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audio: remove swapped-endian spdif formats 

IEC 61937 frames should always be little endian (little endian 16 bit
words). I don't see any apparent need why the audio chain should handle
swapped-endian formats.

It could be that some audio outputs might want them (especially on big
endian architectures). On the other hand, it's not clear how that works
on these architectures, and it's not even known whether the current code
works on big endian at all. If something should break, and it should
turn out that swapped-endian spdif is needed on any platform/AO,
swapping still could be done in-place within the affected AO, and
there's no need for the additional complexity in the rest of the player.

Note that af_lavcac3enc outputs big endian spdif frames for unknown
reasons. Normally, the resulting data is just pulled through an auto-
inserted conversion filter and turned into little endian. Maybe this was
done as a trick so that the code didn't have to byte-swap the actual
audio frame. In any case, just make it output little endian frames.

All of this is untested, because I have no receiver hardware.
This commit is contained in:
wm4 2014-09-23 19:34:14 +02:00
parent 1f4a74cbed
commit 5b5a3d0c46
7 changed files with 29 additions and 37 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

12
audio/decode/ad_spdif.c
View File

@ -117,13 +117,13 @@ static int init(struct dec_audio *da, const char *decoder)
switch (stream->codec->codec_id) {
case AV_CODEC_ID_AAC:
spdif_ctx->iec61937_packet_size = 16384;
sample_format = AF_FORMAT_IEC61937_LE;
sample_format = AF_FORMAT_IEC61937;
samplerate = 48000;
num_channels = 2;
break;
case AV_CODEC_ID_AC3:
spdif_ctx->iec61937_packet_size = 6144;
sample_format = AF_FORMAT_AC3_LE;
sample_format = AF_FORMAT_AC3;
samplerate = 48000;
num_channels = 2;
break;
@ -131,19 +131,19 @@ static int init(struct dec_audio *da, const char *decoder)
if (da->opts->dtshd) {
av_dict_set(&format_opts, "dtshd_rate", "768000", 0); // 4*192000
spdif_ctx->iec61937_packet_size = 32768;
sample_format = AF_FORMAT_IEC61937_LE;
sample_format = AF_FORMAT_IEC61937;
samplerate = 192000;
num_channels = 2*4;
} else {
spdif_ctx->iec61937_packet_size = 32768;
sample_format = AF_FORMAT_AC3_LE;
sample_format = AF_FORMAT_AC3;
samplerate = 48000;
num_channels = 2;
}
break;
case AV_CODEC_ID_EAC3:
spdif_ctx->iec61937_packet_size = 24576;
sample_format = AF_FORMAT_IEC61937_LE;
sample_format = AF_FORMAT_IEC61937;
samplerate = 192000;
num_channels = 2;
break;
@ -155,7 +155,7 @@ static int init(struct dec_audio *da, const char *decoder)
break;
case AV_CODEC_ID_TRUEHD:
spdif_ctx->iec61937_packet_size = 61440;
sample_format = AF_FORMAT_IEC61937_LE;
sample_format = AF_FORMAT_IEC61937;
samplerate = 192000;
num_channels = 8;
break;

16
audio/filter/af_lavcac3enc.c
View File

@ -30,6 +30,7 @@
#include <libavutil/audioconvert.h>
#include <libavutil/intreadwrite.h>
#include <libavutil/common.h>
#include <libavutil/bswap.h>
#include <libavutil/mem.h>
#include "common/common.h"
@ -87,7 +88,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
if (in->nch > AC3_MAX_CHANNELS)
mp_audio_set_num_channels(in, AC3_MAX_CHANNELS);
mp_audio_set_format(af->data, AF_FORMAT_AC3_BE);
mp_audio_set_format(af->data, AF_FORMAT_AC3);
mp_audio_set_num_channels(af->data, 2);
if (!mp_audio_config_equals(in, &orig_in))
@ -150,6 +151,12 @@ static void uninit(struct af_instance* af)
}
}
static void swap_16(uint16_t *ptr, size_t size)
{
for (size_t n = 0; n < size; n++)
ptr[n] = av_bswap16(ptr[n]);
}
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* audio, int flags)
{
@ -223,11 +230,11 @@ static int filter(struct af_instance* af, struct mp_audio* audio, int flags)
frame_size = AC3_FRAME_SIZE * 2 * 2;
header_len = 8;
AV_WB16(hdr, 0xF872); // iec 61937 syncword 1
AV_WB16(hdr + 2, 0x4E1F); // iec 61937 syncword 2
AV_WL16(hdr, 0xF872); // iec 61937 syncword 1
AV_WL16(hdr + 2, 0x4E1F); // iec 61937 syncword 2
hdr[4] = bsmod; // bsmod
hdr[5] = 0x01; // data-type ac3
AV_WB16(hdr + 6, len << 3); // number of bits in payload
AV_WL16(hdr + 6, len << 3); // number of bits in payload
}
size_t max_size = (max_out_samples - out->samples) * out->sstride;
@ -239,6 +246,7 @@ static int filter(struct af_instance* af, struct mp_audio* audio, int flags)
memcpy(buf + header_len, s->pkt.data, s->pkt.size);
memset(buf + header_len + s->pkt.size, 0,
frame_size - (header_len + s->pkt.size));
swap_16((uint16_t *)(buf + header_len), s->pkt.size / 2);
out->samples += frame_size / out->sstride;
}

4
audio/format.c
View File

@ -119,8 +119,8 @@ bool af_fmt_is_planar(int format)
const struct af_fmt_entry af_fmtstr_table[] = {
FMT("mpeg2", AF_FORMAT_MPEG2)
FMT_ENDIAN("ac3", AF_FORMAT_AC3)
FMT_ENDIAN("iec61937", AF_FORMAT_IEC61937)
FMT("ac3", AF_FORMAT_AC3)
FMT("iec61937", AF_FORMAT_IEC61937)
FMT("u8", AF_FORMAT_U8)
FMT("s8", AF_FORMAT_S8)

