From 514c4547702a01adb6f9e6274c40a77c7c511339 Mon Sep 17 00:00:00 2001 From: wm4 Date: Fri, 15 Nov 2013 21:25:05 +0100 Subject: audio: drop "_NE"/"ne" suffix from audio formats You get the native format by not appending any suffix to the format. This change includes user-facing names, e.g. for the --format option. --- DOCS/man/en/af.rst | 12 ++++++------ audio/decode/ad_mpg123.c | 6 +++--- audio/filter/af_bs2b.c | 2 +- audio/filter/af_center.c | 2 +- audio/filter/af_convert24.c | 8 ++++---- audio/filter/af_drc.c | 8 ++++---- audio/filter/af_equalizer.c | 2 +- audio/filter/af_export.c | 2 +- audio/filter/af_extrastereo.c | 4 ++-- audio/filter/af_hrtf.c | 2 +- audio/filter/af_karaoke.c | 2 +- audio/filter/af_ladspa.c | 2 +- audio/filter/af_lavfi.c | 2 +- audio/filter/af_lavrresample.c | 2 +- audio/filter/af_pan.c | 2 +- audio/filter/af_scaletempo.c | 4 ++-- audio/filter/af_sinesuppress.c | 6 +++--- audio/filter/af_sub.c | 2 +- audio/filter/af_surround.c | 2 +- audio/filter/af_sweep.c | 2 +- audio/filter/af_volume.c | 10 +++++----- audio/fmt-conversion.c | 8 ++++---- audio/format.c | 1 - audio/format.h | 11 ----------- audio/out/ao_alsa.c | 4 ++-- audio/out/ao_dsound.c | 4 ++-- audio/out/ao_oss.c | 12 ++++++------ audio/out/ao_portaudio.c | 12 ++++++------ demux/demux_raw.c | 2 +- 29 files changed, 63 insertions(+), 75 deletions(-) diff --git a/DOCS/man/en/af.rst b/DOCS/man/en/af.rst index 67110f9b47..ab1e7cab8c 100644 --- a/DOCS/man/en/af.rst +++ b/DOCS/man/en/af.rst @@ -35,7 +35,7 @@ Available filters are: This filter is automatically enabled if the audio output does not support the audio configuration of the file being played. - It supports only the following sample formats: u8, s16ne, s32ne, floatne. + It supports only the following sample formats: u8, s16, s32, float. ``filter-size=`` Length of the filter with respect to the lower sampling rate. (default: @@ -223,11 +223,11 @@ Available filters are: Force conversion to this format. Use ``--af=format=format=help`` to get a list of valid formats. The general form is 'sbe', where 's' denotes the sign (either 's' for signed or 'u' for unsigned), 'b' denotes the - number of bits per sample (16, 24 or 32) and 'e' denotes the - endianness ('le' means little-endian, 'be' big-endian and 'ne' the + number of bits per sample (16, 24 or 32) and 'e' denotes the endian + ('le' means little-endian, 'be' big-endian and leaving it away the endianness of the computer mpv is running on). Valid values (amongst - others) are: 's16le', 'u32be' and 'u24ne'. Exceptions to this rule that - are also valid format specifiers: u8, s8, floatle, floatbe, floatne, + others) are: 's16le', 'u32be' and 'u24'. Exceptions to this rule that + are also valid format specifiers: u8, s8, floatle, floatbe, float, mpeg2, and ac3. ```` @@ -553,7 +553,7 @@ Available filters are: ``mpv --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg`` Would tweak the quality and performace parameters. - ``mpv --af=format=floatne,scaletempo media.ogg`` + ``mpv --af=format=float,scaletempo media.ogg`` Would make scaletempo use float code. Maybe faster on some platforms. diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c index 322f45826f..777c20c2c9 100644 --- a/audio/decode/ad_mpg123.c +++ b/audio/decode/ad_mpg123.c @@ -149,13 +149,13 @@ static int set_format(sh_audio_t *sh) sh->sample_format = AF_FORMAT_S8; break; case MPG123_ENC_SIGNED_16: - sh->sample_format = AF_FORMAT_S16_NE; + sh->sample_format = AF_FORMAT_S16; break; case MPG123_ENC_SIGNED_32: - sh->sample_format = AF_FORMAT_S32_NE; + sh->sample_format = AF_FORMAT_S32; break; case MPG123_ENC_FLOAT_32: - sh->sample_format = AF_FORMAT_FLOAT_NE; + sh->sample_format = AF_FORMAT_FLOAT; break; default: /* This means we got a funny custom build of libmpg123 that only supports an unknown format. */ diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c index 769a2b4577..c4f826e856 100644 --- a/audio/filter/af_bs2b.c +++ b/audio/filter/af_bs2b.c @@ -135,7 +135,7 @@ static int control(struct af_instance *af, int cmd, void *arg) break; default: af->play = play_f; - mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); + mp_audio_set_format(af->data, AF_FORMAT_FLOAT); break; } diff --git a/audio/filter/af_center.c b/audio/filter/af_center.c index ed482c7a6b..b64d5b54bd 100644 --- a/audio/filter/af_center.c +++ b/audio/filter/af_center.c @@ -50,7 +50,7 @@ static int control(struct af_instance* af, int cmd, void* arg) af->data->rate = ((struct mp_audio*)arg)->rate; mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch)); - mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); + mp_audio_set_format(af->data, AF_FORMAT_FLOAT); return af_test_output(af,(struct mp_audio*)arg); } diff --git a/audio/filter/af_convert24.c b/audio/filter/af_convert24.c index ee8aff5afc..0c3e6f8497 100644 --- a/audio/filter/af_convert24.c +++ b/audio/filter/af_convert24.c @@ -23,10 +23,10 @@ static bool test_conversion(int src_format, int dst_format) { - return (src_format == AF_FORMAT_U24_NE && dst_format == AF_FORMAT_U32_NE) || - (src_format == AF_FORMAT_S24_NE && dst_format == AF_FORMAT_S32_NE) || - (src_format == AF_FORMAT_U32_NE && dst_format == AF_FORMAT_U24_NE) || - (src_format == AF_FORMAT_S32_NE && dst_format == AF_FORMAT_S24_NE); + return (src_format == AF_FORMAT_U24 && dst_format == AF_FORMAT_U32) || + (src_format == AF_FORMAT_S24 && dst_format == AF_FORMAT_S32) || + (src_format == AF_FORMAT_U32 && dst_format == AF_FORMAT_U24) || + (src_format == AF_FORMAT_S32 && dst_format == AF_FORMAT_S24); } static int control(struct af_instance *af, int cmd, void *arg) diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c index 17c4a12a95..685dbcd8d5 100644 --- a/audio/filter/af_drc.c +++ b/audio/filter/af_drc.c @@ -91,8 +91,8 @@ static int control(struct af_instance* af, int cmd, void* arg) mp_audio_force_interleaved_format((struct mp_audio*)arg); mp_audio_copy_config(af->data, (struct mp_audio*)arg); - if(((struct mp_audio*)arg)->format != (AF_FORMAT_S16_NE)){ - mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); + if(((struct mp_audio*)arg)->format != (AF_FORMAT_S16)){ + mp_audio_set_format(af->data, AF_FORMAT_FLOAT); } return af_test_output(af,(struct mp_audio*)arg); case AF_CONTROL_COMMAND_LINE:{ @@ -296,14 +296,14 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) { af_drc_t *s = af->setup; - if(af->data->format == (AF_FORMAT_S16_NE)) + if(af->data->format == (AF_FORMAT_S16)) { if (s->method) method2_int16(s, data); else method1_int16(s, data); } - else if(af->data->format == (AF_FORMAT_FLOAT_NE)) + else if(af->data->format == (AF_FORMAT_FLOAT)) { if (s->method) method2_float(s, data); diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c index 75a489d867..6aff565607 100644 --- a/audio/filter/af_equalizer.c +++ b/audio/filter/af_equalizer.c @@ -98,7 +98,7 @@ static int control(struct af_instance* af, int cmd, void* arg) if(!arg) return AF_ERROR; mp_audio_copy_config(af->data, (struct