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authorwm4 <wm4@nowhere>2012-11-05 17:02:04 +0100
committerwm4 <wm4@nowhere>2012-11-12 20:06:14 +0100
commitd4bdd0473d6f43132257c9fb3848d829755167a3 (patch)
tree8021c2f7da1841393c8c832105e20cd527826d6c /audio/decode
parentbd48deba77bd5582c5829d6fe73a7d2571088aba (diff)
downloadmpv-d4bdd0473d6f43132257c9fb3848d829755167a3.tar.bz2
mpv-d4bdd0473d6f43132257c9fb3848d829755167a3.tar.xz
Rename directories, move files (step 1 of 2) (does not compile)
Tis drops the silly lib prefixes, and attempts to organize the tree in a more logical way. Make the top-level directory less cluttered as well. Renames the following directories: libaf -> audio/filter libao2 -> audio/out libvo -> video/out libmpdemux -> demux Split libmpcodecs: vf* -> video/filter vd*, dec_video.* -> video/decode mp_image*, img_format*, ... -> video/ ad*, dec_audio.* -> audio/decode libaf/format.* is moved to audio/ - this is similar to how mp_image.* is located in video/. Move most top-level .c/.h files to core. (talloc.c/.h is left on top- level, because it's external.) Park some of the more annoying files in compat/. Some of these are relicts from the time mplayer used ffmpeg internals. sub/ is not split, because it's too much of a mess (subtitle code is mixed with OSD display and rendering). Maybe the organization of core is not ideal: it mixes playback core (like mplayer.c) and utility helpers (like bstr.c/h). Should the need arise, the playback core will be moved somewhere else, while core contains all helper and common code.
Diffstat (limited to 'audio/decode')
-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]);
+}
+
+/* This tries to extract a requested amount of decoded data.
+ * Even when you request 0 bytes, it will feed enough input so that
+ * the decoder _could_ have delivered something.
+ * Returns byte count >= 0, -1 on error.
+ *
+ * Thoughts on exact pts keeping:
+ * We have to assume that MPEG frames are cut in pieces by packet boundaries.
+ * Also, it might be possible that the first packet does not contain enough
+ * data to ensure initial stream sync... or re-sync on erroneous streams.
+ * So we need something robust to relate the decoded byte count to the correct
+ * time stamp. This is tricky, though. From the outside, you cannot tell if,
+ * after having fed two packets until the first output arrives, one should
+ * start counting from the first packet's pts or the second packet's.
+ * So, let's just count from the last fed package's pts. If the packets are
+ * exactly cut to MPEG frames, this will cause one frame mismatch in the
+ * beginning (when mpg123 peeks ahead for the following header), but will
+ * be corrected with the third frame already. One might add special code to
+ * not increment the base pts past the first packet's after a resync before
+ * the first decoded bytes arrived. */
+static int decode_a_bit(sh_audio_t *sh, unsigned char *buf, int count)
+{
+ int ret = MPG123_OK;
+ int got = 0;
+ struct ad_mpg123_context *con = sh->context;
+
+ /* There will be one MPG123_NEW_FORMAT message on first open.
+ * This will be handled in init(). */
+ do {
+ size_t got_now = 0;
+
+ /* Feed the decoder. This will only fire from the second round on. */
+ if (ret == MPG123_NEED_MORE) {
+ int incount;
+ double pts;
+ unsigned char *inbuf;
+ /* Feed more input data. */
+ incount = ds_get_packet_pts(sh->ds, &inbuf, &pts);
+ if (incount <= 0)
+ break; /* Apparently that's it. EOF. */
+
+ /* Next bytes from that presentation time. */
+ if (pts != MP_NOPTS_VALUE) {
+ sh->pts = pts;
+ sh->pts_bytes = 0;
+ }
+
+#ifdef AD_MPG123_FRAMEWISE
+ /* Have to use mpg123_feed() to avoid decoding here. */
+ ret = mpg123_feed(con->handle, inbuf, incount);
+#else
+ /* Do not use mpg123_feed(), added in later libmpg123 versions. */
+ ret = mpg123_decode(con->handle, inbuf, incount, NULL, 0, NULL);
+#endif
+ if (ret == MPG123_ERR)
+ break;
+ }
+ /* Theoretically, mpg123 could return MPG123_DONE, so be prepared.
+ * Should not happen in our usage, but it is a valid return code. */
+ else if (ret == MPG123_ERR || ret == MPG123_DONE)
+ break;
+
+ /* Try to decode a bit. This is the return value that counts
+ * for the loop condition. */
+#ifdef AD_MPG123_FRAMEWISE
+ if (!buf) { /* fake call just for feeding to get format */
+ ret = mpg123_getformat(con->handle, NULL, NULL, NULL);
+ } else { /* This is the decoding. One frame at a time. */
+ ret = mpg123_replace_buffer(con->handle, buf, count);
+ if (ret == MPG123_OK)
+ ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now);
+ }
+#else
+ ret = mpg123_decode(con->handle, NULL, 0, buf + got, count - got,
+ &got_now);
+#endif
+
+ got += got_now;
+ sh->pts_bytes += got_now;
+
+#ifdef AD_MPG123_FRAMEWISE
+ } while (ret == MPG123_NEED_MORE || (got == 0 && count != 0));
+#else
+ } while (ret == MPG123_NEED_MORE || got < count);
+#endif
+
+ if (ret == MPG123_ERR) {
+ mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 decoding failed: %s\n",
+ mpg123_strerror(con->handle));
+ mpg123_close(con->handle);
+ return -1;
+ }
+
+ return got;
+}
+
+/* Close, reopen stream. Feed data until we know the format of the stream.
+ * 1 on success, 0 on error */
+static int reopen_stream(sh_audio_t *sh)
+{
+ struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context;
+
+ mpg123_close(con->handle);
+ /* No resetting of the context:
+ * We do not want to loose the mean bitrate data. */
+
+ /* Open and make sure we have fed enough data to get stream properties. */
+ if (MPG123_OK == mpg123_open_feed(con->handle) &&
+ /* Feed data until mpg123 is ready (has found stream beginning). */
+ !decode_a_bit(sh, NULL, 0)) {
+ return 1;
+ } else {
+ mp_msg(MSGT_DECAUDIO, MSGL_ERR,
+ "mpg123 failed to reopen stream: %s\n",
+ mpg123_strerror(con->handle));
+ mpg123_close(con->handle);
+ return 0;
+ }
+}
+
+/* Now we really start accessing some data and determining file format.
