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[alsa] Fix playback of 24 bit audio (issue #830)

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The player will write 24 bit samples using 3 bytes, not 4, so the appropriate
sample format is SND_PCM_FORMAT_S24_3LE, not SND_PCM_FORMAT_S24_LE.

For extra protection we also use snd_pcm_bytes_to_frames() instead of BTOS(),
because that way we can be more certain that the buffer is not too short for
snd_pcm_writei().
This commit is contained in:
ejurgensen 2019-10-22 19:41:48 +02:00
parent 41f51d43ac
commit 85ab7c3057
1 changed files with 12 additions and 9 deletions
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21
src/outputs/alsa.c
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@ -179,7 +179,7 @@ bps2format(int bits_per_sample)
if (bits_per_sample == 16)
return SND_PCM_FORMAT_S16_LE;
else if (bits_per_sample == 24)
return SND_PCM_FORMAT_S24_LE;
return SND_PCM_FORMAT_S24_3LE;
else if (bits_per_sample == 32)
return SND_PCM_FORMAT_S32_LE;
else
@ -631,7 +631,7 @@ static int
buffer_write(struct alsa_playback_session *pb, struct output_data *odata, snd_pcm_sframes_t avail)
{
uint8_t *buf;
size_t bufsize;
ssize_t bufsize;
size_t wrote;
snd_pcm_sframes_t nsamp;
snd_pcm_sframes_t ret;
@ -643,7 +643,7 @@ buffer_write(struct alsa_playback_session *pb, struct output_data *odata, snd_pc
if (wrote < odata->bufsize)
DPRINTF(E_WARN, L_LAUDIO, "Bug! Partial prebuf write %zu/%zu\n", wrote, odata->bufsize);
nsamp = BTOS(wrote, pb->quality.bits_per_sample, pb->quality.channels);
nsamp = snd_pcm_bytes_to_frames(pb->pcm, wrote);
return nsamp;
}
@ -651,7 +651,7 @@ buffer_write(struct alsa_playback_session *pb, struct output_data *odata, snd_pc
if (pb->prebuf.read_avail != 0)
{
// Maximum amount of bytes we want to read
bufsize = STOB(avail, pb->quality.bits_per_sample, pb->quality.channels);
bufsize = snd_pcm_frames_to_bytes(pb->pcm, avail);
bufsize = ringbuffer_read(&buf, bufsize, &pb->prebuf);
if (bufsize == 0)
@ -659,7 +659,8 @@ buffer_write(struct alsa_playback_session *pb, struct output_data *odata, snd_pc
// DPRINTF(E_DBG, L_LAUDIO, "Writing prebuffer (read_avail=%zu, bufsize=%zu, avail=%li)\n", pb->prebuf.read_avail, bufsize, avail);
nsamp = BTOS(bufsize, pb->quality.bits_per_sample, pb->quality.channels);
nsamp = snd_pcm_bytes_to_frames(pb->pcm, bufsize);
ret = snd_pcm_writei(pb->pcm, buf, nsamp);
if (ret < 0)
return ret;
@ -680,7 +681,9 @@ buffer_write(struct alsa_playback_session *pb, struct output_data *odata, snd_pc
return odata->samples;
}
ret = snd_pcm_writei(pb->pcm, odata->buffer, odata->samples);
nsamp = snd_pcm_bytes_to_frames(pb->pcm, odata->bufsize);
ret = snd_pcm_writei(pb->pcm, odata->buffer, nsamp);
if (ret < 0)
return ret;
@ -808,7 +811,7 @@ static int
playback_drain(struct alsa_playback_session *pb)
{
uint8_t *buf;
size_t bufsize;
ssize_t bufsize;
snd_pcm_state_t state;
snd_pcm_sframes_t avail;
snd_pcm_sframes_t delay;
@ -836,7 +839,7 @@ playback_drain(struct alsa_playback_session *pb)
}
// Maximum amount of bytes we want to read
bufsize = STOB(avail, pb->quality.bits_per_sample, pb->quality.channels);
bufsize = snd_pcm_frames_to_bytes(pb->pcm, avail);
bufsize = ringbuffer_read(&buf, bufsize, &pb->prebuf);
if (bufsize == 0)
@ -844,7 +847,7 @@ playback_drain(struct alsa_playback_session *pb)
// DPRINTF(E_DBG, L_LAUDIO, "Draining prebuffer (read_avail=%zu, bufsize=%zu, avail=%li)\n", pb->prebuf.read_avail / 4, bufsize, avail);
nsamp = BTOS(bufsize, pb->quality.bits_per_sample, pb->quality.channels);
nsamp = snd_pcm_bytes_to_frames(pb->pcm, bufsize);
ret = snd_pcm_writei(pb->pcm, buf, nsamp);