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-rw-r--r--audio/decode/ad_spdif.c14
-rw-r--r--audio/filter/af_convertsignendian.c2
-rw-r--r--audio/filter/af_lavcac3enc.c2
-rw-r--r--audio/format.c23
-rw-r--r--audio/format.h69
-rw-r--r--audio/out/ao_alsa.c24
-rw-r--r--audio/out/ao_coreaudio_device.c5
-rw-r--r--audio/out/ao_coreaudio_utils.c5
-rw-r--r--audio/out/ao_dsound.c10
-rw-r--r--audio/out/ao_oss.c25
-rw-r--r--audio/out/ao_pcm.c4
-rwxr-xr-xaudio/out/ao_wasapi_utils.c2
-rw-r--r--demux/demux_raw.c2
-rw-r--r--player/audio.c2
14 files changed, 95 insertions, 94 deletions
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 3c5e0ef6e1..bb39298dd6 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -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;
+ sample_format = AF_FORMAT_S_AAC;
samplerate = 48000;
num_channels = 2;
break;
case AV_CODEC_ID_AC3:
spdif_ctx->iec61937_packet_size = 6144;
- sample_format = AF_FORMAT_AC3;
+ sample_format = AF_FORMAT_S_AC3;
samplerate = 48000;
num_channels = 2;
break;
@@ -131,31 +131,31 @@ 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;
+ sample_format = AF_FORMAT_S_DTSHD;
samplerate = 192000;
num_channels = 2*4;
} else {
spdif_ctx->iec61937_packet_size = 32768;
- sample_format = AF_FORMAT_AC3;
+ sample_format = AF_FORMAT_S_DTS;
samplerate = 48000;
num_channels = 2;
}
break;
case AV_CODEC_ID_EAC3:
spdif_ctx->iec61937_packet_size = 24576;
- sample_format = AF_FORMAT_IEC61937;
+ sample_format = AF_FORMAT_S_EAC3;
samplerate = 192000;
num_channels = 2;
break;
case AV_CODEC_ID_MP3:
spdif_ctx->iec61937_packet_size = 4608;
- sample_format = AF_FORMAT_MPEG2;
+ sample_format = AF_FORMAT_S_MP3;
samplerate = 48000;
num_channels = 2;
break;
case AV_CODEC_ID_TRUEHD:
spdif_ctx->iec61937_packet_size = 61440;
- sample_format = AF_FORMAT_IEC61937;
+ sample_format = AF_FORMAT_S_TRUEHD;
samplerate = 192000;
num_channels = 8;
break;
diff --git a/audio/filter/af_convertsignendian.c b/audio/filter/af_convertsignendian.c
index 7e7e436352..52ae8164df 100644
--- a/audio/filter/af_convertsignendian.c
+++ b/audio/filter/af_convertsignendian.c
@@ -29,7 +29,7 @@ static bool test_conversion(int src_format, int dst_format)
return false;
if (((src_format & ~AF_FORMAT_SIGN_MASK) ==
(dst_format & ~AF_FORMAT_SIGN_MASK)) &&
- ((src_format & AF_FORMAT_POINT_MASK) == AF_FORMAT_I))
+ ((src_format & AF_FORMAT_TYPE_MASK) == AF_FORMAT_I))
return true;
return false;
}
diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c
index 0763057665..18bca8e2cb 100644
--- a/audio/filter/af_lavcac3enc.c
+++ b/audio/filter/af_lavcac3enc.c
@@ -88,7 +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);
+ mp_audio_set_format(af->data, AF_FORMAT_S_AC3);
mp_audio_set_num_channels(af->data, 2);
