making act segmenter

1 files changed, 181 insertions, 0 deletions

diff --git a/ffmpeg/libavformat/westwood_aud.c b/ffmpeg/libavformat/westwood_aud.c
new file mode 100644
index 0000000..d666117
--- /dev/null
+++ b/ffmpeg/libavformat/westwood_aud.c
@@ -0,0 +1,181 @@
+/*
+ * Westwood Studios AUD Format Demuxer
+ * Copyright (c) 2003 The ffmpeg Project
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Westwood Studios AUD file demuxer
+ * by Mike Melanson (melanson@pcisys.net)
+ * for more information on the Westwood file formats, visit:
+ *   http://www.pcisys.net/~melanson/codecs/
+ *   http://www.geocities.com/SiliconValley/8682/aud3.txt
+ *
+ * Implementation note: There is no definite file signature for AUD files.
+ * The demuxer uses a probabilistic strategy for content detection. This
+ * entails performing sanity checks on certain header values in order to
+ * qualify a file. Refer to wsaud_probe() for the precise parameters.
+ */
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/intreadwrite.h"
+#include "avformat.h"
+#include "internal.h"
+
+#define AUD_HEADER_SIZE 12
+#define AUD_CHUNK_PREAMBLE_SIZE 8
+#define AUD_CHUNK_SIGNATURE 0x0000DEAF
+
+static int wsaud_probe(AVProbeData *p)
+{
+    int field;
+
+    /* Probabilistic content detection strategy: There is no file signature
+     * so perform sanity checks on various header parameters:
+     *   8000 <= sample rate (16 bits) <= 48000  ==> 40001 acceptable numbers
+     *   flags <= 0x03 (2 LSBs are used)         ==> 4 acceptable numbers
+     *   compression type (8 bits) = 1 or 99     ==> 2 acceptable numbers
+     *   first audio chunk signature (32 bits)   ==> 1 acceptable number
+     * The number space contains 2^64 numbers. There are 40001 * 4 * 2 * 1 =
+     * 320008 acceptable number combinations.
+     */
+
+    if (p->buf_size < AUD_HEADER_SIZE + AUD_CHUNK_PREAMBLE_SIZE)
+        return 0;
+
+    /* check sample rate */
+    field = AV_RL16(&p->buf[0]);
+    if ((field < 8000) || (field > 48000))
+        return 0;
+
+    /* enforce the rule that the top 6 bits of this flags field are reserved (0);
+     * this might not be true, but enforce it until deemed unnecessary */
+    if (p->buf[10] & 0xFC)
+        return 0;
+
+    if (p->buf[11] != 99 && p->buf[11] != 1)
+        return 0;
+
+    /* read ahead to the first audio chunk and validate the first header signature */
+    if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE)
+        return 0;
+
+    /* return 1/2 certainty since this file check is a little sketchy */
+    return AVPROBE_SCORE_MAX / 2;
+}
+
+static int wsaud_read_header(AVFormatContext *s)
+{
+    AVIOContext *pb = s->pb;
+    AVStream *st;
+    unsigned char header[AUD_HEADER_SIZE];
+    int sample_rate, channels, codec;
+
+    if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
+        return AVERROR(EIO);
+
+    sample_rate = AV_RL16(&header[0]);
+    channels    = (header[10] & 0x1) + 1;
+    codec       = header[11];
+
+    /* initialize the audio decoder stream */
+    st = avformat_new_stream(s, NULL);
+    if (!st)
+        return AVERROR(ENOMEM);
+
+    switch (codec) {
+    case  1:
+        if (channels != 1) {
+            avpriv_request_sample(s, "Stereo WS-SND1");
+            return AVERROR_PATCHWELCOME;
+        }
+        st->codec->codec_id = AV_CODEC_ID_WESTWOOD_SND1;
+        break;
+    case 99:
+        st->codec->codec_id = AV_CODEC_ID_ADPCM_IMA_WS;
+        st->codec->bits_per_coded_sample = 4;
+        st->codec->bit_rate = channels * sample_rate * 4;
+        break;
+    default:
+        avpriv_request_sample(s, "Unknown codec: %d", codec);
+        return AVERROR_PATCHWELCOME;
+    }
+    avpriv_set_pts_info(st, 64, 1, sample_rate);
+    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
+    st->codec->channels    = channels;
+    st->codec->channel_layout = channels == 1 ? AV_CH_LAYOUT_MONO :
+                                                AV_CH_LAYOUT_STEREO;
+    st->codec->sample_rate = sample_rate;
+
+    return 0;
+}
+
+static int wsaud_read_packet(AVFormatContext *s,
+                             AVPacket *pkt)
+{
+    AVIOContext *pb = s->pb;
+    unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
+    unsigned int chunk_size;
+    int ret = 0;
+    AVStream *st = s->streams[0];
+
+    if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
+        AUD_CHUNK_PREAMBLE_SIZE)
+        return AVERROR(EIO);
+
+    /* validate the chunk */
+    if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
+        return AVERROR_INVALIDDATA;
+
+    chunk_size = AV_RL16(&preamble[0]);
+
+    if (st->codec->codec_id == AV_CODEC_ID_WESTWOOD_SND1) {
+        /* For Westwood SND1 audio we need to add the output size and input
+           size to the start of the packet to match what is in VQA.
+           Specifically, this is needed to signal when a packet should be
+           decoding as raw 8-bit pcm or variable-size ADPCM. */
+        int out_size = AV_RL16(&preamble[2]);
+        if ((ret = av_new_packet(pkt, chunk_size + 4)))
+            return ret;
+        if ((ret = avio_read(pb, &pkt->data[4], chunk_size)) != chunk_size)
+            return ret < 0 ? ret : AVERROR(EIO);
+        AV_WL16(&pkt->data[0], out_size);
+        AV_WL16(&pkt->data[2], chunk_size);
+
+        pkt->duration = out_size;
+    } else {
+        ret = av_get_packet(pb, pkt, chunk_size);
+        if (ret != chunk_size)
+            return AVERROR(EIO);
+
+        /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
+        pkt->duration = (chunk_size * 2) / st->codec->channels;
+    }
+    pkt->stream_index = st->index;
+
+    return ret;
+}
+
+AVInputFormat ff_wsaud_demuxer = {
+    .name           = "wsaud",
+    .long_name      = NULL_IF_CONFIG_SMALL("Westwood Studios audio"),
+    .read_probe     = wsaud_probe,
+    .read_header    = wsaud_read_header,
+    .read_packet    = wsaud_read_packet,
+};