+ * Paranoia note: The mpg123_close() on errors is not really necessary,
+ * But it ensures that we don't accidentally continue decoding with a
+ * bad state (possibly interpreting the format badly or whatnot). */
+static int init(sh_audio_t *sh)
+{
+ long rate = 0;
+ int channels = 0;
+ int encoding = 0;
+ mpg123_id3v2 *v2;
+ struct mpg123_frameinfo finfo;
+ struct ad_mpg123_context *con = sh->context;
+
+ /* We're open about any output format that libmpg123 will suggest.
+ * Note that a standard build will always default to 16 bit signed and
+ * the native sample rate of the file. */
+ if (MPG123_OK == mpg123_format_all(con->handle) &&
+ reopen_stream(sh) &&
+ MPG123_OK == mpg123_getformat(con->handle, &rate, &channels, &encoding) &&
+ /* Forbid the format to change later on. */
+ MPG123_OK == mpg123_format_none(con->handle) &&
+ MPG123_OK == mpg123_format(con->handle, rate, channels, encoding) &&
+ /* Get MPEG header info. */
+ MPG123_OK == mpg123_info(con->handle, &finfo) &&
+ /* Since we queried format, mpg123 should have read past ID3v2 tags.
+ * We need to decide if printing of UTF-8 encoded text info is wanted. */
+ MPG123_OK == mpg123_id3(con->handle, NULL, &v2)) {
+ /* If we are here, we passed all hurdles. Yay! Extract the info. */
+ print_header_compact(&finfo);
+ /* Do we want to print out the UTF-8 Id3v2 info?
+ if (v2)
+ print_id3v2(v2); */
+
+ /* Have kb/s, want B/s
+ * For VBR, the first frame will be a bad estimate. */
+ sh->i_bps = (finfo.bitrate ? finfo.bitrate : compute_bitrate(&finfo))
+ * 1000 / 8;
+#ifdef AD_MPG123_MEAN_BITRATE
+ con->delay = 1;
+ con->mean_rate = 0.;
+ con->mean_count = 0;
+#endif
+ con->vbr = (finfo.vbr != MPG123_CBR);
+ sh->channels = channels;
+ sh->samplerate = rate;
+ /* Without external force, mpg123 will always choose signed encoding,
+ * and non-16-bit only on builds that don't support it.
+ * Be reminded that it doesn't matter to the MPEG file what encoding
+ * is produced from it. */
+ switch (encoding) {
+ case MPG123_ENC_SIGNED_8:
+ sh->sample_format = AF_FORMAT_S8;
+ sh->samplesize = 1;
+ break;
+ case MPG123_ENC_SIGNED_16:
+ sh->sample_format = AF_FORMAT_S16_NE;
+ sh->samplesize = 2;
+ break;
+ /* To stay compatible with the oldest libmpg123 headers, do not rely
+ * on float and 32 bit encoding symbols being defined.
+ * Those formats came later */
+ case 0x1180: /* MPG123_ENC_SIGNED_32 */
+ sh->sample_format = AF_FORMAT_S32_NE;
+ sh->samplesize = 4;
+ break;
+ case 0x200: /* MPG123_ENC_FLOAT_32 */
+ sh->sample_format = AF_FORMAT_FLOAT_NE;
+ sh->samplesize = 4;
+ break;
+ default:
+ mp_msg(MSGT_DECAUDIO, MSGL_ERR,
+ "Bad encoding from mpg123: %i.\n", encoding);
+ mpg123_close(con->handle);
+ return 0;
+ }
+#ifdef AD_MPG123_FRAMEWISE
+ /* Going to decode directly to MPlayer's memory. It is important
+ * to have MPG123_AUTO_RESAMPLE disabled for the buffer size
+ * being an all-time limit. */
+ sh->audio_out_minsize = 1152 * 2 * sh->samplesize;
+#endif
+
+ return 1;
+ } else {
+ mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 init error: %s\n",
+ mpg123_strerror(con->handle));
+ mpg123_close(con->handle);
+ return 0;
+ }
+}
+
+static void uninit(sh_audio_t *sh)
+{
+ struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context;
+
+ mpg123_close(con->handle);
+ mpg123_delete(con->handle);
+ free(sh->context);
+ sh->context = NULL;
+ mpg123_exit();
+}
+
+#ifdef AD_MPG123_MEAN_BITRATE
+/* Update mean bitrate. This could be dropped if accurate time display
+ * on audio file playback is not desired. */
+static void update_info(sh_audio_t *sh)
+{
+ struct ad_mpg123_context *con = sh->context;
+ if (con->vbr && --con->delay < 1) {
+ struct mpg123_frameinfo finfo;
+ if (MPG123_OK == mpg123_info(con->handle, &finfo)) {
+ if (++con->mean_count > ((unsigned int) -1) / 2)
+ con->mean_count = ((unsigned int) -1) / 4;
+
+ /* Might not be numerically optimal, but works fine enough. */
+ con->mean_rate = ((con->mean_count - 1) * con->mean_rate +
+ finfo.bitrate) / con->mean_count;
+ sh->i_bps = (int) (con->mean_rate * 1000 / 8);
+
+ con->delay = 10;
+ }
+ }
+}
+#endif
+
+static int decode_audio(sh_audio_t *sh, unsigned char *buf, int minlen,
+ int maxlen)
+{
+ int bytes;
+
+ bytes = decode_a_bit(sh, buf, maxlen);
+ if (bytes == 0)
+ return -1; /* EOF */
+
+#ifdef AD_MPG123_MEAN_BITRATE
+ update_info(sh);
+#endif
+ return bytes;
+}
+
+static int control(sh_audio_t *sh, int cmd, void *arg, ...)
+{
+ switch (cmd) {
+ case ADCTRL_RESYNC_STREAM:
+ /* Close/reopen the stream for mpg123 to make sure it doesn't
+ * think that it still knows the exact stream position.
+ * Otherwise, we would have funny effects from the gapless code.
+ * Oh, and it helps to minimize artifacts from jumping in the stream. */
+ if (reopen_stream(sh)) {
+#ifdef AD_MPG123_MEAN_BITRATE
+ update_info(sh);
+#endif
+ return CONTROL_TRUE;
+ } else {
+ /* MPlayer ignores this case! It just keeps on decoding.