if (!mp_audio_config_equals(in, &orig_in))
diff --git a/audio/format.c b/audio/format.c
index 60a86ea5ae..618a00e0c1 100644
--- a/audio/format.c
+++ b/audio/format.c
@@ -30,9 +30,6 @@
int af_fmt2bps(int format)
{
- if (AF_FORMAT_IS_AC3(format)) return 2;
- if (format == AF_FORMAT_UNKNOWN)
- return 0;
switch (format & AF_FORMAT_BITS_MASK) {
case AF_FORMAT_8BIT: return 1;
case AF_FORMAT_16BIT: return 2;
@@ -62,7 +59,7 @@ static int bits_to_mask(int bits)
int af_fmt_change_bits(int format, int bits)
{
- if (!af_fmt_is_valid(format) || (format & AF_FORMAT_SPECIAL_MASK))
+ if (!af_fmt_is_valid(format))
return 0;
int mask = bits_to_mask(bits);
format = (format & ~AF_FORMAT_BITS_MASK) | mask;
@@ -110,10 +107,6 @@ bool af_fmt_is_planar(int format)
}
const struct af_fmt_entry af_fmtstr_table[] = {
- {"mpeg2", AF_FORMAT_MPEG2},
- {"ac3", AF_FORMAT_AC3},
- {"iec61937", AF_FORMAT_IEC61937},
-
{"u8", AF_FORMAT_U8},
{"s8", AF_FORMAT_S8},
{"u16", AF_FORMAT_U16},
@@ -131,6 +124,14 @@ const struct af_fmt_entry af_fmtstr_table[] = {
{"floatp", AF_FORMAT_FLOATP},
{"doublep", AF_FORMAT_DOUBLEP},
+ {"spdif-aac", AF_FORMAT_S_AAC},
+ {"spdif-ac3", AF_FORMAT_S_AC3},
+ {"spdif-dts", AF_FORMAT_S_DTS},
+ {"spdif-dtshd", AF_FORMAT_S_DTSHD},
+ {"spdif-eac3", AF_FORMAT_S_EAC3},
+ {"spdif-mp3", AF_FORMAT_S_MP3},
+ {"spdif-truehd",AF_FORMAT_S_TRUEHD},
+
{0}
};
@@ -199,9 +200,9 @@ int af_format_conversion_score(int dst_format, int src_format)
score -= 1; // has to (de-)planarize
if (FMT_DIFF(AF_FORMAT_SIGN_MASK, dst_format, src_format))
score -= 4; // has to swap sign
- if (FMT_DIFF(AF_FORMAT_POINT_MASK, dst_format, src_format)) {
+ if (FMT_DIFF(AF_FORMAT_TYPE_MASK, dst_format, src_format)) {
int dst_bits = dst_format & AF_FORMAT_BITS_MASK;
- if ((dst_format & AF_FORMAT_POINT_MASK) == AF_FORMAT_F) {
+ if ((dst_format & AF_FORMAT_TYPE_MASK) == AF_FORMAT_F) {
// For int->float, always prefer 32 bit float.
score -= dst_bits == AF_FORMAT_32BIT ? 8 : 0;
} else {
@@ -217,7 +218,7 @@ int af_format_conversion_score(int dst_format, int src_format)
}
}
// Consider this the worst case.
- if (FMT_DIFF(AF_FORMAT_POINT_MASK, dst_format, src_format))
+ if (FMT_DIFF(AF_FORMAT_TYPE_MASK, dst_format, src_format))
score -= 2048; // has to convert float<->int
return score;
}
diff --git a/audio/format.h b/audio/format.h
index 3662d817f1..06c5d566e5 100644
--- a/audio/format.h
+++ b/audio/format.h
@@ -28,39 +28,38 @@
#include "misc/bstr.h"
// Signed/unsigned
-#define AF_FORMAT_SI (0<<1) // Signed
-#define AF_FORMAT_US (1<<1) // Unsigned
-#define AF_FORMAT_SIGN_MASK (1<<1)
+#define AF_FORMAT_SI (0<<0) // Signed
+#define AF_FORMAT_US (1<<0) // Unsigned
+#define AF_FORMAT_SIGN_MASK (1<<0)
// Bits used
// Some code assumes they're sorted by size.
