[alsa-devel] [PATCH] firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio data
According to AM824 in IEC 61883-6:2002, 2 bits in LSB of label for Raw Audio data means Valid Length Code (VBL). Ths value is: - b00 for 24 bits sample (label is 0x40) - b01 for 20 bits sample (label is 0x41) - b10 for 16 bits sample (label is 0x42)
But current firewire-lib apply 24 bits label for both of 16/24 bits samples.
As long as developers investigate BeBoB/Fireworks/OXFW/Dice, all of them have a behaviour to ignore the label. They can generate correct sound even if firewire-lib gives wrong label (i.e. 0xff). On BeBoB, this is not only for Raw Audio data channel, but also for IEC 60958 Conformant data channel.
So there is little possibility of regression.
Acked-by: Clemens Ladisch clemens@ladisch.de Signed-off-by: Takashi Sakamoto o-takashi@sakamocchi.jp --- sound/firewire/amdtp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 31dd1cf..f96bf4c 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -418,7 +418,7 @@ static void amdtp_write_s16(struct amdtp_stream *s, for (i = 0; i < frames; ++i) { for (c = 0; c < channels; ++c) { buffer[s->pcm_positions[c]] = - cpu_to_be32((*src << 8) | 0x40000000); + cpu_to_be32((*src << 8) | 0x42000000); src++; } buffer += s->data_block_quadlets;
At Mon, 2 Jun 2014 01:50:16 +0900, Takashi Sakamoto wrote:
According to AM824 in IEC 61883-6:2002, 2 bits in LSB of label for Raw Audio data means Valid Length Code (VBL). Ths value is:
- b00 for 24 bits sample (label is 0x40)
- b01 for 20 bits sample (label is 0x41)
- b10 for 16 bits sample (label is 0x42)
But current firewire-lib apply 24 bits label for both of 16/24 bits samples.
As long as developers investigate BeBoB/Fireworks/OXFW/Dice, all of them have a behaviour to ignore the label. They can generate correct sound even if firewire-lib gives wrong label (i.e. 0xff). On BeBoB, this is not only for Raw Audio data channel, but also for IEC 60958 Conformant data channel.
So there is little possibility of regression.
Acked-by: Clemens Ladisch clemens@ladisch.de Signed-off-by: Takashi Sakamoto o-takashi@sakamocchi.jp
Thanks, applied.
Takashi
sound/firewire/amdtp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 31dd1cf..f96bf4c 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -418,7 +418,7 @@ static void amdtp_write_s16(struct amdtp_stream *s, for (i = 0; i < frames; ++i) { for (c = 0; c < channels; ++c) { buffer[s->pcm_positions[c]] =
cpu_to_be32((*src << 8) | 0x40000000);
} buffer += s->data_block_quadlets;cpu_to_be32((*src << 8) | 0x42000000); src++;
-- 1.8.3.2
participants (2)
-
Takashi Iwai
-
Takashi Sakamoto