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-rw-r--r--audio/decode/ad.c50
-rw-r--r--audio/decode/ad.h54
-rw-r--r--audio/decode/ad_dvdpcm.c162
-rw-r--r--audio/decode/ad_internal.h46
-rw-r--r--audio/decode/ad_lavc.c413
-rw-r--r--audio/decode/ad_mpg123.c489
-rw-r--r--audio/decode/ad_pcm.c220
-rw-r--r--audio/decode/ad_spdif.c310
-rw-r--r--audio/decode/dec_audio.c462
-rw-r--r--audio/decode/dec_audio.h38
-rw-r--r--audio/filter/af.c700
-rw-r--r--audio/filter/af.h349
-rw-r--r--audio/filter/af_bs2b.c274
-rw-r--r--audio/filter/af_center.c129
-rw-r--r--audio/filter/af_channels.c306
-rw-r--r--audio/filter/af_delay.c200
-rw-r--r--audio/filter/af_dummy.c76
-rw-r--r--audio/filter/af_equalizer.c248
-rw-r--r--audio/filter/af_export.c273
-rw-r--r--audio/filter/af_extrastereo.c157
-rw-r--r--audio/filter/af_format.c519
-rw-r--r--audio/filter/af_format_alaw.h324
-rw-r--r--audio/filter/af_format_ulaw.h837
-rw-r--r--audio/filter/af_hrtf.c670
-rw-r--r--audio/filter/af_hrtf.h511
-rw-r--r--audio/filter/af_karaoke.c98
-rw-r--r--audio/filter/af_ladspa.c915
-rw-r--r--audio/filter/af_lavcac3enc.c332
-rw-r--r--audio/filter/af_lavcresample.c213
-rw-r--r--audio/filter/af_pan.c210
-rw-r--r--audio/filter/af_resample.c394
-rw-r--r--audio/filter/af_resample_template.c171
-rw-r--r--audio/filter/af_scaletempo.c581
-rw-r--r--audio/filter/af_sinesuppress.c184
-rw-r--r--audio/filter/af_sub.c188
-rw-r--r--audio/filter/af_surround.c273
-rw-r--r--audio/filter/af_sweep.c103
-rw-r--r--audio/filter/af_tools.c110
-rw-r--r--audio/filter/af_volnorm.c353
-rw-r--r--audio/filter/af_volume.c226
-rw-r--r--audio/filter/control.h257
-rw-r--r--audio/filter/dsp.h32
-rw-r--r--audio/filter/equalizer.h48
-rw-r--r--audio/filter/filter.c360
-rw-r--r--audio/filter/filter.h75
-rw-r--r--audio/filter/window.c213
-rw-r--r--audio/filter/window.h43
-rw-r--r--audio/format.c134
-rw-r--r--audio/format.h137
-rw-r--r--audio/mixer.c292
-rw-r--r--audio/mixer.h61
-rw-r--r--audio/out/ao.c294
-rw-r--r--audio/out/ao.h140
-rw-r--r--audio/out/ao_alsa.c868
-rw-r--r--audio/out/ao_coreaudio.c1283
-rw-r--r--audio/out/ao_dsound.c648
-rw-r--r--audio/out/ao_jack.c361
-rw-r--r--audio/out/ao_lavc.c621
-rw-r--r--audio/out/ao_null.c129
-rw-r--r--audio/out/ao_openal.c280
-rw-r--r--audio/out/ao_oss.c560
-rw-r--r--audio/out/ao_pcm.c256
-rw-r--r--audio/out/ao_portaudio.c431
-rw-r--r--audio/out/ao_pulse.c554
-rw-r--r--audio/out/ao_rsound.c214
-rw-r--r--audio/out/audio_out_internal.h65
-rw-r--r--audio/reorder_ch.c1400
-rw-r--r--audio/reorder_ch.h133
68 files changed, 22057 insertions, 0 deletions
diff --git a/audio/decode/ad.c b/audio/decode/ad.c
new file mode 100644
index 0000000000..93cebed86d
--- /dev/null
+++ b/audio/decode/ad.c
@@ -0,0 +1,50 @@
+/*
+ * audio decoder interface
+ *
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "config.h"
+
+#include "stream/stream.h"
+#include "libmpdemux/demuxer.h"
+#include "libmpdemux/stheader.h"
+#include "ad.h"
+
+/* Missed vorbis, mad, dshow */
+
+extern const ad_functions_t mpcodecs_ad_mpg123;