+ * So we have to make sure resync never fails ... */
+ mp_msg(MSGT_DECAUDIO, MSGL_ERR,
+ "mpg123 cannot reopen stream for resync.\n");
+ return CONTROL_FALSE;
+ }
+ break;
+ }
+ return CONTROL_UNKNOWN;
+}
diff --git a/audio/decode/ad_pcm.c b/audio/decode/ad_pcm.c
new file mode 100644
index 0000000000..c265dfcd56
--- /dev/null
+++ b/audio/decode/ad_pcm.c
@@ -0,0 +1,220 @@
+/*
+ * 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 <libavutil/common.h>
+
+#include "talloc.h"
+#include "config.h"
+#include "ad_internal.h"
+#include "libaf/format.h"
+#include "libaf/reorder_ch.h"
+
+static const ad_info_t info = {
+ "Uncompressed PCM audio decoder",
+ "pcm",
+ "Nick Kurshev",
+ "A'rpi",
+ ""
+};
+
+struct ad_pcm_context {
+ unsigned char *buffer;
+ int buffer_pos;
+ int buffer_len;
+ int buffer_size;
+};
+
+LIBAD_EXTERN(pcm)
+
+static int init(sh_audio_t * sh_audio)
+{
+ WAVEFORMATEX *h = sh_audio->wf;
+ if (!h)
+ return 0;
+ sh_audio->i_bps = h->nAvgBytesPerSec;
+ sh_audio->channels = h->nChannels;
+ sh_audio->samplerate = h->nSamplesPerSec;
+ sh_audio->samplesize = (h->wBitsPerSample + 7) / 8;
+ sh_audio->sample_format = AF_FORMAT_S16_LE; // default
+ switch (sh_audio->format) { /* hardware formats: */
+ case 0x0:
+ case 0x1: // Microsoft PCM
+ case 0xfffe: // Extended
+ switch (sh_audio->samplesize) {
+ case 1: sh_audio->sample_format = AF_FORMAT_U8; break;
+ case 2: sh_audio->sample_format = AF_FORMAT_S16_LE; break;
+ case 3: sh_audio->sample_format = AF_FORMAT_S24_LE; break;
+ case 4: sh_audio->sample_format = AF_FORMAT_S32_LE; break;
+ }
+ break;
+ case 0x3: // IEEE float
+ sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
+ break;
+ case 0x6: sh_audio->sample_format = AF_FORMAT_A_LAW; break;
+ case 0x7: sh_audio->sample_format = AF_FORMAT_MU_LAW; break;
+ case 0x11: sh_audio->sample_format = AF_FORMAT_IMA_ADPCM; break;
+ case 0x50: sh_audio->sample_format = AF_FORMAT_MPEG2; break;
+/* case 0x2000: sh_audio->sample_format=AFMT_AC3; */
+ case 0x20776172: // 'raw '
+ sh_audio->sample_format = AF_FORMAT_S16_BE;
+ if (sh_audio->samplesize == 1)
+ sh_audio->sample_format = AF_FORMAT_U8;
+ break;
+ case 0x736F7774: // 'twos'
+ sh_audio->sample_format = AF_FORMAT_S16_BE;
+ // intended fall-through
+ case 0x74776F73: // 'sowt'
+ if (sh_audio->samplesize == 1)
+ sh_audio->sample_format = AF_FORMAT_S8;
+ break;
+ case 0x32336c66: // 'fl32', bigendian float32
+ case 0x32334C46: // 'FL32', bigendian float32 in aiff
+ sh_audio->sample_format = AF_FORMAT_FLOAT_BE;
+ sh_audio->samplesize = 4;
+ break;
+ case 0x666c3332: // '23lf', little endian float32, MPlayer internal fourCC
+ case 0x6D63706C: // 'lpcm'
+ sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
+ sh_audio->samplesize = 4;
+ break;
+/* case 0x34366c66: // 'fl64', bigendian float64
+ sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
+ sh_audio->samplesize=8;
+ break;
+ case 0x666c3634: // '46lf', little endian float64, MPlayer internal fourCC
+ sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
+ sh_audio->samplesize=8;
+ break;*/
+ case 0x34326e69: // 'in24', bigendian int24
+ sh_audio->sample_format = AF_FORMAT_S24_BE;
+ sh_audio->samplesize = 3;
+ break;
+ case 0x696e3234: // '42ni', little endian int24, MPlayer internal fourCC
+ sh_audio->sample_format = AF_FORMAT_S24_LE;
+ sh_audio->samplesize = 3;
+ break;
+ case 0x32336e69: // 'in32', bigendian int32
+ sh_audio->sample_format = AF_FORMAT_S32_BE;
+ sh_audio->samplesize = 4;
+ break;
+ case 0x696e3332: // '23ni', little endian int32, MPlayer internal fourCC
+ sh_audio->sample_format = AF_FORMAT_S32_LE;
+ sh_audio->samplesize = 4;
+ break;
+ case MKTAG('M', 'P', 'a', 'f'):
+ sh_audio->sample_format = h->wFormatTag;
+ sh_audio->samplesize = (af_fmt2bits(sh_audio->sample_format) + 7) / 8;
+ break;
+ default:
+ if (sh_audio->samplesize != 2)
+ sh_audio->sample_format = AF_FORMAT_U8;
+ }
+ if (!sh_audio->samplesize) // this would cause MPlayer to hang later
+ sh_audio->samplesize = 2;
+ sh_audio->context = talloc_zero(NULL, struct ad_pcm_context);
+ return 1;
+}
+
+static int preinit(sh_audio_t *sh)
+{
+ sh->audio_out_minsize = 2048;
+ return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+ talloc_free(sh->context);
+}
+
+static int control(sh_audio_t *sh, int cmd, void *arg, ...)
