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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
10 files changed, 2244 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;
+ }
+ if (++tries >= 5) {
+ mp_msg(MSGT_DECAUDIO, MSGL_ERR,
+ "ad_ffmpeg: initial decode failed\n");
+ uninit(sh_audio);
+ return 0;
+ }
+ }
+
+ sh_audio->i_bps = lavc_context->bit_rate / 8;
+ if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
+ sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
+
+ switch (av_get_packed_sample_fmt(lavc_context->sample_fmt)) {
+ case AV_SAMPLE_FMT_U8:
+ case AV_SAMPLE_FMT_S16:
+ case AV_SAMPLE_FMT_S32:
+ case AV_SAMPLE_FMT_FLT:
+ break;
+ default:
+ uninit(sh_audio);
+ return 0;
+ }
+ return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+ sh->codecname = NULL;
+ struct priv *ctx = sh->context;
+ if (!ctx)
+ return;
+ AVCodecContext *lavc_context = ctx->avctx;
+
+ if (lavc_context) {
+ if (avcodec_close(lavc_context) < 0)
+ mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
+ av_freep(&lavc_context->extradata);
+ av_freep(&lavc_context);
+ }
+ avcodec_free_frame(&ctx->avframe);
+ talloc_free(ctx);
+ sh->context = NULL;
+}
+
+static int control(sh_audio_t *sh, int cmd, void *arg, ...)
+{
+ struct priv *ctx = sh->context;
+ switch (cmd) {
+ case ADCTRL_RESYNC_STREAM:
+ avcodec_flush_buffers(ctx->avctx);
+ ds_clear_parser(sh->ds);
+ ctx->previous_data_left = 0;
+ ctx->output_left = 0;
+ return CONTROL_TRUE;
+ }
+ return CONTROL_UNKNOWN;
+}
+
+static av_always_inline void deplanarize(struct sh_audio *sh)
+{
+ struct priv *priv = sh->context;
+
+ size_t bps = av_get_bytes_per_sample(priv->avctx->sample_fmt);
+ size_t nb_samples = priv->avframe->nb_samples;
+ size_t channels = priv->avctx->channels;
+ size_t size = bps * nb_samples * channels;
+
+ if (talloc_get_size(priv->output_packed) != size)
+ priv->output_packed =
+ talloc_realloc_size(priv, priv->output_packed, size);
+
+ size_t offset = 0;
+ unsigned char *output_ptr = priv->output_packed;
+ unsigned char **src = priv->avframe->data;
+
+ for (size_t s = 0; s < nb_samples; s++) {
+ for (size_t c = 0; c < channels; c++) {
+ memcpy(output_ptr, src[c] + offset, bps);
+ output_ptr += bps;
+ }
+ offset += bps;
+ }
+
+ priv->output = priv->output_packed;
+}
+
+static int decode_new_packet(struct sh_audio *sh)
+{
+ struct priv *priv = sh->context;
+ AVCodecContext *avctx = priv->avctx;
+ double pts = MP_NOPTS_VALUE;
+ int insize;
+ bool packet_already_used = priv->previous_data_left;
+ struct demux_packet *mpkt = ds_get_packet2(sh->ds,
+ priv->previous_data_left);
+ unsigned char *start;
+ if (!mpkt) {
+ assert(!priv->previous_data_left);
+ start = NULL;
+ insize = 0;
+ ds_parse(sh->ds, &start, &insize, pts, 0);
+ if (insize <= 0)
+ return -1; // error or EOF
+ } else {
+ assert(mpkt->len >= priv->previous_data_left);
+ if (!priv->previous_data_left) {
+ priv->previous_data_left = mpkt->len;
+ pts = mpkt->pts;
+ }
+ insize = priv->previous_data_left;
+ start = mpkt->buffer + mpkt->len - priv->previous_data_left;
+ int consumed = ds_parse(sh->ds, &start, &insize, pts, 0);
+ priv->previous_data_left -= consumed;
+ priv->previous_data_left = FFMAX(priv->previous_data_left, 0);
+ }
+
+ AVPacket pkt;
+ av_init_packet(&pkt);
+ pkt.data = start;
+ pkt.size = insize;
+ if (mpkt && mpkt->avpacket) {
+ pkt.side_data = mpkt->avpacket->side_data;
+ pkt.side_data_elems = mpkt->avpacket->side_data_elems;
+ }
+ if (pts != MP_NOPTS_VALUE && !packet_already_used) {
+ sh->pts = pts;
+ sh->pts_bytes = 0;
+ }
+ int got_frame = 0;
+ int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
+ // LATM may need many packets to find mux info
+ if (ret == AVERROR(EAGAIN))
+ return 0;
+ if (ret < 0) {
+ mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n");
+ return -1;
+ }
+ // The "insize >= ret" test is sanity check against decoder overreads
+ if (!sh->parser && insize >= ret)
+ priv->previous_data_left = insize - ret;
+ if (!got_frame)
+ return 0;
+ uint64_t unitsize = (uint64_t)av_get_bytes_per_sample(avctx->sample_fmt) *
+ avctx->channels;
+ if (unitsize > 100000)
