@@ -44,7 +44,6 @@

 typedef struct DPCMContext {

     AVFrame frame;

-    int channels;

     int16_t roq_square_array[256];

     int sample[2];                  ///< previous sample (for SOL_DPCM)

     const int8_t *sol_table;        ///< delta table for SOL_DPCM

@@ -123,7 +122,6 @@ static av_cold int dpcm_decode_init(AVCodecContext *avctx)

         return AVERROR(EINVAL);

     }

-    s->channels = avctx->channels;

     s->sample[0] = s->sample[1] = 0;

     switch(avctx->codec->id) {

@@ -179,7 +177,7 @@ static int dpcm_decode_frame(AVCodecContext *avctx, void *data,

     int out = 0, ret;

     int predictor[2];

     int ch = 0;

-    int stereo = s->channels - 1;

+    int stereo = avctx->channels - 1;

     int16_t *output_samples, *samples_end;

     GetByteContext gb;

@@ -193,10 +191,10 @@ static int dpcm_decode_frame(AVCodecContext *avctx, void *data,

         out = buf_size - 8;

         break;

     case AV_CODEC_ID_INTERPLAY_DPCM:

-        out = buf_size - 6 - s->channels;

+        out = buf_size - 6 - avctx->channels;

         break;

     case AV_CODEC_ID_XAN_DPCM:

-        out = buf_size - 2 * s->channels;

+        out = buf_size - 2 * avctx->channels;

         break;

     case AV_CODEC_ID_SOL_DPCM:

         if (avctx->codec_tag != 3)

@@ -209,12 +207,12 @@ static int dpcm_decode_frame(AVCodecContext *avctx, void *data,

         av_log(avctx, AV_LOG_ERROR, "packet is too small\n");

         return AVERROR(EINVAL);

     }

-    if (out % s->channels) {

+    if (out % avctx->channels) {

         av_log(avctx, AV_LOG_WARNING, "channels have differing number of samples\n");

     }

     /* get output buffer */

-    s->frame.nb_samples = (out + s->channels - 1) / s->channels;

+    s->frame.nb_samples = (out + avctx->channels - 1) / avctx->channels;

     if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {

         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

         return ret;

@@ -248,7 +246,7 @@ static int dpcm_decode_frame(AVCodecContext *avctx, void *data,

     case AV_CODEC_ID_INTERPLAY_DPCM:

         bytestream2_skipu(&gb, 6);  /* skip over the stream mask and stream length */

-        for (ch = 0; ch < s->channels; ch++) {

+        for (ch = 0; ch < avctx->channels; ch++) {

             predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);

             *output_samples++ = predictor[ch];

         }

@@ -268,7 +266,7 @@ static int dpcm_decode_frame(AVCodecContext *avctx, void *data,

     {

         int shift[2] = { 4, 4 };

-        for (ch = 0; ch < s->channels; ch++)

+        for (ch = 0; ch < avctx->channels; ch++)

             predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);

         ch = 0;

@@ -19,6 +19,7 @@

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

  */

+#include "libavutil/audioconvert.h"

 #include "libavutil/crc.h"

 #include "libavutil/log.h"

 #include "bytestream.h"

@@ -28,6 +29,15 @@

 static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 };

+static const int64_t flac_channel_layouts[6] = {

+    AV_CH_LAYOUT_MONO,

+    AV_CH_LAYOUT_STEREO,

+    AV_CH_LAYOUT_SURROUND,

+    AV_CH_LAYOUT_QUAD,

+    AV_CH_LAYOUT_5POINT0,

+    AV_CH_LAYOUT_5POINT1

+};

+

 static int64_t get_utf8(GetBitContext *gb)

 {

     int64_t val;

@@ -181,6 +191,14 @@ int avpriv_flac_is_extradata_valid(AVCodecContext *avctx,

     return 1;

 }

+void ff_flac_set_channel_layout(AVCodecContext *avctx)

+{

+    if (avctx->channels <= FF_ARRAY_ELEMS(flac_channel_layouts))