13
audio/format.h
View File

@ -101,12 +101,8 @@ enum af_format {
AF_FORMAT_DOUBLE_LE = (AF_FORMAT_F|AF_FORMAT_64BIT|AF_FORMAT_LE),
AF_FORMAT_DOUBLE_BE = (AF_FORMAT_F|AF_FORMAT_64BIT|AF_FORMAT_BE),
AF_FORMAT_AC3_LE = (AF_FORMAT_S_AC3|AF_FORMAT_16BIT|AF_FORMAT_LE),
AF_FORMAT_AC3_BE = (AF_FORMAT_S_AC3|AF_FORMAT_16BIT|AF_FORMAT_BE),
AF_FORMAT_IEC61937_LE = (AF_FORMAT_S_IEC61937|AF_FORMAT_16BIT|AF_FORMAT_LE),
AF_FORMAT_IEC61937_BE = (AF_FORMAT_S_IEC61937|AF_FORMAT_16BIT|AF_FORMAT_BE),
AF_FORMAT_AC3 = (AF_FORMAT_S_AC3|AF_FORMAT_16BIT|AF_FORMAT_LE),
AF_FORMAT_IEC61937 = (AF_FORMAT_S_IEC61937|AF_FORMAT_16BIT|AF_FORMAT_LE),
AF_FORMAT_MPEG2 = (AF_FORMAT_S_MPEG2),
// Planar variants
@ -126,13 +122,10 @@ enum af_format {
AF_FORMAT_FLOAT = AF_SELECT_LE_BE(AF_FORMAT_FLOAT_LE, AF_FORMAT_FLOAT_BE),
AF_FORMAT_DOUBLE = AF_SELECT_LE_BE(AF_FORMAT_DOUBLE_LE, AF_FORMAT_DOUBLE_BE),
AF_FORMAT_AC3 = AF_SELECT_LE_BE(AF_FORMAT_AC3_LE, AF_FORMAT_AC3_BE),
AF_FORMAT_IEC61937 = AF_SELECT_LE_BE(AF_FORMAT_IEC61937_LE, AF_FORMAT_IEC61937_BE),
};
#define AF_FORMAT_IS_AC3(fmt) \
(((fmt) & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_S_AC3)
((fmt) == AF_FORMAT_AC3)
#define AF_FORMAT_IS_IEC61937(fmt) \
(AF_FORMAT_IS_AC3(fmt) || \

10
audio/out/ao_alsa.c
View File

@ -222,10 +222,8 @@ static const int mp_to_alsa_format[][2] = {
{AF_FORMAT_S24_BE, SND_PCM_FORMAT_S24_3BE},
{AF_FORMAT_FLOAT_LE, SND_PCM_FORMAT_FLOAT_LE},
{AF_FORMAT_FLOAT_BE, SND_PCM_FORMAT_FLOAT_BE},
{AF_FORMAT_AC3_LE, SND_PCM_FORMAT_S16_LE},
{AF_FORMAT_AC3_BE, SND_PCM_FORMAT_S16_BE},
{AF_FORMAT_IEC61937_LE, SND_PCM_FORMAT_S16_LE},
{AF_FORMAT_IEC61937_BE, SND_PCM_FORMAT_S16_BE},
{AF_FORMAT_AC3, SND_PCM_FORMAT_S16_LE},
{AF_FORMAT_IEC61937, SND_PCM_FORMAT_S16_LE},
{AF_FORMAT_MPEG2, SND_PCM_FORMAT_MPEG},
{AF_FORMAT_UNKNOWN, SND_PCM_FORMAT_UNKNOWN},
};
@ -421,9 +419,9 @@ static int init(struct ao *ao)
af_fmt_to_str(ao->format));
p->alsa_fmt = SND_PCM_FORMAT_S16_LE;
if (AF_FORMAT_IS_AC3(ao->format))
ao->format = AF_FORMAT_AC3_LE;
ao->format = AF_FORMAT_AC3;
else if (AF_FORMAT_IS_IEC61937(ao->format))
ao->format = AF_FORMAT_IEC61937_LE;
ao->format = AF_FORMAT_IEC61937;
else
ao->format = AF_FORMAT_S16_LE;
}

9
audio/out/ao_coreaudio_device.c
View File

@ -521,14 +521,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
err = ca_enable_device_listener(p->device, changed);
CHECK_CA_ERROR("cannot install format change listener during init");
#if BYTE_ORDER == BIG_ENDIAN
if (!(p->stream_asdb.mFormatFlags & kAudioFormatFlagIsBigEndian))
#else
/* tell mplayer that we need a byteswap on AC3 streams, */
if (p->stream_asbd.mFormatID & kAudioFormat60958AC3)
ao->format = AF_FORMAT_AC3_LE;
else if (p->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian)
#endif
if (p->stream_asdb.mFormatFlags & kAudioFormatFlagIsBigEndian)
MP_WARN(ao, "stream has non-native byte order, output may fail\n");
ao->samplerate = p->stream_asbd.mSampleRate;

2
audio/out/ao_pcm.c
View File

@ -130,7 +130,7 @@ static int init(struct ao *ao)
case AF_FORMAT_S24_LE:
case AF_FORMAT_S32_LE:
case AF_FORMAT_FLOAT_LE:
case AF_FORMAT_AC3_LE:
case AF_FORMAT_AC3:
break;
default:
ao->format = AF_FORMAT_S16_LE;