+extern const ad_functions_t mpcodecs_ad_ffmpeg;
+extern const ad_functions_t mpcodecs_ad_pcm;
+extern const ad_functions_t mpcodecs_ad_dvdpcm;
+extern const ad_functions_t mpcodecs_ad_spdif;
+
+const ad_functions_t * const mpcodecs_ad_drivers[] =
+{
+#ifdef CONFIG_MPG123
+ &mpcodecs_ad_mpg123,
+#endif
+ &mpcodecs_ad_ffmpeg,
+ &mpcodecs_ad_pcm,
+ &mpcodecs_ad_dvdpcm,
+ &mpcodecs_ad_spdif,
+ NULL
+};
diff --git a/audio/decode/ad.h b/audio/decode/ad.h
new file mode 100644
index 0000000000..5396085d04
--- /dev/null
+++ b/audio/decode/ad.h
@@ -0,0 +1,54 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPLAYER_AD_H
+#define MPLAYER_AD_H
+
+#include "mpc_info.h"
+#include "libmpdemux/stheader.h"
+
+typedef struct mp_codec_info ad_info_t;
+
+/* interface of video decoder drivers */
+typedef struct ad_functions
+{
+ const ad_info_t *info;
+ int (*preinit)(sh_audio_t *sh);
+ int (*init)(sh_audio_t *sh);
+ void (*uninit)(sh_audio_t *sh);
+ int (*control)(sh_audio_t *sh,int cmd,void* arg, ...);
+ int (*decode_audio)(sh_audio_t *sh, unsigned char *buffer, int minlen,
+ int maxlen);
+} ad_functions_t;
+
+// NULL terminated array of all drivers
+extern const ad_functions_t * const mpcodecs_ad_drivers[];
+
+// fallback if ADCTRL_RESYNC not implemented: sh_audio->a_in_buffer_len=0;
+#define ADCTRL_RESYNC_STREAM 1 // resync, called after seeking
+
+// fallback if ADCTRL_SKIP not implemented: ds_fill_buffer(sh_audio->ds);
+#define ADCTRL_SKIP_FRAME 2 // skip block/frame, called while seeking
+
+// fallback if ADCTRL_QUERY_FORMAT not implemented: sh_audio->sample_format
+#define ADCTRL_QUERY_FORMAT 3 // test for availabilty of a format
+
+// fallback: use hw mixer in libao
+#define ADCTRL_SET_VOLUME 4 // not used at the moment
+
+#endif /* MPLAYER_AD_H */
diff --git a/audio/decode/ad_dvdpcm.c b/audio/decode/ad_dvdpcm.c
new file mode 100644
index 0000000000..41f6a1426d
--- /dev/null
+++ b/audio/decode/ad_dvdpcm.c
@@ -0,0 +1,162 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "mp_msg.h"
+#include "ad_internal.h"
+
+static const ad_info_t info =
+{
+ "Uncompressed DVD/VOB LPCM audio decoder",
+ "dvdpcm",
+ "Nick Kurshev",
+ "A'rpi",
+ ""
+};
+
+LIBAD_EXTERN(dvdpcm)
+
+static int init(sh_audio_t *sh)
+{
+/* DVD PCM Audio:*/
+ sh->i_bps = 0;
+ if(sh->codecdata_len==3){
+ // we have LPCM header:
+ unsigned char h=sh->codecdata[1];
+ sh->channels=1+(h&7);
+ switch((h>>4)&3){
+ case 0: sh->samplerate=48000;break;
+ case 1: sh->samplerate=96000;break;
+ case 2: sh->samplerate=44100;break;
+ case 3: sh->samplerate=32000;break;
+ }
+ switch ((h >> 6) & 3) {
+ case 0:
+ sh->sample_format = AF_FORMAT_S16_BE;
+ sh->samplesize = 2;
+ break;
+ case 1:
+ mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Samples of this format are needed to improve support. Please contact the developers.\n");
+ sh->i_bps = sh->channels * sh->samplerate * 5 / 2;
+ case 2:
+ sh->sample_format = AF_FORMAT_S24_BE;
+ sh->samplesize = 3;
+ break;
+ default:
+ sh->sample_format = AF_FORMAT_S16_BE;
+ sh->samplesize = 2;