+        avctx->channel_layout = flac_channel_layouts[avctx->channels - 1];

+    else

+        avctx->channel_layout = 0;

+}

+

 void avpriv_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,

                               const uint8_t *buffer)

 {

@@ -205,6 +223,7 @@ void avpriv_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *

     avctx->channels = s->channels;

     avctx->sample_rate = s->samplerate;

     avctx->bits_per_raw_sample = s->bps;

+    ff_flac_set_channel_layout(avctx);

     s->samples = get_bits_longlong(&gb, 36);

@@ -137,4 +137,7 @@ int ff_flac_get_max_frame_size(int blocksize, int ch, int bps);

  */

 int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,

                                 FLACFrameInfo *fi, int log_level_offset);

+

+void ff_flac_set_channel_layout(AVCodecContext *avctx);

+

 #endif /* AVCODEC_FLAC_H */

@@ -459,6 +459,7 @@ static int get_best_header(FLACParseContext* fpc, const uint8_t **poutbuf,

     fpc->avctx->sample_rate = header->fi.samplerate;

     fpc->avctx->channels    = header->fi.channels;

+    ff_flac_set_channel_layout(fpc->avctx);

     fpc->pc->duration       = header->fi.blocksize;

     *poutbuf = flac_fifo_read_wrap(fpc, header->offset, *poutbuf_size,

                                         &fpc->wrap_buf,

@@ -58,20 +58,13 @@ typedef struct FLACContext {

     int got_streaminfo;                     ///< indicates if the STREAMINFO has been read

     int32_t *decoded[FLAC_MAX_CHANNELS];    ///< decoded samples

+    uint8_t *decoded_buffer;

+    unsigned int decoded_buffer_size;

     FLACDSPContext dsp;

 } FLACContext;

-static const int64_t flac_channel_layouts[6] = {

-    AV_CH_LAYOUT_MONO,

-    AV_CH_LAYOUT_STEREO,

-    AV_CH_LAYOUT_SURROUND,

-    AV_CH_LAYOUT_QUAD,

-    AV_CH_LAYOUT_5POINT0,

-    AV_CH_LAYOUT_5POINT1

-};

-

-static void allocate_buffers(FLACContext *s);

+static int allocate_buffers(FLACContext *s);

 static void flac_set_bps(FLACContext *s)

 {

@@ -99,6 +92,7 @@ static av_cold int flac_decode_init(AVCodecContext *avctx)

 {

     enum FLACExtradataFormat format;

     uint8_t *streaminfo;

+    int ret;

     FLACContext *s = avctx->priv_data;

     s->avctx = avctx;

@@ -112,7 +106,9 @@ static av_cold int flac_decode_init(AVCodecContext *avctx)

     /* initialize based on the demuxer-supplied streamdata header */

     avpriv_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);

-    allocate_buffers(s);

+    ret = allocate_buffers(s);

+    if (ret < 0)

+        return ret;

     flac_set_bps(s);

     ff_flacdsp_init(&s->dsp, avctx->sample_fmt, s->bps);

     s->got_streaminfo = 1;

@@ -120,9 +116,6 @@ static av_cold int flac_decode_init(AVCodecContext *avctx)

     avcodec_get_frame_defaults(&s->frame);

     avctx->coded_frame = &s->frame;

-    if (avctx->channels <= FF_ARRAY_ELEMS(flac_channel_layouts))

-        avctx->channel_layout = flac_channel_layouts[avctx->channels - 1];

-

     return 0;

 }

@@ -135,15 +128,24 @@ static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s)

     av_log(avctx, AV_LOG_DEBUG, "  Bits: %d\n", s->bps);

 }

-static void allocate_buffers(FLACContext *s)

+static int allocate_buffers(FLACContext *s)

 {

-    int i;

+    int buf_size;

     av_assert0(s->max_blocksize);

-    for (i = 0; i < s->channels; i++) {

-        s->decoded[i] = av_malloc(sizeof(int32_t)*s->max_blocksize);