+ abort();
+ priv->unitsize = unitsize;
+ uint64_t output_left = unitsize * priv->avframe->nb_samples;
+ if (output_left > 500000000)
+ abort();
+ priv->output_left = output_left;
+ if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1) {
+ deplanarize(sh);
+ } else {
+ priv->output = priv->avframe->data[0];
+ }
+ mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d \n", insize,
+ priv->output_left);
+ return 0;
+}
+
+
+static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
+ int maxlen)
+{
+ struct priv *priv = sh_audio->context;
+ AVCodecContext *avctx = priv->avctx;
+
+ int len = -1;
+ while (len < minlen) {
+ if (!priv->output_left) {
+ if (decode_new_packet(sh_audio) < 0)
+ break;
+ continue;
+ }
+ if (setup_format(sh_audio, avctx))
+ return len;
+ int size = (minlen - len + priv->unitsize - 1);
+ size -= size % priv->unitsize;
+ size = FFMIN(size, priv->output_left);
+ if (size > maxlen)
+ abort();
+ memcpy(buf, priv->output, size);
+ priv->output += size;
+ priv->output_left -= size;
+ if (avctx->channels >= 5) {
+ int samplesize = av_get_bytes_per_sample(avctx->sample_fmt);
+ reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
+ AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
+ avctx->channels,
+ size / samplesize, samplesize);
+ }
+ if (len < 0)
+ len = size;
+ else
+ len += size;
+ buf += size;
+ maxlen -= size;
+ sh_audio->pts_bytes += size;
+ }
+ return len;
+}
diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c
new file mode 100644
index 0000000000..a3ce2cdcf6
--- /dev/null
+++ b/audio/decode/ad_mpg123.c
@@ -0,0 +1,489 @@
+/*
+ * MPEG 1.0/2.0/2.5 audio layer I, II, III decoding with libmpg123
+ *
+ * Copyright (C) 2010-2012 Thomas Orgis <thomas@orgis.org>
+ *
+ * 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 "ad_internal.h"
+
+static const ad_info_t info = {
+ "MPEG 1.0/2.0/2.5 layers I, II, III",
+ "mpg123",
+ "Thomas Orgis",
+ "mpg123.org",
+ "High-performance decoder using libmpg123."
+};
+
+LIBAD_EXTERN(mpg123)
+
+/* Reducing the ifdeffery to two main variants:
+ * 1. most compatible to any libmpg123 version
+ * 2. fastest variant with recent libmpg123 (>=1.14)
+ * Running variant 2 on older libmpg123 versions may work in
+ * principle, but is not supported.
+ * So, please leave the check for MPG123_API_VERSION there, m-kay?
+ */
+#include <mpg123.h>
+
+/* Enable faster mode of operation with newer libmpg123, avoiding
+ * unnecessary memcpy() calls. */
+#if (defined MPG123_API_VERSION) && (MPG123_API_VERSION >= 33)
+#define AD_MPG123_FRAMEWISE
+#endif
+
+/* Switch for updating bitrate info of VBR files. Not essential. */
+#define AD_MPG123_MEAN_BITRATE
+
+/* Funny thing, that. I assume I shall use it for selecting mpg123 channels.
+ * Please correct me if I guessed wrong. */
+extern int fakemono;
+
+struct ad_mpg123_context {
+ mpg123_handle *handle;
+#ifdef AD_MPG123_MEAN_BITRATE
+ /* Running mean for bit rate, stream length estimation. */
+ float mean_rate;
+ unsigned int mean_count;
+ /* Time delay for updates. */
+ short delay;
+#endif
+ /* If the stream is actually VBR. */
+ char vbr;
+};
+
+/* This initializes libmpg123 and prepares the handle, including funky
+ * parameters. */
+static int preinit(sh_audio_t *sh)
+{
+ int err, flag;
+ struct ad_mpg123_context *con;
+ /* Assumption: You always call preinit + init + uninit, on every file.
+ * But you stop at preinit in case it fails.
+ * If that is not true, one must ensure not to call mpg123_init / exit
+ * twice in a row. */
+ if (mpg123_init() != MPG123_OK)
+ return 0;
+
+ sh->context = malloc(sizeof(struct ad_mpg123_context));
+ con = sh->context;
+ /* Auto-choice of optimized decoder (first argument NULL). */
+ con->handle = mpg123_new(NULL, &err);
+ if (!con->handle)
+ goto bad_end;
+
+ /* Guessing here: Default value triggers forced upmix of mono to stereo. */
+ flag = fakemono == 0 ? MPG123_FORCE_STEREO :
+ fakemono == 1 ? MPG123_MONO_LEFT :
+ fakemono == 2 ? MPG123_MONO_RIGHT : 0;
+ if (mpg123_param(con->handle, MPG123_ADD_FLAGS, flag, 0.0) != MPG123_OK)
+ goto bad_end;
+
+ /* Basic settings.
