> > added in future commits.
> >
> > Signed-off-by: Simon Ser <simon.ser at intel.com>
> > ---
> > lib/igt_alsa.c | 23 +++++++++++++----------
> > lib/igt_alsa.h | 7 ++++---
> > tests/kms_chamelium.c | 7 ++++---
> > 3 files changed, 21 insertions(+), 16 deletions(-)
> >
> > diff --git a/lib/igt_alsa.c b/lib/igt_alsa.c
> > index 28a866e0ee74..2dfb8ccd3ad9 100644
> > --- a/lib/igt_alsa.c
> > +++ b/lib/igt_alsa.c
> > @@ -27,7 +27,6 @@
> > #include "config.h"
> >
> > #include <limits.h>
> > -#include <alsa/asoundlib.h>
> >
> > #include "igt_alsa.h"
> > #include "igt_aux.h"
> > @@ -47,10 +46,11 @@
> > struct alsa {
> > snd_pcm_t *output_handles[HANDLES_MAX];
> > int output_handles_count;
> > + snd_pcm_format_t output_format;
> > int output_sampling_rate;
> > int output_channels;
> >
> > - int (*output_callback)(void *data, short *buffer, int samples);
> > + int (*output_callback)(void *data, void *buffer, int samples);
> > void *output_callback_data;
> > int output_samples_trigger;
> > };
> > @@ -342,8 +342,8 @@ bool alsa_test_output_configuration(struct alsa *alsa, int channels,
> > * Configure the output devices with the configuration specified by @channels
> > * and @sampling_rate.
> > */
> > -void alsa_configure_output(struct alsa *alsa, int channels,
> > - int sampling_rate)
> > +void alsa_configure_output(struct alsa *alsa, snd_pcm_format_t fmt,
> > + int channels, int sampling_rate)
> > {
> > snd_pcm_t *handle;
> > int ret;
> > @@ -354,13 +354,14 @@ void alsa_configure_output(struct alsa *alsa, int channels,
> > for (i = 0; i < alsa->output_handles_count; i++) {
> > handle = alsa->output_handles[i];
> >
> > - ret = snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE,
> > + ret = snd_pcm_set_params(handle, fmt,
> > SND_PCM_ACCESS_RW_INTERLEAVED,
> > channels, sampling_rate,
> > soft_resample, latency);
> > igt_assert(ret >= 0);
> > }
> >
> > + alsa->output_format = fmt;
> > alsa->output_channels = channels;
> > alsa->output_sampling_rate = sampling_rate;
> > }
> > @@ -379,7 +380,7 @@ void alsa_configure_output(struct alsa *alsa, int channels,
> > * for failure.
> > */
> > void alsa_register_output_callback(struct alsa *alsa,
> > - int (*callback)(void *data, short *buffer, int samples),
> > + int (*callback)(void *data, void *buffer, int samples),
> > void *callback_data, int samples_trigger)
> > {
> > alsa->output_callback = callback;
> > @@ -402,12 +403,13 @@ void alsa_register_output_callback(struct alsa *alsa,
> > int alsa_run(struct alsa *alsa, int duration_ms)
> > {
> > snd_pcm_t *handle;
> > - short *output_buffer = NULL;
> > + char *output_buffer = NULL;
>
> Why not void*?
Below in the snd_pcm_writei call, we need to add an offset to this
address. Since it's not possible to use the array syntax on void * I
used char *. That way output_buffer[i] is the i-th byte of
output_buffer.
(Note that doing pointer arithmetic on void * is undefined behaviour.)
> > int output_limit;
> > int output_total = 0;
> > int output_counts[alsa->output_handles_count];
> > bool output_ready = false;
> > int output_channels;
> > + int bytes_per_sample;
> > int output_trigger;
> > bool reached;
> > int index;
> > @@ -418,9 +420,10 @@ int alsa_run(struct alsa *alsa, int duration_ms)
> >
> > output_limit = alsa->output_sampling_rate * duration_ms / 1000;
> > output_channels = alsa->output_channels;
> > + bytes_per_sample = snd_pcm_format_physical_width(alsa->output_format) / 8;
> > output_trigger = alsa->output_samples_trigger;
> > - output_buffer = malloc(sizeof(short) * output_channels *
> > - output_trigger);
> > + output_buffer = malloc(output_channels * output_trigger *
> > + bytes_per_sample);
> >
> > do {
> > reached = true;
> > @@ -454,7 +457,7 @@ int alsa_run(struct alsa *alsa, int duration_ms)
> > count = avail < count ? avail : count;
> >
> > ret = snd_pcm_writei(handle,
> > - &output_buffer[index],
> > + &output_buffer[index * bytes_per_sample],
> > count);
> > if (ret < 0) {
> > ret = snd_pcm_recover(handle,
> > diff --git a/lib/igt_alsa.h b/lib/igt_alsa.h
> > index a10985ff777f..46b3568d26fd 100644
> > --- a/lib/igt_alsa.h
> > +++ b/lib/igt_alsa.h
> > @@ -29,6 +29,7 @@
> >
> > #include "config.h"
> >
> > +#include <alsa/asoundlib.h>
> > #include <stdbool.h>
> >
> > struct alsa;
> > @@ -39,10 +40,10 @@ int alsa_open_output(struct alsa *alsa, const char *device_name);
> > void alsa_close_output(struct alsa *alsa);
> > bool alsa_test_output_configuration(struct alsa *alsa, int channels,
> > int sampling_rate);
> > -void alsa_configure_output(struct alsa *alsa, int channels,
> > - int sampling_rate);
> > +void alsa_configure_output(struct alsa *alsa, snd_pcm_format_t fmt,
> > + int channels, int sampling_rate);
> > void alsa_register_output_callback(struct alsa *alsa,
> > - int (*callback)(void *data, short *buffer, int samples),
> > + int (*callback)(void *data, void *buffer, int samples),
> > void *callback_data, int samples_trigger);
> > int alsa_run(struct alsa *alsa, int duration_ms);
> >
> > diff --git a/tests/kms_chamelium.c b/tests/kms_chamelium.c
> > index c8b6b22d7b4a..2b465565418d 100644
> > --- a/tests/kms_chamelium.c
> > +++ b/tests/kms_chamelium.c
> > @@ -801,11 +801,11 @@ struct audio_state {
> > };
> >
> > static int
> > -audio_output_callback(void *data, short *buffer, int frames)
> > +audio_output_callback(void *data, void *buffer, int samples)
> > {
> > struct audio_state *state = data;
> >
> > - audio_signal_fill_s16_le(state->signal, buffer, frames);
> > + audio_signal_fill_s16_le(state->signal, buffer, samples);
> >
> > return state->run ? 0 : -1;
> > }
> > @@ -843,7 +843,8 @@ do_test_display_audio(data_t *data, struct chamelium_port *port,
> >
> > igt_debug("Testing with playback sampling rate %d Hz and %d channels\n",
> > playback_rate, playback_channels);
> > - alsa_configure_output(alsa, playback_channels, playback_rate);
> > + alsa_configure_output(alsa, SND_PCM_FORMAT_S16_LE,
> > + playback_channels, playback_rate);
> >
> > chamelium_start_capturing_audio(data->chamelium, port, false);
> >
> >
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