+ }
+ } else {
+ // use defaults:
+ sh->channels=2;
+ sh->samplerate=48000;
+ sh->sample_format = AF_FORMAT_S16_BE;
+ sh->samplesize = 2;
+ }
+ if (!sh->i_bps)
+ sh->i_bps = sh->samplesize * sh->channels * sh->samplerate;
+ return 1;
+}
+
+static int preinit(sh_audio_t *sh)
+{
+ sh->audio_out_minsize=2048;
+ return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+}
+
+static int control(sh_audio_t *sh,int cmd,void* arg, ...)
+{
+ int skip;
+ switch(cmd)
+ {
+ case ADCTRL_SKIP_FRAME:
+ skip=sh->i_bps/16;
+ skip=skip&(~3);
+ demux_read_data(sh->ds,NULL,skip);
+ return CONTROL_TRUE;
+ }
+ return CONTROL_UNKNOWN;
+}
+
+static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
+{
+ int j,len;
+ if (sh_audio->samplesize == 3) {
+ if (((sh_audio->codecdata[1] >> 6) & 3) == 1) {
+ // 20 bit
+ // not sure if the "& 0xf0" and "<< 4" are the right way around
+ // can somebody clarify?
+ for (j = 0; j < minlen; j += 12) {
+ char tmp[10];
+ len = demux_read_data(sh_audio->ds, tmp, 10);
+ if (len < 10) break;
+ // first sample
+ buf[j + 0] = tmp[0];
+ buf[j + 1] = tmp[1];
+ buf[j + 2] = tmp[8] & 0xf0;
+ // second sample
+ buf[j + 3] = tmp[2];
+ buf[j + 4] = tmp[3];
+ buf[j + 5] = tmp[8] << 4;
+ // third sample
+ buf[j + 6] = tmp[4];
+ buf[j + 7] = tmp[5];
+ buf[j + 8] = tmp[9] & 0xf0;
+ // fourth sample
+ buf[j + 9] = tmp[6];
+ buf[j + 10] = tmp[7];
+ buf[j + 11] = tmp[9] << 4;
+ }
+ len = j;
+ } else {
+ // 24 bit
+ for (j = 0; j < minlen; j += 12) {
+ char tmp[12];
+ len = demux_read_data(sh_audio->ds, tmp, 12);
+ if (len < 12) break;
+ // first sample
+ buf[j + 0] = tmp[0];
+ buf[j + 1] = tmp[1];
+ buf[j + 2] = tmp[8];
+ // second sample
+ buf[j + 3] = tmp[2];
+ buf[j + 4] = tmp[3];
+ buf[j + 5] = tmp[9];
+ // third sample
+ buf[j + 6] = tmp[4];
+ buf[j + 7] = tmp[5];
+ buf[j + 8] = tmp[10];
+ // fourth sample
+ buf[j + 9] = tmp[6];
+ buf[j + 10] = tmp[7];
+ buf[j + 11] = tmp[11];
+ }
+ len = j;
+ }
+ } else
+ len=demux_read_data(sh_audio->ds,buf,(minlen+3)&(~3));
+ return len;
+}
diff --git a/audio/decode/ad_internal.h b/audio/decode/ad_internal.h
new file mode 100644
index 0000000000..4cffc95126
--- /dev/null
+++ b/audio/decode/ad_internal.h
@@ -0,0 +1,46 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPLAYER_AD_INTERNAL_H
+#define MPLAYER_AD_INTERNAL_H
+
+#include "codec-cfg.h"
+#include "libaf/format.h"
+
+#include "stream/stream.h"
+#include "libmpdemux/demuxer.h"
+#include "libmpdemux/stheader.h"
+
+#include "ad.h"
+
+static int init(sh_audio_t *sh);
+static int preinit(sh_audio_t *sh);
+static void uninit(sh_audio_t *sh);
+static int control(sh_audio_t *sh,int cmd,void* arg, ...);
+static int decode_audio(sh_audio_t *sh,unsigned char *buffer,int minlen,int maxlen);
+
+#define LIBAD_EXTERN(x) const ad_functions_t mpcodecs_ad_##x = {\
+ &info,\
+ preinit,\
+ init,\
+ uninit,\
+ control,\
+ decode_audio\
+};
+
+#endif /* MPLAYER_AD_INTERNAL_H */
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
new file mode 100644
index 0000000000..2eacfadb8f
--- /dev/null