+{
+ struct ad_pcm_context *ctx = sh->context;
+ int skip;
+ switch (cmd) {
+ case ADCTRL_RESYNC_STREAM:
+ ctx->buffer_len = 0;
+ return true;
+ 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 unitsize = sh_audio->channels * sh_audio->samplesize;
+ minlen = (minlen + unitsize - 1) / unitsize * unitsize;
+ if (minlen > maxlen)
+ // if someone needs hundreds of channels adjust audio_out_minsize
+ // based on channels in preinit()
+ return -1;
+
+ int len = 0;
+ struct ad_pcm_context *ctx = sh_audio->context;
+ while (len < minlen) {
+ if (ctx->buffer_len - ctx->buffer_pos <= 0) {
+ double pts;
+ unsigned char *ptr;
+ int plen = ds_get_packet_pts(sh_audio->ds, &ptr, &pts);
+ if (plen < 0)
+ break;
+ if (ctx->buffer_size < plen) {
+ talloc_free(ctx->buffer);
+ ctx->buffer = talloc_size(ctx, plen);
+ ctx->buffer_size = plen;
+ }
+ memcpy(ctx->buffer, ptr, plen);
+ ctx->buffer_len = plen;
+ ctx->buffer_pos = 0;
+ if (pts != MP_NOPTS_VALUE) {
+ sh_audio->pts = pts;
+ sh_audio->pts_bytes = 0;
+ }
+ }
+ int from_stored = ctx->buffer_len - ctx->buffer_pos;
+ if (from_stored > minlen - len)
+ from_stored = minlen - len;
+ memcpy(buf + len, ctx->buffer + ctx->buffer_pos, from_stored);
+ ctx->buffer_pos += from_stored;
+ sh_audio->pts_bytes += from_stored;
+ len += from_stored;
+ }
+ if (len % unitsize) {
+ mp_msg(MSGT_DECAUDIO, MSGL_WARN, "[ad_pcm] discarding partial sample "
+ "at end\n");
+ len -= len % unitsize;
+ }
+ if (len == 0)
+ len = -1; // The loop above only exits at error/EOF
+ if (len > 0 && sh_audio->channels >= 5) {
+ reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
+ AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
+ sh_audio->channels, len / sh_audio->samplesize,
+ sh_audio->samplesize);
+ }
+ return len;
+}
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
new file mode 100644
index 0000000000..877bc99317
--- /dev/null
+++ b/audio/decode/ad_spdif.c
@@ -0,0 +1,310 @@
+/*
+ * 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 <string.h>
+
+#include <libavformat/avformat.h>
+#include <libavcodec/avcodec.h>
+#include <libavutil/opt.h>
+
+#include "config.h"
+#include "mp_msg.h"
+#include "ad_internal.h"
+
+static const ad_info_t info = {
+ "libavformat/spdifenc audio pass-through decoder.",
+ "spdif",
+ "Naoya OYAMA",
+ "Naoya OYAMA",
+ "For ALL hardware decoders"
+};
+
+LIBAD_EXTERN(spdif)
+
+#define FILENAME_SPDIFENC "spdif"
+#define OUTBUF_SIZE 65536
+struct spdifContext {
+ AVFormatContext *lavf_ctx;
+ int iec61937_packet_size;
+ int out_buffer_len;
+ int out_buffer_size;
+ uint8_t *out_buffer;
+ uint8_t pb_buffer[OUTBUF_SIZE];
+};
+
+static int read_packet(void *p, uint8_t *buf, int buf_size)
+{
+ // spdifenc does not use read callback.
+ return 0;
+}
+
+static int write_packet(void *p, uint8_t *buf, int buf_size)
+{
+ int len;
+ struct spdifContext *ctx = p;
+
+ len = FFMIN(buf_size, ctx->out_buffer_size -ctx->out_buffer_len);
+ memcpy(&ctx->out_buffer[ctx->out_buffer_len], buf, len);
+ ctx->out_buffer_len += len;
+ return len;
+}
+
+static int64_t seek(void *p, int64_t offset, int whence)
+{
+ // spdifenc does not use seek callback.
+ return 0;
+}
+
+static int preinit(sh_audio_t *sh)
+{
+ sh->samplesize = 2;
+ return 1;
+}
+
+static int init(sh_audio_t *sh)
+{
+ int i, x, in_size, srate, bps, *dtshd_rate;
+ unsigned char *start;
+ double pts;
+ static const struct {
+ const char *name; enum CodecID id;
+ } fmt_id_type[] = {
+ { "aac" , CODEC_ID_AAC },
+ { "ac3" , CODEC_ID_AC3 },
+ { "dca" , CODEC_ID_DTS },
+ { "eac3", CODEC_ID_EAC3 },
+ { "mpa" , CODEC_ID_MP3 },
+ { "thd" , CODEC_ID_TRUEHD },
+ { NULL , 0 }
+ };
+ AVFormatContext *lavf_ctx = NULL;
+ AVStream *stream = NULL;
+ const AVOption *opt = NULL;
+ struct spdifContext *spdif_ctx = NULL;
+
+ spdif_ctx = av_mallocz(sizeof(*spdif_ctx));
+ if (!spdif_ctx)
+ goto fail;
+ spdif_ctx->lavf_ctx = avformat_alloc_context();
+ if (!spdif_ctx->lavf_ctx)
+ goto fail;
+
+ sh->context = spdif_ctx;
+ lavf_ctx = spdif_ctx->lavf_ctx;
+
+ lavf_ctx->oformat = av_guess_format(FILENAME_SPDIFENC, NULL, NULL);
+ if (!lavf_ctx->oformat)
+ goto fail;
+ lavf_ctx->priv_data = av_mallocz(lavf_ctx->oformat->priv_data_size);
+ if (!lavf_ctx->priv_data)
+ goto fail;
+ lavf_ctx->pb = avio_alloc_context(spdif_ctx->pb_buffer, OUTBUF_SIZE, 1, spdif_ctx,