-    }

+    buf_size = av_samples_get_buffer_size(NULL, s->channels, s->max_blocksize,

+                                          AV_SAMPLE_FMT_S32P, 0);

+    if (buf_size < 0)

+        return buf_size;

+

+    av_fast_malloc(&s->decoded_buffer, &s->decoded_buffer_size, buf_size);

+    if (!s->decoded_buffer)

+        return AVERROR(ENOMEM);

+

+    return av_samples_fill_arrays((uint8_t **)s->decoded, NULL,

+                                  s->decoded_buffer, s->channels,

+                                  s->max_blocksize, AV_SAMPLE_FMT_S32P, 0);

 }

 /**

@@ -155,7 +157,7 @@ static void allocate_buffers(FLACContext *s)

  */

 static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size)

 {

-    int metadata_type, metadata_size;

+    int metadata_type, metadata_size, ret;

     if (buf_size < FLAC_STREAMINFO_SIZE+8) {

         /* need more data */

@@ -167,7 +169,9 @@ static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size)

         return AVERROR_INVALIDDATA;

     }

     avpriv_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]);

-    allocate_buffers(s);

+    ret = allocate_buffers(s);

+    if (ret < 0)

+        return ret;

     flac_set_bps(s);

     ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);

     s->got_streaminfo = 1;

@@ -403,7 +407,7 @@ static inline int decode_subframe(FLACContext *s, int channel)

 static int decode_frame(FLACContext *s)

 {

-    int i;

+    int i, ret;

     GetBitContext *gb = &s->gb;

     FLACFrameInfo fi;

@@ -412,12 +416,15 @@ static int decode_frame(FLACContext *s)

         return -1;

     }

-    if (s->channels && fi.channels != s->channels) {

-        av_log(s->avctx, AV_LOG_ERROR, "switching channel layout mid-stream "

-                                       "is not supported\n");

-        return -1;

+    if (s->channels && fi.channels != s->channels && s->got_streaminfo) {

+        s->channels = s->avctx->channels = fi.channels;

+        ff_flac_set_channel_layout(s->avctx);

+        ret = allocate_buffers(s);

+        if (ret < 0)

+            return ret;

     }

     s->channels = s->avctx->channels = fi.channels;

+    ff_flac_set_channel_layout(s->avctx);

     s->ch_mode = fi.ch_mode;

     if (!s->bps && !fi.bps) {

@@ -451,16 +458,14 @@ static int decode_frame(FLACContext *s)

                                         " or frame header\n");

         return -1;

     }

-    if (fi.samplerate == 0) {

+    if (fi.samplerate == 0)

         fi.samplerate = s->samplerate;

-    } else if (s->samplerate && fi.samplerate != s->samplerate) {

-        av_log(s->avctx, AV_LOG_WARNING, "sample rate changed from %d to %d\n",

-               s->samplerate, fi.samplerate);

-    }

     s->samplerate = s->avctx->sample_rate = fi.samplerate;

     if (!s->got_streaminfo) {

-        allocate_buffers(s);

+        ret = allocate_buffers(s);

+        if (ret < 0)

+            return ret;

         ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);

         s->got_streaminfo = 1;

         dump_headers(s->avctx, (FLACStreaminfo *)s);

@@ -550,11 +555,8 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data,

 static av_cold int flac_decode_close(AVCodecContext *avctx)

 {

     FLACContext *s = avctx->priv_data;

-    int i;

-    for (i = 0; i < s->channels; i++) {

-        av_freep(&s->decoded[i]);

-    }

+    av_freep(&s->decoded_buffer);

     return 0;

 }

@@ -22,6 +22,8 @@

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

  */

 #include <limits.h>

+

+#include "libavutil/audioconvert.h"

 #include "libavutil/avassert.h"

 #include "libavutil/opt.h"

 #include "avcodec.h"

@@ -418,18 +420,8 @@ static av_cold int g726_decode_init(AVCodecContext *avctx)