-#define AF_FORMAT_8BIT (0<<3)
-#define AF_FORMAT_16BIT (1<<3)
-#define AF_FORMAT_24BIT (2<<3)
-#define AF_FORMAT_32BIT (3<<3)
-#define AF_FORMAT_64BIT (4<<3)
-#define AF_FORMAT_BITS_MASK (7<<3)
-
-// Special flags refering to non pcm data (note: 1<<6, 2<<6, 5<<6 unused)
-#define AF_FORMAT_S_MPEG2 (3<<6) // MPEG(2) audio
-#define AF_FORMAT_S_AC3 (4<<6) // Dolby Digital AC3
-#define AF_FORMAT_S_IEC61937 (6<<6)
-#define AF_FORMAT_SPECIAL_MASK (7<<6)
-
-// Fixed or floating point
-#define AF_FORMAT_I (1<<9) // Int
-#define AF_FORMAT_F (2<<9) // Foating point
-#define AF_FORMAT_POINT_MASK (3<<9)
+#define AF_FORMAT_8BIT (0<<1)
+#define AF_FORMAT_16BIT (1<<1)
+#define AF_FORMAT_24BIT (2<<1)
+#define AF_FORMAT_32BIT (3<<1)
+#define AF_FORMAT_64BIT (4<<1)
+#define AF_FORMAT_BITS_MASK (7<<1)
+
+// Fixed/floating point/special
+#define AF_FORMAT_I (1<<4) // Int
+#define AF_FORMAT_F (2<<4) // Foating point
+#define AF_FORMAT_S (4<<4) // special (IEC61937)
+#define AF_FORMAT_TYPE_MASK (7<<4)
// Interleaving (planar formats have data for each channel in separate planes)
-#define AF_FORMAT_INTERLEAVED (0<<11) // must be 0
-#define AF_FORMAT_PLANAR (1<<11)
-#define AF_FORMAT_INTERLEAVING_MASK (1<<11)
+#define AF_FORMAT_INTERLEAVED (0<<7) // must be 0
+#define AF_FORMAT_PLANAR (1<<7)
+#define AF_FORMAT_INTERLEAVING_MASK (1<<7)
+
+#define AF_FORMAT_S_CODEC(n) ((n)<<8)
+#define AF_FORMAT_S_CODEC_MASK (15 <<8) // 16 codecs max.
#define AF_FORMAT_MASK ((1<<12)-1)
#define AF_INTP (AF_FORMAT_I|AF_FORMAT_PLANAR)
#define AF_FLTP (AF_FORMAT_F|AF_FORMAT_PLANAR)
+#define AF_FORMAT_S_(n) (AF_FORMAT_S_CODEC(n)|AF_FORMAT_S|AF_FORMAT_16BIT)
// actual sample formats
enum af_format {
@@ -78,27 +77,25 @@ enum af_format {
AF_FORMAT_FLOAT = (AF_FORMAT_F|AF_FORMAT_32BIT),
AF_FORMAT_DOUBLE = (AF_FORMAT_F|AF_FORMAT_64BIT),
- AF_FORMAT_AC3 = (AF_FORMAT_S_AC3|AF_FORMAT_16BIT),
- AF_FORMAT_IEC61937 = (AF_FORMAT_S_IEC61937|AF_FORMAT_16BIT),
- AF_FORMAT_MPEG2 = (AF_FORMAT_S_MPEG2),
-
// Planar variants
AF_FORMAT_U8P = (AF_INTP|AF_FORMAT_US|AF_FORMAT_8BIT),
AF_FORMAT_S16P = (AF_INTP|AF_FORMAT_SI|AF_FORMAT_16BIT),
AF_FORMAT_S32P = (AF_INTP|AF_FORMAT_SI|AF_FORMAT_32BIT),
AF_FORMAT_FLOATP = (AF_FLTP|AF_FORMAT_32BIT),
AF_FORMAT_DOUBLEP = (AF_FLTP|AF_FORMAT_64BIT),
-};
-#define AF_FORMAT_IS_AC3(fmt) \
- ((fmt) == AF_FORMAT_AC3)
-
-#define AF_FORMAT_IS_IEC61937(fmt) \
- (AF_FORMAT_IS_AC3(fmt) || \
- ((fmt) & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_S_IEC61937)
+ // All of these use IEC61937 framing, and otherwise pretend to be like PCM.