+ read_packet, write_packet, seek);
+ if (!lavf_ctx->pb)
+ goto fail;
+ stream = avformat_new_stream(lavf_ctx, 0);
+ if (!stream)
+ goto fail;
+ lavf_ctx->duration = AV_NOPTS_VALUE;
+ lavf_ctx->start_time = AV_NOPTS_VALUE;
+ for (i = 0; fmt_id_type[i].name; i++) {
+ if (!strcmp(sh->codec->dll, fmt_id_type[i].name)) {
+ lavf_ctx->streams[0]->codec->codec_id = fmt_id_type[i].id;
+ break;
+ }
+ }
+ lavf_ctx->raw_packet_buffer_remaining_size = RAW_PACKET_BUFFER_SIZE;
+ if (AVERROR_PATCHWELCOME == lavf_ctx->oformat->write_header(lavf_ctx)) {
+ mp_msg(MSGT_DECAUDIO,MSGL_INFO,
+ "This codec is not supported by spdifenc.\n");
+ goto fail;
+ }
+
+ // get sample_rate & bitrate from parser
+ bps = srate = 0;
+ x = ds_get_packet_pts(sh->ds, &start, &pts);
+ in_size = x;
+ if (x <= 0) {
+ pts = MP_NOPTS_VALUE;
+ x = 0;
+ }
+ ds_parse(sh->ds, &start, &x, pts, 0);
+ if (x == 0) { // not enough buffer
+ srate = 48000; //fake value
+ bps = 768000/8; //fake value
+ } else if (sh->avctx) {
+ if (sh->avctx->sample_rate < 44100) {
+ mp_msg(MSGT_DECAUDIO,MSGL_INFO,
+ "This stream sample_rate[%d Hz] may be broken. "
+ "Force reset 48000Hz.\n",
+ sh->avctx->sample_rate);
+ srate = 48000; //fake value
+ } else
+ srate = sh->avctx->sample_rate;
+ bps = sh->avctx->bit_rate/8;
+ }
+ sh->ds->buffer_pos -= in_size;
+
+ switch (lavf_ctx->streams[0]->codec->codec_id) {
+ case CODEC_ID_AAC:
+ spdif_ctx->iec61937_packet_size = 16384;
+ sh->sample_format = AF_FORMAT_IEC61937_LE;
+ sh->samplerate = srate;
+ sh->channels = 2;
+ sh->i_bps = bps;
+ break;
+ case CODEC_ID_AC3:
+ spdif_ctx->iec61937_packet_size = 6144;
+ sh->sample_format = AF_FORMAT_IEC61937_LE;
+ sh->samplerate = srate;
+ sh->channels = 2;
+ sh->i_bps = bps;
+ break;
+ case CODEC_ID_DTS: // FORCE USE DTS-HD
+ opt = av_opt_find(&lavf_ctx->oformat->priv_class,
+ "dtshd_rate", NULL, 0, 0);
+ if (!opt)
+ goto fail;
+ dtshd_rate = (int*)(((uint8_t*)lavf_ctx->priv_data) +
+ opt->offset);
+ *dtshd_rate = 192000*4;
+ spdif_ctx->iec61937_packet_size = 32768;
+ sh->sample_format = AF_FORMAT_IEC61937_LE;
+ sh->samplerate = 192000; // DTS core require 48000
+ sh->channels = 2*4;
+ sh->i_bps = bps;
+ break;
+ case CODEC_ID_EAC3:
+ spdif_ctx->iec61937_packet_size = 24576;
+ sh->sample_format = AF_FORMAT_IEC61937_LE;
+ sh->samplerate = 192000;
+ sh->channels = 2;
+ sh->i_bps = bps;
+ break;
+ case CODEC_ID_MP3:
+ spdif_ctx->iec61937_packet_size = 4608;
+ sh->sample_format = AF_FORMAT_MPEG2;
+ sh->samplerate = srate;
+ sh->channels = 2;
+ sh->i_bps = bps;
+ break;
+ case CODEC_ID_TRUEHD:
+ spdif_ctx->iec61937_packet_size = 61440;
+ sh->sample_format = AF_FORMAT_IEC61937_LE;
+ sh->samplerate = 192000;
+ sh->channels = 8;
+ sh->i_bps = bps;
+ break;
+ default:
+ break;
+ }
+
+ return 1;
+
+fail:
+ uninit(sh);
+ return 0;
+}
+
+static int decode_audio(sh_audio_t *sh, unsigned char *buf,
+ int minlen, int maxlen)
+{
+ struct spdifContext *spdif_ctx = sh->context;
+ AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
+ AVPacket pkt;
+ double pts;
+ int ret, in_size, consumed, x;
+ unsigned char *start = NULL;
+
+ consumed = spdif_ctx->out_buffer_len = 0;
+ spdif_ctx->out_buffer_size = maxlen;
+ spdif_ctx->out_buffer = buf;
+ while (spdif_ctx->out_buffer_len + spdif_ctx->iec61937_packet_size < maxlen
+ && spdif_ctx->out_buffer_len < minlen) {
+ if (sh->ds->eof)
+ break;
+ x = ds_get_packet_pts(sh->ds, &start, &pts);
+ if (x <= 0) {
+ x = 0;
+ ds_parse(sh->ds, &start, &x, MP_NOPTS_VALUE, 0);
+ if (x == 0)
+ continue; // END_NOT_FOUND
+ in_size = x;
+ } else {
+ in_size = x;
+ consumed = ds_parse(sh->ds, &start, &x, pts, 0);
+ if (x == 0) {
+ mp_msg(MSGT_DECAUDIO,MSGL_V,
+ "start[%p] in_size[%d] consumed[%d] x[%d].\n",
+ start, in_size, consumed, x);
+ continue; // END_NOT_FOUND
+ }
+ sh->ds->buffer_pos -= in_size - consumed;
+ }
+ av_init_packet(&pkt);
+ pkt.data = start;
+ pkt.size = x;
+ mp_msg(MSGT_DECAUDIO,MSGL_V,
+ "start[%p] pkt.size[%d] in_size[%d] consumed[%d] x[%d].\n",
+ start, pkt.size, in_size, consumed, x);
+ if (pts != MP_NOPTS_VALUE) {
+ sh->pts = pts;
+ sh->pts_bytes = 0;
+ }
+ ret = lavf_ctx->oformat->write_packet(lavf_ctx, &pkt);
+ if (ret < 0)
+ break;
+ }
+ sh->pts_bytes += spdif_ctx->out_buffer_len;
+ return spdif_ctx->out_buffer_len;
+}
+
+static int control(sh_audio_t *sh, int cmd, void* arg, ...)