mp_audio*)arg); - mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); + mp_audio_set_format(af->data, AF_FORMAT_FLOAT); // Calculate number of active filters s->K=KM; diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c index 5e1096f85a..6c1ea6459b 100644 --- a/audio/filter/af_export.c +++ b/audio/filter/af_export.c @@ -88,7 +88,7 @@ static int control(struct af_instance* af, int cmd, void* arg) // Accept only int16_t as input format (which sucks) mp_audio_copy_config(af->data, (struct mp_audio*)arg); - mp_audio_set_format(af->data, AF_FORMAT_S16_NE); + mp_audio_set_format(af->data, AF_FORMAT_S16); // If buffer length isn't set, set it to the default value if(s->sz == 0) diff --git a/audio/filter/af_extrastereo.c b/audio/filter/af_extrastereo.c index 4561b60690..6a00fb7e65 100644 --- a/audio/filter/af_extrastereo.c +++ b/audio/filter/af_extrastereo.c @@ -51,12 +51,12 @@ static int control(struct af_instance* af, int cmd, void* arg) mp_audio_copy_config(af->data, (struct mp_audio*)arg); mp_audio_force_interleaved_format(af->data); mp_audio_set_num_channels(af->data, 2); - if (af->data->format == AF_FORMAT_FLOAT_NE) + if (af->data->format == AF_FORMAT_FLOAT) { af->play = play_float; }// else { - mp_audio_set_format(af->data, AF_FORMAT_S16_NE); + mp_audio_set_format(af->data, AF_FORMAT_S16); af->play = play_s16; } diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c index ed51351750..bfb619b040 100644 --- a/audio/filter/af_hrtf.c +++ b/audio/filter/af_hrtf.c @@ -311,7 +311,7 @@ static int control(struct af_instance *af, int cmd, void* arg) } else if (af->data->nch < 5) mp_audio_set_channels_old(af->data, 5); - mp_audio_set_format(af->data, AF_FORMAT_S16_NE); + mp_audio_set_format(af->data, AF_FORMAT_S16); test_output_res = af_test_output(af, (struct mp_audio*)arg); // after testing input set the real output format mp_audio_set_num_channels(af->data, 2); diff --git a/audio/filter/af_karaoke.c b/audio/filter/af_karaoke.c index 8c633b136c..07ef0579bc 100644 --- a/audio/filter/af_karaoke.c +++ b/audio/filter/af_karaoke.c @@ -35,7 +35,7 @@ static int control(struct af_instance* af, int cmd, void* arg) switch(cmd){ case AF_CONTROL_REINIT: mp_audio_copy_config(af->data, (struct mp_audio*)arg); - mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); + mp_audio_set_format(af->data, AF_FORMAT_FLOAT); return af_test_output(af,(struct mp_audio*)arg); } return AF_UNKNOWN; diff --git a/audio/filter/af_ladspa.c b/audio/filter/af_ladspa.c index df88c06ab2..50d0bb5d85 100644 --- a/audio/filter/af_ladspa.c +++ b/audio/filter/af_ladspa.c @@ -495,7 +495,7 @@ static int control(struct af_instance *af, int cmd, void *arg) { /* accept FLOAT, let af_format do conversion */ mp_audio_copy_config(af->data, (struct mp_audio*)arg); - mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); + mp_audio_set_format(af->data, AF_FORMAT_FLOAT); return af_test_output(af, (struct mp_audio*)arg); case AF_CONTROL_COMMAND_LINE: { diff --git a/audio/filter/af_lavfi.c b/audio/filter/af_lavfi.c index 24ff8c5985..57a1055149 100644 --- a/audio/filter/af_lavfi.c +++ b/audio/filter/af_lavfi.c @@ -175,7 +175,7 @@ static int control(struct af_instance *af, int cmd, void *arg) struct mp_audio *out = af->data; if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE) - mp_audio_set_format(in, AF_FORMAT_FLOAT_NE); + mp_audio_set_format(in, AF_FORMAT_FLOAT); if (!mp_chmap_is_lavc(&in->channels)) mp_chmap_reorder_to_lavc(&in->channels); // will always work diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c index 2e298a3f39..9b4d8ca586 100644 --- a/audio/filter/af_lavrresample.c +++ b/audio/filter/af_lavrresample.c @@ -245,7 +245,7 @@ static int control(struct af_instance *af, int cmd, void *arg) mp_audio_set_channels(out, &in->channels); if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE) - mp_audio_set_format(in, AF_FORMAT_FLOAT_NE); + mp_audio_set_format(in, AF_FORMAT_FLOAT); if (af_to_avformat(out->format) == AV_SAMPLE_FMT_NONE) mp_audio_set_format(out, in->format); diff --git a/audio/filter/af_pan.c b/audio/filter/af_pan.c index 3d8c6045d0..29d38c3860 100644 --- a/audio/filter/af_pan.c +++ b/audio/filter/af_pan.c @@ -55,7 +55,7 @@ static int control(struct af_instance* af, int cmd, void* arg) if(!arg) return AF_ERROR; af->data->rate = ((struct mp_audio*)arg)->rate; - mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); + mp_audio_set_format(af->data, AF_FORMAT_FLOAT); set_channels(af->data, s->nch ? s->nch: ((struct mp_audio*)arg)->nch); if((af->data->format != ((struct mp_audio*)arg)->format) || diff --git a/audio/filter/af_scaletempo.c b/audio/filter/af_scaletempo.c index c8560af502..659b7971f5 100644 --- a/audio/filter/af_scaletempo.c +++ b/audio/filter/af_scaletempo.c @@ -290,10 +290,10 @@ static int control(struct af_instance *af, int cmd, void *arg) return af_test_output(af, data); } - if (data->format == AF_FORMAT_S16_NE) { + if (data->format == AF_FORMAT_S16) { use_int = 1; } else { - mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); + mp_audio_set_format(af->data, AF_FORMAT_FLOAT); } int bps = af->data->bps; diff --git a/audio/filter/af_sinesuppress.c b/audio/filter/af_sinesuppress.c index ef6fd7d37b..f241c5475a 100644 --- a/audio/filter/af_sinesuppress.c +++ b/audio/filter/af_sinesuppress.c @@ -57,15 +57,15 @@ static int control(struct af_instance* af, int cmd, void* arg) mp_audio_copy_config(af->data, (struct mp_audio*)arg); mp_audio_set_num_channels(af->data, 1); #if 0 - if (((struct mp_audio*)arg)->format == AF_FORMAT_FLOAT_NE) + if (((struct mp_audio*)arg)->format == AF_FORMAT_FLOAT) { - af->data->format = AF_FORMAT_FLOAT_NE; + af->data->format = AF_FORMAT_FLOAT; af->data->bps = 4; af->play = play_float; }// else #endif { - mp_audio_set_format(af->data, AF_FORMAT_S16_NE); + mp_audio_set_format(af->data, AF_FORMAT_S16); af->play = play_s16; } diff --git a/audio/filter/af_sub.c b/audio/filter/af_sub.c index 4fd16904c9..cdb4c04ea3 100644 --- a/audio/filter/af_sub.c +++ b/audio/filter/af_sub.c @@ -72,7 +72,7 @@ static int control(struct af_instance* af, int cmd, void* arg) af->data->rate = ((struct mp_audio*)arg)->rate; mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch)); - mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); + mp_audio_set_format(af->data, AF_FORMAT_FLOAT); // Design low-pass filter s->k = 1.0; diff --git a/audio/filter/af_surround.c b/audio/filter/af_surround.c index efeecdc1e3..f06789eabe 100644 --- a/audio/filter/af_surround.c +++ b/audio/filter/af_surround.c @@ -98,7 +98,7 @@ static int control(struct af_instance* af, int cmd, void* arg) return AF_DETACH; } - mp_audio_set_format(in, AF_FORMAT_FLOAT_NE); + mp_audio_set_format(in, AF_FORMAT_FLOAT); mp_audio_copy_config(af->data, in); mp_audio_set_channels_old(af->data, in->nch * 2); diff --git a/audio/filter/af_sweep.c b/audio/filter/af_sweep.c index c153d4261a..a06cad7600 100644 --- a/audio/filter/af_sweep.c +++ b/audio/filter/af_sweep.c @@ -42,7 +42,7 @@ static int control(struct af_instance* af, int cmd, void* arg) switch(cmd){ case AF_CONTROL_REINIT: mp_audio_copy_config(af->data, data); - mp_audio_set_format(af->data, AF_FORMAT_S16_NE); + mp_audio_set_format(af->data, AF_FORMAT_S16); return af_test_output(af, data); case AF_CONTROL_COMMAND_LINE: diff --git a/audio/filter/af_volume.c b/audio/filter/af_volume.c index b0eef1865c..edf29d00f2 100644 --- a/audio/filter/af_volume.c +++ b/audio/filter/af_volume.c @@ -47,10 +47,10 @@ static int control(struct af_instance *af, int cmd, void *arg) mp_audio_copy_config(af->data, in); mp_audio_force_interleaved_format(af->data); - if (s->fast && af_fmt_from_planar(in->format) != AF_FORMAT_FLOAT_NE) { - mp_audio_set_format(af->data, AF_FORMAT_S16_NE); + if (s->fast && af_fmt_from_planar(in->format) != AF_FORMAT_FLOAT) { + mp_audio_set_format(af->data, AF_FORMAT_S16); } else { - mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); + mp_audio_set_format(af->data, AF_FORMAT_FLOAT); } if (af_fmt_is_planar(in->format)) mp_audio_set_format(af->data, af_fmt_to_planar(af->data->format)); @@ -70,7 +70,7 @@ static void filter_plane(struct af_instance *af, void *ptr, int num_samples) { struct priv *s = af->priv; - if (af_fmt_from_planar(af->data->format) == AF_FORMAT_S16_NE) { + if (af_fmt_from_planar(af->data->format) == AF_FORMAT_S16) { int16_t *a = ptr; int vol = 256.0 * s->level; if (vol != 256) { @@ -79,7 +79,7 @@ static void filter_plane(struct af_instance *af, void *ptr, int num_samples) a[i] = MPCLAMP(x, SHRT_MIN, SHRT_MAX); } } - } else if (af_fmt_from_planar(af->data->format) == AF_FORMAT_FLOAT_NE) { + } else if (af_fmt_from_planar(af->data->format) == AF_FORMAT_FLOAT) { float *a = ptr; float vol = s->level; if (vol != 1.0) { diff --git a/audio/fmt-conversion.c b/audio/fmt-conversion.c index 93fda3eaa0..da770a8eda 100644 --- a/audio/fmt-conversion.c +++ b/audio/fmt-conversion.c @@ -27,10 +27,10 @@ static const struct { int fmt; } audio_conversion_map[] = { {AV_SAMPLE_FMT_U8, AF_FORMAT_U8}, - {AV_SAMPLE_FMT_S16, AF_FORMAT_S16_NE}, - {AV_SAMPLE_FMT_S32, AF_FORMAT_S32_NE}, - {AV_SAMPLE_FMT_FLT, AF_FORMAT_FLOAT_NE}, - {AV_SAMPLE_FMT_DBL, AF_FORMAT_DOUBLE_NE}, + {AV_SAMPLE_FMT_S16, AF_FORMAT_S16}, + {AV_SAMPLE_FMT_S32, AF_FORMAT_S32}, + {AV_SAMPLE_FMT_FLT, AF_FORMAT_FLOAT}, + {AV_SAMPLE_FMT_DBL, AF_FORMAT_DOUBLE}, {AV_SAMPLE_FMT_U8P, AF_FORMAT_U8P}, {AV_SAMPLE_FMT_S16P, AF_FORMAT_S16P}, diff --git a/audio/format.c b/audio/format.c index d0cd04cb88..3c22de476a 100644 --- a/audio/format.c +++ b/audio/format.c @@ -109,7 +109,6 @@ bool af_fmt_is_planar(int format) #define FMT_ENDIAN(string, id) \ {string, id}, \ - {string "ne", id}, \ {string "le", MP_CONCAT(id, _LE)}, \ {string "be", MP_CONCAT(id, _BE)}, \ diff --git a/audio/format.h b/audio/format.h index 9b855e4689..43f0da33f6 100644 --- a/audio/format.h +++ b/audio/format.h @@ -130,17 +130,6 @@ enum af_format { AF_FORMAT_IEC61937 = AF_SELECT_LE_BE(AF_FORMAT_IEC61937_LE, AF_FORMAT_IEC61937_BE), }; -#define AF_FORMAT_U16_NE AF_FORMAT_U16 -#define AF_FORMAT_S16_NE AF_FORMAT_S16 -#define AF_FORMAT_U24_NE AF_FORMAT_U24 -#define AF_FORMAT_S24_NE AF_FORMAT_S24 -#define AF_FORMAT_U32_NE AF_FORMAT_U32 -#define AF_FORMAT_S32_NE AF_FORMAT_S32 -#define AF_FORMAT_FLOAT_NE AF_FORMAT_FLOAT -#define AF_FORMAT_DOUBLE_NE AF_FORMAT_DOUBLE -#define AF_FORMAT_AC3_NE AF_FORMAT_AC3 -#define AF_FORMAT_IEC61937_NE AF_FORMAT_IEC61937 - #define AF_FORMAT_IS_AC3(fmt) \ (((fmt) & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_S_AC3) diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c index 86c813c143..a8df1c9f25 100644 --- a/audio/out/ao_alsa.c +++ b/audio/out/ao_alsa.c @@ -387,7 +387,7 @@ static int init(struct ao *ao) ao->channels.num); } else { device = select_chmap(ao); - if (strcmp(device, "default") != 0 && ao->format == AF_FORMAT_FLOAT_NE) + if (strcmp(device, "default") != 0 && ao->format == AF_FORMAT_FLOAT) { // hack - use the converter plugin (why the heck?) device = talloc_asprintf(ao, "plug:%s", device); @@ -435,7 +435,7 @@ static int init(struct ao *ao) 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_NE; + ao->format = AF_FORMAT_S16; } err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt); diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c index f828a210dc..ec5e83bd50 100644 --- a/audio/out/ao_dsound.c +++ b/audio/out/ao_dsound.c @@ -392,7 +392,7 @@ static int init(struct ao *ao) int rate = ao->samplerate; if (AF_FORMAT_IS_AC3(format)) - format = AF_FORMAT_AC3_NE; + format = AF_FORMAT_AC3; else { struct mp_chmap_sel sel = {0}; mp_chmap_sel_add_waveext(&sel); @@ -400,7 +400,7 @@ static int init(struct ao *ao) return -1; } switch (format) { - case AF_FORMAT_AC3_NE: + case AF_FORMAT_AC3: case AF_FORMAT_S24_LE: case AF_FORMAT_S16_LE: case AF_FORMAT_U8: diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c index 09a2951629..b1f2028af2 100644 --- a/audio/out/ao_oss.c +++ b/audio/out/ao_oss.c @@ -89,14 +89,14 @@ static int format_table[][2] = { {AFMT_S32_BE, AF_FORMAT_S32_BE}, #endif #ifdef AFMT_FLOAT - {AFMT_FLOAT, AF_FORMAT_FLOAT_NE}, + {AFMT_FLOAT, AF_FORMAT_FLOAT}, #endif // SPECIALS #ifdef AFMT_MPEG {AFMT_MPEG, AF_FORMAT_MPEG2}, #endif #ifdef AFMT_AC3 - {AFMT_AC3, AF_FORMAT_AC3_NE}, + {AFMT_AC3, AF_FORMAT_AC3}, #endif {-1, -1} }; @@ -269,7 +269,7 @@ static int init(struct ao *ao) ac3_retry: if (AF_FORMAT_IS_AC3(ao->format)) - ao->format = AF_FORMAT_AC3_NE; + ao->format = AF_FORMAT_AC3; oss_format = format2oss(ao->format); if (oss_format == -1) { MP_VERBOSE(ao, "Unknown/not supported internal format: %s\n", @@ -279,15 +279,15 @@ ac3_retry: #else oss_format = AFMT_S16_LE; #endif - ao->format = AF_FORMAT_S16_NE; + ao->format = AF_FORMAT_S16; } if (ioctl(p->audio_fd, SNDCTL_DSP_SETFMT, &oss_format) < 0 || oss_format != format2oss(ao->format)) { MP_WARN(ao, "Can't set audio device %s to %s output, trying %s...\n", p->dsp, af_fmt_to_str(ao->format), - af_fmt_to_str(AF_FORMAT_S16_NE)); - ao->format = AF_FORMAT_S16_NE; + af_fmt_to_str(AF_FORMAT_S16)); + ao->format = AF_FORMAT_S16; goto ac3_retry; } diff --git a/audio/out/ao_portaudio.c b/audio/out/ao_portaudio.c index 9c0d7804f8..fad1dc12d8 100644 --- a/audio/out/ao_portaudio.c +++ b/audio/out/ao_portaudio.c @@ -56,12 +56,12 @@ struct format_map { static const struct format_map format_maps[] = { // first entry is the default format - {AF_FORMAT_S16_NE, paInt16}, - {AF_FORMAT_S24_NE, paInt24}, - {AF_FORMAT_S32_NE, paInt32}, - {AF_FORMAT_S8, paInt8}, - {AF_FORMAT_U8, paUInt8}, - {AF_FORMAT_FLOAT_NE, paFloat32}, + {AF_FORMAT_S16, paInt16}, + {AF_FORMAT_S24, paInt24}, + {AF_FORMAT_S32, paInt32}, + {AF_FORMAT_S8, paInt8}, + {AF_FORMAT_U8, paUInt8}, + {AF_FORMAT_FLOAT, paFloat32}, {AF_FORMAT_UNKNOWN, 0} }; diff --git a/demux/demux_raw.c b/demux/demux_raw.c index c702756971..7fed124ae3 100644 --- a/demux/demux_raw.c +++ b/demux/demux_raw.c @@ -41,7 +41,7 @@ struct priv { static struct mp_chmap channels = MP_CHMAP_INIT_STEREO; static int samplerate = 44100; -static int aformat = AF_FORMAT_S16_NE; +static int aformat = AF_FORMAT_S16; const m_option_t demux_rawaudio_opts[] = { { "channels", &channels, &m_option_type_chmap, CONF_MIN, 1 }, -- cgit v1.2.3