+ AF_FORMAT_S_AAC = AF_FORMAT_S_(0),
+ AF_FORMAT_S_AC3 = AF_FORMAT_S_(1),
+ AF_FORMAT_S_DTS = AF_FORMAT_S_(2),
+ AF_FORMAT_S_DTSHD = AF_FORMAT_S_(3),
+ AF_FORMAT_S_EAC3 = AF_FORMAT_S_(4),
+ AF_FORMAT_S_MP3 = AF_FORMAT_S_(5),
+ AF_FORMAT_S_TRUEHD = AF_FORMAT_S_(6),
+};
-#define AF_FORMAT_IS_SPECIAL(fmt) \
- ((fmt & AF_FORMAT_SPECIAL_MASK) != 0)
+#define AF_FORMAT_IS_IEC61937(f) (((f) & AF_FORMAT_TYPE_MASK) == AF_FORMAT_S)
+#define AF_FORMAT_IS_SPECIAL(f) AF_FORMAT_IS_IEC61937(f)
struct af_fmt_entry {
const char *name;
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index c344f6f20e..52f410cf5e 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -100,7 +100,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
long get_vol, set_vol;
float f_multi;
- if (AF_FORMAT_IS_IEC61937(ao->format))
+ if (AF_FORMAT_IS_SPECIAL(ao->format))
return CONTROL_FALSE;
//allocate simple id
@@ -218,9 +218,6 @@ static const int mp_to_alsa_format[][2] = {
{AF_FORMAT_S24,
MP_SELECT_LE_BE(SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_S24_3BE)},
{AF_FORMAT_FLOAT, SND_PCM_FORMAT_FLOAT},
- {AF_FORMAT_AC3, SND_PCM_FORMAT_S16},
- {AF_FORMAT_IEC61937, SND_PCM_FORMAT_S16},
- {AF_FORMAT_MPEG2, SND_PCM_FORMAT_MPEG},
{AF_FORMAT_UNKNOWN, SND_PCM_FORMAT_UNKNOWN},
};
@@ -403,7 +400,15 @@ static int init(struct ao *ao)
err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams);
CHECK_ALSA_ERROR("Unable to get initial parameters");
- p->alsa_fmt = find_alsa_format(ao->format);
+ if (AF_FORMAT_IS_IEC61937(ao->format)) {
+ if (ao->format == AF_FORMAT_S_MP3) {
+ p->alsa_fmt = SND_PCM_FORMAT_MPEG;
+ } else {
+ p->alsa_fmt = SND_PCM_FORMAT_S16;
+ }
+ } else {
+ p->alsa_fmt = find_alsa_format(ao->format);
+ }
if (p->alsa_fmt == SND_PCM_FORMAT_UNKNOWN) {
p->alsa_fmt = SND_PCM_FORMAT_S16;
ao->format = AF_FORMAT_S16;
@@ -411,15 +416,12 @@ static int init(struct ao *ao)
err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt);
if (err < 0) {
+ if (AF_FORMAT_IS_IEC61937(ao->format))
+ CHECK_ALSA_ERROR("Unable to set IEC61937 format");
MP_INFO(ao, "Format %s is not supported by hardware, trying default.\n",
af_fmt_to_str(ao->format));
p->alsa_fmt = SND_PCM_FORMAT_S16;
- if (AF_FORMAT_IS_AC3(ao->format))
- ao->format = AF_FORMAT_AC3;
- else if (AF_FORMAT_IS_IEC61937(ao->format))
- ao->format = AF_FORMAT_IEC61937;
- else
- ao->format = AF_FORMAT_S16;
+ ao->format = AF_FORMAT_S16;
}
err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt);
diff --git a/audio/out/ao_coreaudio_device.c b/audio/out/ao_coreaudio_device.c
index 3ba1f0ce37..c221b62e59 100644
--- a/audio/out/ao_coreaudio_device.c
+++ b/audio/out/ao_coreaudio_device.c
@@ -410,8 +410,9 @@ static int init(struct ao *ao)
ao->format = af_fmt_from_planar(ao->format);
bool supports_digital = false;
- /* Probe whether device support S/PDIF stream output if input is AC3 stream. */
- if (AF_FORMAT_IS_AC3(ao->format)) {
+ /* Probe whether device support S/PDIF stream output if input is AC3 stream,
+ * or anything else IEC61937-framed. */
+ if (AF_FORMAT_IS_IEC61937(ao->format)) {
if (ca_device_supports_digital(ao, p->device))
supports_digital = true;
}
diff --git a/audio/out/ao_coreaudio_utils.c b/audio/out/ao_coreaudio_utils.c
index b39323ce72..08214ef387 100644
--- a/audio/out/ao_coreaudio_utils.c
+++ b/audio/out/ao_coreaudio_utils.c
@@ -130,14 +130,15 @@ bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message)
void ca_fill_asbd(struct ao *ao, AudioStreamBasicDescription *asbd)
{
asbd->mSampleRate = ao->samplerate;
- asbd->mFormatID = AF_FORMAT_IS_AC3(ao->format) ?