+{
+ unsigned char *start;
+ double pts;
+
+ switch (cmd) {
+ case ADCTRL_RESYNC_STREAM:
+ case ADCTRL_SKIP_FRAME:
+ ds_get_packet_pts(sh->ds, &start, &pts);
+ return CONTROL_TRUE;
+ }
+ return CONTROL_UNKNOWN;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+ struct spdifContext *spdif_ctx = sh->context;
+ AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
+
+ if (lavf_ctx) {
+ if (lavf_ctx->oformat)
+ lavf_ctx->oformat->write_trailer(lavf_ctx);
+ av_freep(&lavf_ctx->pb);
+ if (lavf_ctx->streams) {
+ av_freep(&lavf_ctx->streams[0]->codec);
+ av_freep(&lavf_ctx->streams[0]->info);
+ av_freep(&lavf_ctx->streams[0]);
+ }
+ av_freep(&lavf_ctx->streams);
+ av_freep(&lavf_ctx->priv_data);
+ }
+ av_freep(&lavf_ctx);
+ av_freep(&spdif_ctx);
+}
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
new file mode 100644
index 0000000000..2602352e52
--- /dev/null
+++ b/audio/decode/dec_audio.c
@@ -0,0 +1,462 @@
+/*
+ * 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 <assert.h>
+
+#include "config.h"
+#include "mp_msg.h"
+#include "bstr.h"
+
+#include "stream/stream.h"
+#include "libmpdemux/demuxer.h"
+
+#include "codec-cfg.h"
+#include "libmpdemux/stheader.h"
+
+#include "dec_audio.h"
+#include "ad.h"
+#include "libaf/format.h"
+
+#include "libaf/af.h"
+
+int fakemono = 0;
+
+struct af_cfg af_cfg = { 1, NULL }; // Configuration for audio filters
+
+void afm_help(void)
+{
+ int i;
+ mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Available (compiled-in) audio codec families/drivers:\n");
+ mp_msg(MSGT_IDENTIFY, MSGL_INFO, "ID_AUDIO_DRIVERS\n");
+ mp_msg(MSGT_DECAUDIO, MSGL_INFO, " afm: info: (comment)\n");
+ for (i = 0; mpcodecs_ad_drivers[i] != NULL; i++)
+ if (mpcodecs_ad_drivers[i]->info->comment
+ && mpcodecs_ad_drivers[i]->info->comment[0])
+ mp_msg(MSGT_DECAUDIO, MSGL_INFO, "%9s %s (%s)\n",
+ mpcodecs_ad_drivers[i]->info->short_name,
+ mpcodecs_ad_drivers[i]->info->name,
+ mpcodecs_ad_drivers[i]->info->comment);
+ else
+ mp_msg(MSGT_DECAUDIO, MSGL_INFO, "%9s %s\n",
+ mpcodecs_ad_drivers[i]->info->short_name,
+ mpcodecs_ad_drivers[i]->info->name);
+}
+
+static int init_audio_codec(sh_audio_t *sh_audio)
+{
+ assert(!sh_audio->initialized);
+ resync_audio_stream(sh_audio);
+ if ((af_cfg.force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT) {
+ int fmt = AF_FORMAT_FLOAT_NE;
+ if (sh_audio->ad_driver->control(sh_audio, ADCTRL_QUERY_FORMAT,
+ &fmt) == CONTROL_TRUE) {
+ sh_audio->sample_format = fmt;
+ sh_audio->samplesize = 4;
+ }
+ }
+ sh_audio->audio_out_minsize = 8192; // default, preinit() may change it
+ if (!sh_audio->ad_driver->preinit(sh_audio)) {
+ mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "ADecoder preinit failed :(\n");
+ return 0;
+ }
+
+ /* allocate audio in buffer: */
+ if (sh_audio->audio_in_minsize > 0) {
+ sh_audio->a_in_buffer_size = sh_audio->audio_in_minsize;
+ mp_tmsg(MSGT_DECAUDIO, MSGL_V, "dec_audio: Allocating %d bytes for input buffer.\n",
+ sh_audio->a_in_buffer_size);
+ sh_audio->a_in_buffer = av_mallocz(sh_audio->a_in_buffer_size);
+ }
+
+ const int base_size = 65536;
+ // At least 64 KiB plus rounding up to next decodable unit size
+ sh_audio->a_buffer_size = base_size + sh_audio->audio_out_minsize;
+
+ mp_tmsg(MSGT_DECAUDIO, MSGL_V, "dec_audio: Allocating %d + %d = %d bytes for output buffer.\n",
+ sh_audio->audio_out_minsize, base_size, sh_audio->a_buffer_size);
+
+ sh_audio->a_buffer = av_mallocz(sh_audio->a_buffer_size);
+ if (!sh_audio->a_buffer)
+ abort();
+ sh_audio->a_buffer_len = 0;
+
+ if (!sh_audio->ad_driver->init(sh_audio)) {
+ mp_tmsg(MSGT_DECAUDIO, MSGL_V, "ADecoder init failed :(\n");
+ uninit_audio(sh_audio); // free buffers
+ return 0;
+ }
+
+ sh_audio->initialized = 1;
+
+ if (!sh_audio->channels || !sh_audio->samplerate) {
+ mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder did not specify "
+ "audio format!\n");
+ uninit_audio(sh_audio); // free buffers
+ return 0;
+ }
+
+ if (!sh_audio->o_bps)
+ sh_audio->o_bps = sh_audio->channels * sh_audio->samplerate
+ * sh_audio->samplesize;
+ return 1;
+}
+
+static int init_audio(sh_audio_t *sh_audio, char *codecname, char *afm,
+ int status, stringset_t *selected)
+{
+ int force = 0;
+ if (codecname && codecname[0] == '+') {
+ codecname = &codecname[1];
+ force = 1;
+ }
+ sh_audio->codec = NULL;
+ while (1) {
+ const ad_functions_t *mpadec;
+ sh_audio->ad_driver = 0;
+ if (!(sh_audio->codec = find_audio_codec(sh_audio->format,
+ NULL,
+ sh_audio->codec, force)))
+ break;
+ // ok we found one codec
+ if (stringset_test(selected, sh_audio->codec->name))
+ continue; // already tried & failed
+ if (codecname && strcmp(sh_audio->codec->name, codecname))
+ continue; // -ac
+ if (afm && strcmp(sh_audio->codec->drv, afm))
+ continue; // afm doesn't match
+ if (!force && sh_audio->codec->status < status)
+ continue; // too unstable
+ stringset_add(selected, sh_audio->codec->name); // tagging it
+ // ok, it matches all rules, let's find the driver!
+ int i;
+ for (i = 0; mpcodecs_ad_drivers[i] != NULL; i++)
+ if (!strcmp(mpcodecs_ad_drivers[i]->info->short_name,
+ sh_audio->codec->drv))
+ break;
+ mpadec = mpcodecs_ad_drivers[i];
+ if (!mpadec) { // driver not available (==compiled in)
+ mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
+ "Requested audio codec family [%s] (afm=%s) not available.\nEnable it at compilation.\n",
+ sh_audio->codec->name, sh_audio->codec->drv);
+ continue;
+ }
+ // it's available, let's try to init!
+ // init()
+ mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Opening audio decoder: [%s] %s\n",
+ mpadec->info->short_name, mpadec->info->name);
+ sh_audio->ad_driver = mpadec;
+ if (!init_audio_codec(sh_audio)) {
+ mp_tmsg(MSGT_DECAUDIO, MSGL_WARN, "Audio decoder init failed for "
+ "codecs.conf entry \"%s\".\n", sh_audio->codec->name);
+ continue; // try next...