 {

     G726Context* c = avctx->priv_data;

-    if (avctx->strict_std_compliance >= FF_COMPLIANCE_STRICT &&

-        avctx->sample_rate != 8000) {

-        av_log(avctx, AV_LOG_ERROR, "Only 8kHz sample rate is allowed when "

-               "the compliance level is strict. Reduce the compliance level "

-               "if you wish to decode the stream anyway.\n");

-        return AVERROR(EINVAL);

-    }

-

-    if(avctx->channels != 1){

-        av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");

-        return AVERROR(EINVAL);

-    }

+    avctx->channels       = 1;

+    avctx->channel_layout = AV_CH_LAYOUT_MONO;

     c->code_size = avctx->bits_per_coded_sample;

     if (c->code_size < 2 || c->code_size > 5) {

@@ -24,6 +24,7 @@

  * GSM decoder

  */

+#include "libavutil/audioconvert.h"

 #include "avcodec.h"

 #include "get_bits.h"

 #include "msgsmdec.h"

@@ -34,10 +35,10 @@ static av_cold int gsm_init(AVCodecContext *avctx)

 {

     GSMContext *s = avctx->priv_data;

-    avctx->channels = 1;

-    if (!avctx->sample_rate)

-        avctx->sample_rate = 8000;

-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;

+    avctx->channels       = 1;

+    avctx->channel_layout = AV_CH_LAYOUT_MONO;

+    avctx->sample_rate    = 8000;

+    avctx->sample_fmt     = AV_SAMPLE_FMT_S16;

     switch (avctx->codec_id) {

     case AV_CODEC_ID_GSM:

@@ -176,8 +176,10 @@ static av_cold int imc_decode_init(AVCodecContext *avctx)

     IMCContext *q = avctx->priv_data;

     double r1, r2;

-    if ((avctx->codec_id == AV_CODEC_ID_IMC && avctx->channels != 1)

-        || (avctx->codec_id == AV_CODEC_ID_IAC && avctx->channels > 2)) {

+    if (avctx->codec_id == AV_CODEC_ID_IMC)

+        avctx->channels = 1;

+

+    if (avctx->channels > 2) {

         av_log_ask_for_sample(avctx, "Number of channels is not supported\n");

         return AVERROR_PATCHWELCOME;

     }

@@ -29,6 +29,7 @@

 #include <gsm/gsm.h>

+#include "libavutil/audioconvert.h"

 #include "avcodec.h"

 #include "internal.h"

 #include "gsm.h"

@@ -153,19 +154,10 @@ typedef struct LibGSMDecodeContext {

 static av_cold int libgsm_decode_init(AVCodecContext *avctx) {

     LibGSMDecodeContext *s = avctx->priv_data;

-    if (avctx->channels > 1) {

-        av_log(avctx, AV_LOG_ERROR, "Mono required for GSM, got %d channels\n",

-               avctx->channels);

-        return -1;

-    }

-

-    if (!avctx->channels)

-        avctx->channels = 1;

-

-    if (!avctx->sample_rate)

-        avctx->sample_rate = 8000;

-

-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;

+    avctx->channels       = 1;

+    avctx->channel_layout = AV_CH_LAYOUT_MONO;

+    avctx->sample_rate    = 8000;

+    avctx->sample_fmt     = AV_SAMPLE_FMT_S16;

     s->state = gsm_create();

@@ -21,6 +21,7 @@

 #include <ilbc.h>

+#include "libavutil/audioconvert.h"

 #include "avcodec.h"

 #include "libavutil/common.h"

 #include "libavutil/opt.h"

@@ -71,9 +72,10 @@ static av_cold int ilbc_decode_init(AVCodecContext *avctx)

     avcodec_get_frame_defaults(&s->frame);

     avctx->coded_frame = &s->frame;

-    avctx->channels = 1;

-    avctx->sample_rate = 8000;

-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;

+    avctx->channels       = 1;

+    avctx->channel_layout = AV_CH_LAYOUT_MONO;

