[igt-dev] [PATCH i-g-t 1/5] lib/igt_alsa: support all alsa formats
Martin Peres
martin.peres at intel.com
Mon May 20 10:11:09 UTC 2019
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*?
> 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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