+ // Set "AC3" for other spdif formats too - unknown if that works.
+ asbd->mFormatID = AF_FORMAT_IS_IEC61937(ao->format) ?
kAudioFormat60958AC3 :
kAudioFormatLinearPCM;
asbd->mChannelsPerFrame = ao->channels.num;
asbd->mBitsPerChannel = af_fmt2bits(ao->format);
asbd->mFormatFlags = kAudioFormatFlagIsPacked;
- if ((ao->format & AF_FORMAT_POINT_MASK) == AF_FORMAT_F)
+ if ((ao->format & AF_FORMAT_TYPE_MASK) == AF_FORMAT_F)
asbd->mFormatFlags |= kAudioFormatFlagIsFloat;
if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI)
diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c
index 78b39b17b9..cd6a986831 100644
--- a/audio/out/ao_dsound.c
+++ b/audio/out/ao_dsound.c
@@ -382,21 +382,20 @@ static int init(struct ao *ao)
int format = af_fmt_from_planar(ao->format);
int rate = ao->samplerate;
- if (AF_FORMAT_IS_AC3(format))
- format = AF_FORMAT_AC3;
- else {
+ if (!AF_FORMAT_IS_IEC61937(format)) {
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_waveext(&sel);
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
return -1;
}
switch (format) {
- case AF_FORMAT_AC3:
case AF_FORMAT_S24:
case AF_FORMAT_S16:
case AF_FORMAT_U8:
break;
default:
+ if (AF_FORMAT_IS_IEC61937(format))
+ break;
MP_VERBOSE(ao, "format %s not supported defaulting to Signed 16-bit Little-Endian\n",
af_fmt_to_str(format));
format = AF_FORMAT_S16;
@@ -417,7 +416,8 @@ static int init(struct ao *ao)
? sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX) : 0;
wformat.Format.nChannels = ao->channels.num;
wformat.Format.nSamplesPerSec = rate;
- if (AF_FORMAT_IS_AC3(format)) {
+ if (AF_FORMAT_IS_IEC61937(format)) {
+ // Whether it also works with e.g. DTS is unknown, but probably does.
wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
wformat.Format.wBitsPerSample = 16;
wformat.Format.nBlockAlign = 4;
diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c
index de71017432..7780a08978 100644
--- a/audio/out/ao_oss.c
+++ b/audio/out/ao_oss.c
@@ -134,16 +134,13 @@ static const int format_table[][2] = {
#ifdef AFMT_FLOAT
{AFMT_FLOAT, AF_FORMAT_FLOAT},
#endif
- // SPECIALS
-#ifdef AFMT_MPEG
- {AFMT_MPEG, AF_FORMAT_MPEG2},
-#endif
-#ifdef AFMT_AC3
- {AFMT_AC3, AF_FORMAT_AC3},
-#endif
{-1, -1}
};
+#ifndef AFMT_AC3
+#define AFMT_AC3 -1
+#endif
+
static int format2oss(int format)
{
for (int n = 0; format_table[n][0] != -1; n++) {
@@ -204,7 +201,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
return CONTROL_OK;
#endif
- if (AF_FORMAT_IS_AC3(ao->format))
+ if (AF_FORMAT_IS_SPECIAL(ao->format))
return CONTROL_TRUE;
if ((fd = open(p->oss_mixer_device, O_RDONLY)) != -1) {
@@ -293,14 +290,16 @@ static int reopen_device(struct ao *ao, bool allow_format_changes)
fcntl(p->audio_fd, F_SETFD, FD_CLOEXEC);
#endif
- if (AF_FORMAT_IS_AC3(format)) {
+ if (AF_FORMAT_IS_IEC61937(format)) {
ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &samplerate);
}
ac3_retry:
- if (AF_FORMAT_IS_AC3(format))
- format = AF_FORMAT_AC3;
- oss_format = format2oss(format);
+ if (AF_FORMAT_IS_IEC61937(format)) {
+ oss_format = format2oss(format);
+ } else {
+ oss_format = AFMT_AC3;
+ }
if (oss_format == -1) {
MP_VERBOSE(ao, "Unknown/not supported internal format: %s\n",
af_fmt_to_str(format));
@@ -345,7 +344,7 @@ ac3_retry:
MP_VERBOSE(ao, "sample format: %s\n", af_fmt_to_str(format));
- if (!AF_FORMAT_IS_AC3(format)) {
+ if (!AF_FORMAT_IS_IEC61937(format)) {
struct mp_chmap_sel sel = {0};
for (int n = 0; n < MP_NUM_CHANNELS + 1; n++)
mp_chmap_sel_add_map(&sel, &oss_layouts[n]);
diff --git a/audio/out/ao_pcm.c b/audio/out/ao_pcm.c
index eb089c6c42..8357174f95 100644
--- a/audio/out/ao_pcm.c
+++ b/audio/out/ao_pcm.c
@@ -137,10 +137,10 @@ static int init(struct ao *ao)
case AF_FORMAT_S24:
case AF_FORMAT_S32:
case AF_FORMAT_FLOAT:
- case AF_FORMAT_AC3:
break;
default:
- ao->format = AF_FORMAT_S16;
+ if (!AF_FORMAT_IS_IEC61937(ao->format))
+ ao->format = AF_FORMAT_S16;
break;
}
}
diff --git a/audio/out/ao_wasapi_utils.c b/audio/out/ao_wasapi_utils.c
index c39eb6fea2..bf59d7802b 100755
--- a/audio/out/ao_wasapi_utils.c
+++ b/audio/out/ao_wasapi_utils.c
@@ -300,7 +300,7 @@ static int find_formats(struct ao *const ao)
{
struct wasapi_state *state = (struct wasapi_state *)ao->priv;
- if (AF_FORMAT_IS_IEC61937(ao->format) || ao->format == AF_FORMAT_MPEG2) {
+ if (AF_FORMAT_IS_IEC61937(ao->format)) {
if (try_passthrough(state, ao))
return 0;
diff --git a/demux/demux_raw.c b/demux/demux_raw.c
index 5dae30308a..552c74a2a3 100644
--- a/demux/demux_raw.c
+++ b/demux/demux_raw.c
@@ -110,7 +110,7 @@ static int demux_rawaudio_open(demuxer_t *demuxer, enum demux_check check)
if (check != DEMUX_CHECK_REQUEST && check != DEMUX_CHECK_FORCE)
return -1;
- if ((opts->aformat & AF_FORMAT_SPECIAL_MASK) != 0)
+ if (AF_FORMAT_IS_SPECIAL(opts->aformat))
return -1;
sh = new_sh_stream(demuxer, STREAM_AUDIO);
diff --git a/player/audio.c b/player/audio.c
index c74032f174..56e6c233c7 100644
--- a/player/audio.c
+++ b/player/audio.c
@@ -315,7 +315,7 @@ static bool get_sync_samples(struct MPContext *mpctx, int *skip)
ao_get_format(mpctx->ao, &out_format);
double play_samplerate = out_format.rate / opts->playback_speed;
- bool is_pcm = !(out_format.format & AF_FORMAT_SPECIAL_MASK); // no spdif
+ bool is_pcm = !AF_FORMAT_IS_SPECIAL(out_format.format); // no spdif
if (!opts->initial_audio_sync || !is_pcm) {
mpctx->audio_status = STATUS_FILLING;
return true;
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-26 02:44:21 +0000