libmpcodecs: add ad_spdif.c, S/PDIF passthrough decoder

patch by Naoya OYAMA, naoya.oyama gmail com

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@34191 b3059339-0415-0410-9bf9-f77b7e298cf2

fix ad_spdif

Call av_register_all() before initialising the SPDIF muxer.

Fixes playback with -demuxer mpegts -ac spdifac3.
Patch by Naoya OYAMA, naoya D oyama gmail

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@34291 b3059339-0415-0410-9bf9-f77b7e298cf2

Use new API avformat_new_stream() instead of the deprecated
av_new_stream().

Patch by Naoya OYAMA, naoya D oyama gmail

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@34292 b3059339-0415-0410-9bf9-f77b7e298cf2

Cosmetics: Remove empty statement.

Patch by Naoya OYAMA, naoya D oyama gmail

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@34293 b3059339-0415-0410-9bf9-f77b7e298cf2

Use init_avformat() instead of av_register_all().

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@34294 b3059339-0415-0410-9bf9-f77b7e298cf2

Author: diego

2 files changed, 312 insertions, 0 deletions

diff --git a/libmpcodecs/ad.c b/libmpcodecs/ad.c
index dce8f9f710..2e67bd23af 100644
--- a/libmpcodecs/ad.c
+++ b/libmpcodecs/ad.c
@@ -49,6 +49,7 @@ extern const ad_functions_t mpcodecs_ad_libmad;
 extern const ad_functions_t mpcodecs_ad_realaud;
 extern const ad_functions_t mpcodecs_ad_libdv;
 extern const ad_functions_t mpcodecs_ad_qtaudio;
+extern const ad_functions_t mpcodecs_ad_spdif;
 extern const ad_functions_t mpcodecs_ad_libdca;
 
 const ad_functions_t * const mpcodecs_ad_drivers[] =
@@ -62,6 +63,7 @@ const ad_functions_t * const mpcodecs_ad_drivers[] =
   &mpcodecs_ad_hwac3,
   &mpcodecs_ad_hwmpa,
   &mpcodecs_ad_ffmpeg,
+  &mpcodecs_ad_spdif,
   &mpcodecs_ad_pcm,
   &mpcodecs_ad_dvdpcm,
   &mpcodecs_ad_alaw,
diff --git a/libmpcodecs/ad_spdif.c b/libmpcodecs/ad_spdif.c
new file mode 100644
index 0000000000..f6536178bc
--- /dev/null
+++ b/libmpcodecs/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] pkt.size[%d] in_size[%d] consumed[%d] x[%d].\n",
+                       start, pkt.size, 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);
+}