+ }
+ // Yeah! We got it!
+ return 1;
+ }
+ return 0;
+}
+
+int init_best_audio_codec(sh_audio_t *sh_audio, char **audio_codec_list,
+ char **audio_fm_list)
+{
+ stringset_t selected;
+ char *ac_l_default[2] = { "", (char *) NULL };
+ // hack:
+ if (!audio_codec_list)
+ audio_codec_list = ac_l_default;
+ // Go through the codec.conf and find the best codec...
+ sh_audio->initialized = 0;
+ stringset_init(&selected);
+ while (!sh_audio->initialized && *audio_codec_list) {
+ char *audio_codec = *(audio_codec_list++);
+ if (audio_codec[0]) {
+ if (audio_codec[0] == '-') {
+ // disable this codec:
+ stringset_add(&selected, audio_codec + 1);
+ } else {
+ // forced codec by name:
+ mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Forced audio codec: %s\n",
+ audio_codec);
+ init_audio(sh_audio, audio_codec, NULL, -1, &selected);
+ }
+ } else {
+ int status;
+ // try in stability order: UNTESTED, WORKING, BUGGY.
+ // never try CRASHING.
+ if (audio_fm_list) {
+ char **fmlist = audio_fm_list;
+ // try first the preferred codec families:
+ while (!sh_audio->initialized && *fmlist) {
+ char *audio_fm = *(fmlist++);
+ mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Trying to force audio codec driver family %s...\n",
+ audio_fm);
+ for (status = CODECS_STATUS__MAX;
+ status >= CODECS_STATUS__MIN; --status)
+ if (init_audio(sh_audio, NULL, audio_fm, status, &selected))
+ break;
+ }
+ }
+ if (!sh_audio->initialized)
+ for (status = CODECS_STATUS__MAX; status >= CODECS_STATUS__MIN;
+ --status)
+ if (init_audio(sh_audio, NULL, NULL, status, &selected))
+ break;
+ }
+ }
+ stringset_free(&selected);
+
+ if (!sh_audio->initialized) {
+ mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Cannot find codec for audio format 0x%X.\n",
+ sh_audio->format);
+ return 0; // failed
+ }
+
+ mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Selected audio codec: %s [%s]\n",
+ sh_audio->codecname ? sh_audio->codecname : sh_audio->codec->info,
+ sh_audio->ad_driver->info->print_name ?
+ sh_audio->ad_driver->info->print_name :
+ sh_audio->ad_driver->info->short_name);
+ mp_tmsg(MSGT_DECAUDIO, MSGL_V,
+ "Audio codecs.conf entry: %s (%s) afm: %s\n",
+ sh_audio->codec->name, sh_audio->codec->info, sh_audio->codec->drv);
+ mp_msg(MSGT_DECAUDIO, MSGL_V,
+ "AUDIO: %d Hz, %d ch, %s, %3.1f kbit/%3.2f%% (ratio: %d->%d)\n",
+ sh_audio->samplerate, sh_audio->channels,
+ af_fmt2str_short(sh_audio->sample_format),
+ sh_audio->i_bps * 8 * 0.001,
+ ((float) sh_audio->i_bps / sh_audio->o_bps) * 100.0,
+ sh_audio->i_bps, sh_audio->o_bps);
+ mp_msg(MSGT_IDENTIFY, MSGL_INFO,
+ "ID_AUDIO_BITRATE=%d\nID_AUDIO_RATE=%d\n" "ID_AUDIO_NCH=%d\n",
+ sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels);
+
+ return 1; // success
+}
+
+void uninit_audio(sh_audio_t *sh_audio)
+{
+ if (sh_audio->afilter) {
+ mp_msg(MSGT_DECAUDIO, MSGL_V, "Uninit audio filters...\n");
+ af_uninit(sh_audio->afilter);
+ free(sh_audio->afilter);
+ sh_audio->afilter = NULL;
+ }
+ if (sh_audio->initialized) {
+ mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Uninit audio: %s\n",
+ sh_audio->codec->drv);
+ sh_audio->ad_driver->uninit(sh_audio);
+ sh_audio->initialized = 0;
+ }
+ av_freep(&sh_audio->a_buffer);
+ av_freep(&sh_audio->a_in_buffer);
+}
+
+
+int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
+ int *out_samplerate, int *out_channels, int *out_format)
+{
+ struct af_stream *afs = sh_audio->afilter;
+ if (!afs) {
+ afs = calloc(1, sizeof(struct af_stream));
+ afs->opts = sh_audio->opts;
+ }
+ // input format: same as codec's output format:
+ afs->input.rate = in_samplerate;
+ afs->input.nch = sh_audio->channels;
+ afs->input.format = sh_audio->sample_format;
+ af_fix_parameters(&(afs->input));
+
+ // output format: same as ao driver's input format (if missing, fallback to input)
+ afs->output.rate = *out_samplerate;
+ afs->output.nch = *out_channels;
+ afs->output.format = *out_format;
+ af_fix_parameters(&(afs->output));
+
+ // filter config:
+ memcpy(&afs->cfg, &af_cfg, sizeof(struct af_cfg));
+
+ mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Building audio filter chain for %dHz/%dch/%s -> %dHz/%dch/%s...\n",
+ afs->input.rate, afs->input.nch,
+ af_fmt2str_short(afs->input.format), afs->output.rate,
+ afs->output.nch, af_fmt2str_short(afs->output.format));
+
+ // let's autoprobe it!
+ if (0 != af_init(afs)) {
+ sh_audio->afilter = NULL;
+ free(afs);
+ return 0; // failed :(
+ }
+
+ *out_samplerate = afs->output.rate;
+ *out_channels = afs->output.nch;
+ *out_format = afs->output.format;
+
+ // ok!
