@@ -85,7 +85,7 @@ OBJS-$(CONFIG_FOURXM_DEMUXER)            += 4xm.o

 OBJS-$(CONFIG_FRAMECRC_MUXER)            += framecrcenc.o

 OBJS-$(CONFIG_FRAMEMD5_MUXER)            += md5enc.o

 OBJS-$(CONFIG_GIF_MUXER)                 += gif.o

-OBJS-$(CONFIG_GSM_DEMUXER)               += rawdec.o

+OBJS-$(CONFIG_GSM_DEMUXER)               += gsmdec.o

 OBJS-$(CONFIG_GXF_DEMUXER)               += gxf.o

 OBJS-$(CONFIG_GXF_MUXER)                 += gxfenc.o audiointerleave.o

 OBJS-$(CONFIG_G722_DEMUXER)              += rawdec.o

new file mode 100644

@@ -0,0 +1,135 @@

+/*

+ * RAW GSM demuxer

+ * Copyright (c) 2011 Justin Ruggles

+ *

+ * This file is part of Libav.

+ *

+ * Libav 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.

+ *

+ * Libav 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 Libav; if not, write to the Free Software

+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

+ */

+

+#include "libavutil/mathematics.h"

+#include "libavutil/opt.h"

+#include "avformat.h"

+

+#define GSM_BLOCK_SIZE    33

+#define GSM_BLOCK_SAMPLES 160

+#define GSM_SAMPLE_RATE   8000

+

+typedef struct {

+    AVClass *class;

+    int sample_rate;

+} GSMDemuxerContext;

+

+static int gsm_read_packet(AVFormatContext *s, AVPacket *pkt)

+{

+    int ret, size;

+

+    size = GSM_BLOCK_SIZE * 32;

+

+    if (av_new_packet(pkt, size) < 0)

+        return AVERROR(ENOMEM);

+

+    pkt->pos = avio_tell(s->pb);

+    pkt->stream_index = 0;

+

+    ret = av_get_packet(s->pb, pkt, size);

+    if (ret < GSM_BLOCK_SIZE) {

+        av_free_packet(pkt);

+        return ret < 0 ? ret : AVERROR(EIO);

+    }

+    pkt->size     = ret;

+    pkt->duration = ret      / GSM_BLOCK_SIZE;

+    pkt->pts      = pkt->pos / GSM_BLOCK_SIZE;

+

+    return 0;

+}

+

+static int gsm_read_header(AVFormatContext *s, AVFormatParameters *ap)

+{

+    GSMDemuxerContext *c = s->priv_data;

+    AVStream *st = avformat_new_stream(s, NULL);

+    if (!st)

+        return AVERROR(ENOMEM);

+

+    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;

+    st->codec->codec_id    = s->iformat->value;

+    st->codec->channels    = 1;

+    st->codec->sample_rate = c->sample_rate;

+    st->codec->block_align = GSM_BLOCK_SIZE;

+    st->codec->bit_rate    = GSM_BLOCK_SIZE * 8 * c->sample_rate / GSM_BLOCK_SAMPLES;

+

+    av_set_pts_info(st, 64, GSM_BLOCK_SAMPLES, GSM_SAMPLE_RATE);

+

+    return 0;

+}

+

+static int gsm_read_seek2(AVFormatContext *s, int stream_index, int64_t min_ts,

+                          int64_t ts, int64_t max_ts, int flags)

+{

+    GSMDemuxerContext *c = s->priv_data;

+

+    /* convert timestamps to file positions */

+    if (!(flags & AVSEEK_FLAG_BYTE)) {

+        if (stream_index < 0) {

+            AVRational bitrate_q = { GSM_BLOCK_SAMPLES, c->sample_rate * GSM_BLOCK_SIZE };

+            ts     = av_rescale_q(ts,     AV_TIME_BASE_Q, bitrate_q);

+            min_ts = av_rescale_q(min_ts, AV_TIME_BASE_Q, bitrate_q);

+            max_ts = av_rescale_q(max_ts, AV_TIME_BASE_Q, bitrate_q);

+        } else {

+            ts     *= GSM_BLOCK_SIZE;

+            min_ts *= GSM_BLOCK_SIZE;

+            max_ts *= GSM_BLOCK_SIZE;

+        }

+    }

+    /* round to nearest block boundary */

+    ts = (ts + GSM_BLOCK_SIZE / 2) / GSM_BLOCK_SIZE * GSM_BLOCK_SIZE;

+    ts = FFMAX(0, ts);

+

+    /* handle min/max */

+    while (ts < min_ts)

+        ts += GSM_BLOCK_SIZE;

+    while (ts > max_ts)

+        ts -= GSM_BLOCK_SIZE;

+    if (ts < min_ts || ts > max_ts)

+        return -1;

+

+    return avio_seek(s->pb, ts, SEEK_SET);

+}

+

+static const AVOption options[] = {

+    { "sample_rate", "", offsetof(GSMDemuxerContext, sample_rate),

+       AV_OPT_TYPE_INT, {.dbl = GSM_SAMPLE_RATE}, 1, INT_MAX / GSM_BLOCK_SIZE,

+       AV_OPT_FLAG_DECODING_PARAM },

+    { NULL },

+};

+

+static const AVClass class = {

+    .class_name = "gsm demuxer",

+    .item_name  = av_default_item_name,

+    .option     = options,

+    .version    = LIBAVUTIL_VERSION_INT,

+};

+

+AVInputFormat ff_gsm_demuxer = {

+    .name           = "gsm",

+    .long_name      = NULL_IF_CONFIG_SMALL("raw GSM"),

+    .priv_data_size = sizeof(GSMDemuxerContext),

+    .read_header    = gsm_read_header,

+    .read_packet    = gsm_read_packet,

+    .read_seek2     = gsm_read_seek2,

+    .extensions     = "gsm",

+    .value          = CODEC_ID_GSM,

+    .priv_class     = &class,

+};

@@ -186,18 +186,6 @@ AVInputFormat ff_g722_demuxer = {

 };

 #endif

-#if CONFIG_GSM_DEMUXER

-AVInputFormat ff_gsm_demuxer = {

-    .name           = "gsm",

-    .long_name      = NULL_IF_CONFIG_SMALL("raw GSM"),

-    .read_header    = ff_raw_audio_read_header,

-    .read_packet    = ff_raw_read_partial_packet,

-    .flags= AVFMT_GENERIC_INDEX,

-    .extensions = "gsm",

-    .value = CODEC_ID_GSM,

-};

-#endif

-

 #if CONFIG_LATM_DEMUXER

 AVInputFormat ff_latm_demuxer = {

     .name           = "latm",