Commit 8f2afb98 authored by Rémi Denis-Courmont's avatar Rémi Denis-Courmont

ALSA: remove a few useless variables

Overwriting *fmt is safe nowadays.
parent 345a3e31
......@@ -318,10 +318,9 @@ static int Start (audio_output_t *aout, audio_sample_format_t *restrict fmt)
return VLC_ENOMEM;
snd_pcm_format_t pcm_format; /* ALSA sample format */
vlc_fourcc_t fourcc = fmt->i_format;
bool spdif = false;
switch (fourcc)
switch (fmt->i_format)
{
case VLC_CODEC_F64B:
pcm_format = SND_PCM_FORMAT_FLOAT64_BE;
......@@ -376,18 +375,18 @@ static int Start (audio_output_t *aout, audio_sample_format_t *restrict fmt)
spdif = var_InheritBool (aout, "spdif");
if (spdif)
{
fourcc = VLC_CODEC_SPDIFL;
fmt->i_format = VLC_CODEC_SPDIFL;
pcm_format = SND_PCM_FORMAT_S16;
}
else
if (HAVE_FPU)
{
fourcc = VLC_CODEC_FL32;
fmt->i_format = VLC_CODEC_FL32;
pcm_format = SND_PCM_FORMAT_FLOAT;
}
else
{
fourcc = VLC_CODEC_S16N;
fmt->i_format = VLC_CODEC_S16N;
pcm_format = SND_PCM_FORMAT_S16;
}
}
......@@ -473,19 +472,19 @@ static int Start (audio_output_t *aout, audio_sample_format_t *restrict fmt)
else
if (snd_pcm_hw_params_test_format (pcm, hw, SND_PCM_FORMAT_FLOAT) == 0)
{
fourcc = VLC_CODEC_FL32;
fmt->i_format = VLC_CODEC_FL32;
pcm_format = SND_PCM_FORMAT_FLOAT;
}
else
if (snd_pcm_hw_params_test_format (pcm, hw, SND_PCM_FORMAT_S32) == 0)
{
fourcc = VLC_CODEC_S32N;
fmt->i_format = VLC_CODEC_S32N;
pcm_format = SND_PCM_FORMAT_S32;
}
else
if (snd_pcm_hw_params_test_format (pcm, hw, SND_PCM_FORMAT_S16) == 0)
{
fourcc = VLC_CODEC_S16N;
fmt->i_format = VLC_CODEC_S16N;
pcm_format = SND_PCM_FORMAT_S16;
}
else
......@@ -526,15 +525,13 @@ static int Start (audio_output_t *aout, audio_sample_format_t *restrict fmt)
}
/* Set sample rate */
unsigned rate = fmt->i_rate;
val = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, NULL);
val = snd_pcm_hw_params_set_rate_near (pcm, hw, &fmt->i_rate, NULL);
if (val)
{
msg_Err (aout, "cannot set sample rate: %s", snd_strerror (val));
goto error;
}
if (fmt->i_rate != rate)
msg_Dbg (aout, "resampling from %d Hz to %d Hz", fmt->i_rate, rate);
sys->rate = fmt->i_rate;
/* Set buffer size */
param = AOUT_MAX_ADVANCE_TIME;
......@@ -610,9 +607,6 @@ static int Start (audio_output_t *aout, audio_sample_format_t *restrict fmt)
}
/* Setup audio_output_t */
fmt->i_format = fourcc;
fmt->i_rate = rate;
sys->rate = rate;
if (spdif)
{
fmt->i_bytes_per_frame = AOUT_SPDIF_SIZE;
......
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