+ sh_audio->afilter = (void *) afs;
+ return 1;
+}
+
+static void set_min_out_buffer_size(struct bstr *outbuf, int len)
+{
+ size_t oldlen = talloc_get_size(outbuf->start);
+ if (oldlen < len) {
+ assert(outbuf->start); // talloc context should be already set
+ mp_msg(MSGT_DECAUDIO, MSGL_V, "Increasing filtered audio buffer size "
+ "from %zd to %d\n", oldlen, len);
+ outbuf->start = talloc_realloc_size(NULL, outbuf->start, len);
+ }
+}
+
+static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
+{
+ assert(len-1 + sh->audio_out_minsize <= sh->a_buffer_size);
+
+ int error = 0;
+
+ // Decode more bytes if needed
+ int old_samplerate = sh->samplerate;
+ int old_channels = sh->channels;
+ int old_sample_format = sh->sample_format;
+ while (sh->a_buffer_len < len) {
+ unsigned char *buf = sh->a_buffer + sh->a_buffer_len;
+ int minlen = len - sh->a_buffer_len;
+ int maxlen = sh->a_buffer_size - sh->a_buffer_len;
+ int ret = sh->ad_driver->decode_audio(sh, buf, minlen, maxlen);
+ int format_change = sh->samplerate != old_samplerate
+ || sh->channels != old_channels
+ || sh->sample_format != old_sample_format;
+ if (ret <= 0 || format_change) {
+ error = format_change ? -2 : -1;
+ // samples from format-changing call get discarded too
+ len = sh->a_buffer_len;
+ break;
+ }
+ sh->a_buffer_len += ret;
+ }
+
+ // Filter
+ struct mp_audio filter_input = {
+ .audio = sh->a_buffer,
+ .len = len,
+ .rate = sh->samplerate,
+ .nch = sh->channels,
+ .format = sh->sample_format
+ };
+ af_fix_parameters(&filter_input);
+ struct mp_audio *filter_output = af_play(sh->afilter, &filter_input);
+ if (!filter_output)
+ return -1;
+ set_min_out_buffer_size(outbuf, outbuf->len + filter_output->len);
+ memcpy(outbuf->start + outbuf->len, filter_output->audio,
+ filter_output->len);
+ outbuf->len += filter_output->len;
+
+ // remove processed data from decoder buffer:
+ sh->a_buffer_len -= len;
+ memmove(sh->a_buffer, sh->a_buffer + len, sh->a_buffer_len);
+
+ return error;
+}
+
+/* Try to get at least minlen decoded+filtered bytes in outbuf
+ * (total length including possible existing data).
+ * Return 0 on success, -1 on error/EOF (not distinguished).
+ * In the former case outbuf->len is always >= minlen on return.
+ * In case of EOF/error it might or might not be.
+ * Outbuf.start must be talloc-allocated, and will be reallocated
+ * if needed to fit all filter output. */
+int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen)
+{
+ // Indicates that a filter seems to be buffering large amounts of data
+ int huge_filter_buffer = 0;
+ // Decoded audio must be cut at boundaries of this many bytes
+ int unitsize = sh_audio->channels * sh_audio->samplesize * 16;
+
+ /* Filter output size will be about filter_multiplier times input size.
+ * If some filter buffers audio in big blocks this might only hold
+ * as average over time. */
+ double filter_multiplier = af_calc_filter_multiplier(sh_audio->afilter);
+
+ /* If the decoder set audio_out_minsize then it can do the equivalent of
+ * "while (output_len < target_len) output_len += audio_out_minsize;",
+ * so we must guarantee there is at least audio_out_minsize-1 bytes
+ * more space in the output buffer than the minimum length we try to
+ * decode. */
+ int max_decode_len = sh_audio->a_buffer_size - sh_audio->audio_out_minsize;
+ if (!unitsize)
+ return -1;
+ max_decode_len -= max_decode_len % unitsize;
+
+ while (outbuf->len < minlen) {
+ int declen = (minlen - outbuf->len) / filter_multiplier
+ + (unitsize << 5); // some extra for possible filter buffering
+ if (huge_filter_buffer)
+ /* Some filter must be doing significant buffering if the estimated
+ * input length didn't produce enough output from filters.
+ * Feed the filters 2k bytes at a time until we have enough output.
+ * Very small amounts could make filtering inefficient while large
+ * amounts can make MPlayer demux the file unnecessarily far ahead
+ * to get audio data and buffer video frames in memory while doing
+ * so. However the performance impact of either is probably not too
+ * significant as long as the value is not completely insane. */
+ declen = 2000;
+ declen -= declen % unitsize;
+ if (declen > max_decode_len)
+ declen = max_decode_len;
+ else
+ /* if this iteration does not fill buffer, we must have lots
+ * of buffering in filters */
+ huge_filter_buffer = 1;
+ int res = filter_n_bytes(sh_audio, outbuf, declen);
+ if (res < 0)
+ return res;
+ }
+ return 0;
+}
+
+void decode_audio_prepend_bytes(struct bstr *outbuf, int count, int byte)
+{
+ set_min_out_buffer_size(outbuf, outbuf->len + count);
+ memmove(outbuf->start + count, outbuf->start, outbuf->len);
+ memset(outbuf->start, byte, count);
+ outbuf->len += count;
+}
+
+
+void resync_audio_stream(sh_audio_t *sh_audio)
+{
+ sh_audio->a_in_buffer_len = 0; // clear audio input buffer
+ sh_audio->pts = MP_NOPTS_VALUE;
+ if (!sh_audio->initialized)
+ return;
+ sh_audio->ad_driver->control(sh_audio, ADCTRL_RESYNC_STREAM, NULL);
+}
+
+void skip_audio_frame(sh_audio_t *sh_audio)
+{
+ if (!sh_audio->initialized)
+ return;
+ if (sh_audio->ad_driver->control(sh_audio, ADCTRL_SKIP_FRAME, NULL) ==
+ CONTROL_TRUE)
+ return;
+ // default skip code:
+ ds_fill_buffer(sh_audio->ds); // skip block
+}
diff --git a/audio/decode/dec_audio.h b/audio/decode/dec_audio.h
new file mode 100644
index 0000000000..0d4baf0666
--- /dev/null
+++ b/audio/decode/dec_audio.h
@@ -0,0 +1,38 @@
+/*
+ * 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_DEC_AUDIO_H
+#define MPLAYER_DEC_AUDIO_H
+
+#include "libmpdemux/stheader.h"
+
+struct bstr;
+
+// dec_audio.c:
+void afm_help(void);
+int init_best_audio_codec(sh_audio_t *sh_audio, char** audio_codec_list, char** audio_fm_list);
+int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen);
+void decode_audio_prepend_bytes(struct bstr *outbuf, int count, int byte);
+void resync_audio_stream(sh_audio_t *sh_audio);
+void skip_audio_frame(sh_audio_t *sh_audio);
+void uninit_audio(sh_audio_t *sh_audio);
+
+int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
+ int *out_samplerate, int *out_channels, int *out_format);
+
+#endif /* MPLAYER_DEC_AUDIO_H */