+ * Don't spill messages, enable better resync with non-seekable streams.
+ * Give both flags individually without error checking to keep going with
+ * old libmpg123. Generally, it is not fatal if the flags are not
+ * honored */
+ mpg123_param(con->handle, MPG123_ADD_FLAGS, MPG123_QUIET, 0.0);
+ /* Do not bail out on malformed streams at all.
+ * MPlayer does not handle a decoder throwing the towel on crappy input. */
+ mpg123_param(con->handle, MPG123_RESYNC_LIMIT, -1, 0.0);
+
+ /* Open decisions: Configure libmpg123 to force encoding (or stay open about
+ * library builds that support only float or int32 output), (de)configure
+ * gapless decoding (won't work with seeking in MPlayer, though).
+ * Don't forget to eventually enable ReplayGain/RVA support, too.
+ * Let's try to run with the default for now. */
+
+ /* That would produce floating point output.
+ * You can get 32 and 24 bit ints, even 8 bit via format matrix. */
+ /* mpg123_param(con->handle, MPG123_ADD_FLAGS, MPG123_FORCE_FLOAT, 0.); */
+
+ /* Example for RVA choice (available since libmpg123 1.0.0):
+ mpg123_param(con->handle, MPG123_RVA, MPG123_RVA_MIX, 0.0) */
+
+#ifdef AD_MPG123_FRAMEWISE
+ /* Prevent funky automatic resampling.
+ * This way, we can be sure that one frame will never produce
+ * more than 1152 stereo samples. */
+ mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_AUTO_RESAMPLE, 0.);
+#else
+ /* Older mpg123 is vulnerable to concatenated streams when gapless cutting
+ * is enabled (will only play the jingle of a badly constructed radio
+ * stream). The versions using framewise decoding are fine with that. */
+ mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_GAPLESS, 0.);
+#endif
+
+ return 1;
+
+ bad_end:
+ if (!con->handle)
+ mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 preinit error: %s\n",
+ mpg123_plain_strerror(err));
+ else
+ mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 preinit error: %s\n",
+ mpg123_strerror(con->handle));
+
+ if (con->handle)
+ mpg123_delete(con->handle);
+ mpg123_exit();
+ free(sh->context);
+ sh->context = NULL;
+ return 0;
+}
+
+/* Compute bitrate from frame size. */
+static int compute_bitrate(struct mpg123_frameinfo *i)
+{
+ static const int samples_per_frame[4][4] = {
+ {-1, 384, 1152, 1152}, /* MPEG 1 */
+ {-1, 384, 1152, 576}, /* MPEG 2 */
+ {-1, 384, 1152, 576}, /* MPEG 2.5 */
+ {-1, -1, -1, -1}, /* Unknown */
+ };
+ return (int) ((i->framesize + 4) * 8 * i->rate * 0.001 /
+ samples_per_frame[i->version][i->layer] + 0.5);
+}
+
+/* Opted against the header printout from old mp3lib, too much
+ * irrelevant info. This is modelled after the mpg123 app's
+ * standard output line.
+ * If more verbosity is demanded, one can add more detail and
+ * also throw in ID3v2 info which libmpg123 collects anyway. */
+static void print_header_compact(struct mpg123_frameinfo *i)
+{
+ static const char *smodes[5] = {
+ "stereo", "joint-stereo", "dual-channel", "mono", "invalid"
+ };
+ static const char *layers[4] = {
+ "Unknown", "I", "II", "III"
+ };
+ static const char *versions[4] = {
+ "1.0", "2.0", "2.5", "x.x"
+ };
+
+ mp_msg(MSGT_DECAUDIO, MSGL_V, "MPEG %s layer %s, ",
+ versions[i->version], layers[i->layer]);
+ switch (i->vbr) {
+ case MPG123_CBR:
+ if (i->bitrate)
+ mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s", i->bitrate);
+ else
+ mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s (free format)",
+ compute_bitrate(i));
+ break;
+ case MPG123_VBR:
+ mp_msg(MSGT_DECAUDIO, MSGL_V, "VBR");
+ break;
+ case MPG123_ABR:
+ mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s ABR", i->abr_rate);
+ break;
+ default:
+ mp_msg(MSGT_DECAUDIO, MSGL_V, "???");
+ }
+ mp_msg(MSGT_DECAUDIO, MSGL_V, ", %ld Hz %s\n", i->rate,
+ smodes[i->mode]);