+++ b/audio/decode/ad_lavc.c
@@ -0,0 +1,413 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <stdbool.h>
+#include <assert.h>
+
+#include <libavcodec/avcodec.h>
+#include <libavutil/opt.h>
+
+#include "talloc.h"
+
+#include "config.h"
+#include "mp_msg.h"
+#include "options.h"
+
+#include "ad_internal.h"
+#include "libaf/reorder_ch.h"
+
+#include "mpbswap.h"
+
+static const ad_info_t info =
+{
+ "libavcodec audio decoders",
+ "ffmpeg",
+ "",
+ "",
+ "",
+ .print_name = "libavcodec",
+};
+
+LIBAD_EXTERN(ffmpeg)
+
+struct priv {
+ AVCodecContext *avctx;
+ AVFrame *avframe;
+ char *output;
+ char *output_packed; // used by deplanarize to store packed audio samples
+ int output_left;
+ int unitsize;
+ int previous_data_left; // input demuxer packet data
+};
+
+static int preinit(sh_audio_t *sh)
+{
+ return 1;
+}
+
+/* Prefer playing audio with the samplerate given in container data
+ * if available, but take number the number of channels and sample format
+ * from the codec, since if the codec isn't using the correct values for
+ * those everything breaks anyway.
+ */
+static int setup_format(sh_audio_t *sh_audio,
+ const AVCodecContext *lavc_context)
+{
+ int sample_format = sh_audio->sample_format;
+ switch (av_get_packed_sample_fmt(lavc_context->sample_fmt)) {
+ case AV_SAMPLE_FMT_U8: sample_format = AF_FORMAT_U8; break;
+ case AV_SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE; break;
+ case AV_SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE; break;
+ case AV_SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
+ default:
+ mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
+ sample_format = AF_FORMAT_UNKNOWN;
+ }
+
+ bool broken_srate = false;
+ int samplerate = lavc_context->sample_rate;
+ int container_samplerate = sh_audio->container_out_samplerate;
+ if (!container_samplerate && sh_audio->wf)
+ container_samplerate = sh_audio->wf->nSamplesPerSec;
+ if (lavc_context->codec_id == CODEC_ID_AAC
+ && samplerate == 2 * container_samplerate)
+ broken_srate = true;
+ else if (container_samplerate)
+ samplerate = container_samplerate;
+
+ if (lavc_context->channels != sh_audio->channels ||
+ samplerate != sh_audio->samplerate ||
+ sample_format != sh_audio->sample_format) {
+ sh_audio->channels = lavc_context->channels;
+ sh_audio->samplerate = samplerate;
+ sh_audio->sample_format = sample_format;
+ sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
+ if (broken_srate)
+ mp_msg(MSGT_DECAUDIO, MSGL_WARN,
+ "Ignoring broken container sample rate for AAC with SBR\n");
+ return 1;
+ }
+ return 0;
+}
+
+static int init(sh_audio_t *sh_audio)
+{
+ struct MPOpts *opts = sh_audio->opts;
+ AVCodecContext *lavc_context;
+ AVCodec *lavc_codec;
+
+ if (sh_audio->codec->dll) {
+ lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
+ if (!lavc_codec) {
+ mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
+ "Cannot find codec '%s' in libavcodec...\n",
+ sh_audio->codec->dll);
+ return 0;
+ }
+ } else if (!sh_audio->libav_codec_id) {
+ mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "No Libav codec ID known. "
+ "Generic lavc decoder is not applicable.\n");
+ return 0;
+ } else {
+ lavc_codec = avcodec_find_decoder(sh_audio->libav_codec_id);
+ if (!lavc_codec) {
+ mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Libavcodec has no decoder "
+ "for this codec\n");
+ return 0;
+ }
+ }
+
+ sh_audio->codecname = lavc_codec->long_name;
+ if (!sh_audio->codecname)
+ sh_audio->codecname = lavc_codec->name;
+
+ struct priv *ctx = talloc_zero(NULL, struct priv);
+ sh_audio->context = ctx;
+ lavc_context = avcodec_alloc_context3(lavc_codec);
+ ctx->avctx = lavc_context;
+ ctx->avframe = avcodec_alloc_frame();
+
+ // Always try to set - option only exists for AC3 at the moment
+ av_opt_set_double(lavc_context, "drc_scale", opts->drc_level,
+ AV_OPT_SEARCH_CHILDREN);
+ lavc_context->sample_rate = sh_audio->samplerate;
+ lavc_context->bit_rate = sh_audio->i_bps * 8;
+ if (sh_audio->wf) {
+ lavc_context->channels = sh_audio->wf->nChannels;
+ lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
+ lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
+ lavc_context->block_align = sh_audio->wf->nBlockAlign;
+ lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
+ }
+ lavc_context->request_channels = opts->audio_output_channels;
+ lavc_context->codec_tag = sh_audio->format; //FOURCC
+ if (sh_audio->gsh->lavf_codec_tag)
+ lavc_context->codec_tag = sh_audio->gsh->lavf_codec_tag;
+ lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
+ lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
+
+ /* alloc extra data */
+ if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
+ lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
+ lavc_context->extradata_size = sh_audio->wf->cbSize;
+ memcpy(lavc_context->extradata, sh_audio->wf + 1,
+ lavc_context->extradata_size);
+ }
+
+ // for QDM2
+ if (sh_audio->codecdata_len && sh_audio->codecdata &&
+ !lavc_context->extradata) {
+ lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
+ FF_INPUT_BUFFER_PADDING_SIZE);
+ lavc_context->extradata_size = sh_audio->codecdata_len;
+ memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
+ lavc_context->extradata_size);
+ }
+
+ /* open it */
+ if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
+ mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
+ uninit(sh_audio);
+ return 0;
+ }
+ mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
+ lavc_codec->name);
+
+ if (sh_audio->format == 0x3343414D) {
+ // MACE 3:1
+ sh_audio->ds->ss_div = 2 * 3; // 1 samples/packet
+ sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
+ } else if (sh_audio->format == 0x3643414D) {
+ // MACE 6:1
+ sh_audio->ds->ss_div = 2 * 6; // 1 samples/packet
+ sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
+ }
+
+ // Decode at least 1 byte: (to get header filled)
+ for (int tries = 0;;) {
+ int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
+ sh_audio->a_buffer_size);
+ if (x > 0) {
+ sh_audio->a_buffer_len = x;
+ break;
+ }