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| author | Max Kellermann <max@duempel.org> | 2012-03-27 01:05:33 +0200 |
|---|---|---|
| committer | Max Kellermann <max@duempel.org> | 2012-03-27 01:22:50 +0200 |
| commit | 2803ec2e96096dfc461f3e9a522e27b23453d076 (patch) | |
| tree | b622b0fdc5450e8b3c6d73e93dfc04ab7816b021 | |
| parent | ddd4f675a2d42b0c8156e03bf3e93f39df2fe609 (diff) | |
| download | mpd-2803ec2e96096dfc461f3e9a522e27b23453d076.tar.gz mpd-2803ec2e96096dfc461f3e9a522e27b23453d076.tar.xz mpd-2803ec2e96096dfc461f3e9a522e27b23453d076.zip | |
output/alsa: support 32 bit DSD-over-USB
| -rw-r--r-- | src/output/alsa_output_plugin.c | 19 |
1 files changed, 15 insertions, 4 deletions
diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c index c9dd1ee1b..9faacb0a9 100644 --- a/src/output/alsa_output_plugin.c +++ b/src/output/alsa_output_plugin.c @@ -588,7 +588,8 @@ error: static bool alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format, - bool *packed_r, bool *reverse_endian_r, GError **error_r) + bool *shift8_r, bool *packed_r, bool *reverse_endian_r, + GError **error_r) { assert(ad->dsd_usb); assert(audio_format->format == SAMPLE_FORMAT_DSD); @@ -604,6 +605,15 @@ alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format, if (!alsa_setup(ad, &usb_format, packed_r, reverse_endian_r, error_r)) return false; + /* if the device allows only 32 bit, shift all DSD-over-USB + samples left by 8 bit and leave the lower 8 bit cleared; + the DSD-over-USB documentation does not specify whether + this is legal, but there is anecdotical evidence that this + is possible (and the only option for some devices) */ + *shift8_r = usb_format.format == SAMPLE_FORMAT_S32; + if (usb_format.format == SAMPLE_FORMAT_S32) + usb_format.format = SAMPLE_FORMAT_S24_P32; + if (!audio_format_equals(&usb_format, &check)) { /* no bit-perfect playback, which is required for DSD over USB */ @@ -620,12 +630,13 @@ static bool alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format, GError **error_r) { - bool packed, reverse_endian; + bool shift8 = false, packed, reverse_endian; const bool dsd_usb = ad->dsd_usb && audio_format->format == SAMPLE_FORMAT_DSD; const bool success = dsd_usb - ? alsa_setup_dsd(ad, audio_format, &packed, &reverse_endian, + ? alsa_setup_dsd(ad, audio_format, + &shift8, &packed, &reverse_endian, error_r) : alsa_setup(ad, audio_format, &packed, &reverse_endian, error_r); @@ -634,7 +645,7 @@ alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format, pcm_export_open(&ad->export, audio_format->format, audio_format->channels, - dsd_usb, false, packed, reverse_endian); + dsd_usb, shift8, packed, reverse_endian); return true; } |