+    avctx->sample_rate    = 8000;

+    avctx->sample_fmt     = AV_SAMPLE_FMT_S16;

     return 0;

 }

@@ -19,6 +19,7 @@

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

  */

+#include "libavutil/audioconvert.h"

 #include "avcodec.h"

 #include "libavutil/avstring.h"

 #include "libavutil/common.h"

@@ -30,13 +31,16 @@ static void amr_decode_fix_avctx(AVCodecContext *avctx)

 {

     const int is_amr_wb = 1 + (avctx->codec_id == AV_CODEC_ID_AMR_WB);

-    if (!avctx->sample_rate)

-        avctx->sample_rate = 8000 * is_amr_wb;

+    avctx->sample_rate = 8000 * is_amr_wb;

-    if (!avctx->channels)

-        avctx->channels = 1;

+    if (avctx->channels > 1) {

+        av_log_missing_feature(avctx, "multi-channel AMR", 0);

+        return AVERROR_PATCHWELCOME;

+    }

-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;

+    avctx->channels       = 1;

+    avctx->channel_layout = AV_CH_LAYOUT_MONO;

+    avctx->sample_fmt     = AV_SAMPLE_FMT_S16;

 }

 #if CONFIG_LIBOPENCORE_AMRNB

@@ -129,6 +129,7 @@ static av_cold int decode_init(AVCodecContext * avctx) {

     if (!ff_sine_128[127])

         ff_init_ff_sine_windows(7);

+    avctx->channels       = 1;

     avctx->channel_layout = AV_CH_LAYOUT_MONO;

     avcodec_get_frame_defaults(&s->frame);

@@ -29,6 +29,7 @@

 #include <stddef.h>

+#include "libavutil/audioconvert.h"

 #include "avcodec.h"

 #include "internal.h"

 #include "get_bits.h"

@@ -89,7 +90,9 @@ static av_cold int qcelp_decode_init(AVCodecContext *avctx)

     QCELPContext *q = avctx->priv_data;

     int i;

-    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

+    avctx->channels       = 1;

+    avctx->channel_layout = AV_CH_LAYOUT_MONO;

+    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;

     for (i = 0; i < 10; i++)

         q->prev_lspf[i] = (i + 1) / 11.;

@@ -36,6 +36,7 @@

 #include <stdio.h>

 #define BITSTREAM_READER_LE

+#include "libavutil/audioconvert.h"

 #include "avcodec.h"

 #include "get_bits.h"

 #include "dsputil.h"

@@ -550,10 +551,6 @@ static void fill_tone_level_array (QDM2Context *q, int flag)

     int i, sb, ch, sb_used;

     int tmp, tab;

-    // This should never happen

-    if (q->nb_channels <= 0)

-        return;

-

     for (ch = 0; ch < q->nb_channels; ch++)

         for (sb = 0; sb < 30; sb++)

             for (i = 0; i < 8; i++) {

@@ -649,10 +646,6 @@ static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_arra

     int add1, add2, add3, add4;

     int64_t multres;

-    // This should never happen

-    if (nb_channels <= 0)

-        return;

-

     if (!superblocktype_2_3) {

         /* This case is untested, no samples available */

         SAMPLES_NEEDED

@@ -1792,10 +1785,12 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)

     avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);

     extradata += 4;

-    if (s->channels > MPA_MAX_CHANNELS) {

-        av_log(avctx, AV_LOG_ERROR, "Too many channels\n");

+    if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {

+        av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");

         return AVERROR_INVALIDDATA;

     }

+    avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :

+                                                   AV_CH_LAYOUT_MONO;

     avctx->sample_rate = AV_RB32(extradata);

     extradata += 4;

@@ -22,6 +22,7 @@

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

  */

+#include "libavutil/audioconvert.h"

 #include "libavutil/intmath.h"

 #include "avcodec.h"

 #include "get_bits.h"

@@ -37,7 +38,9 @@ static av_cold int ra144_decode_init(AVCodecContext * avctx)

     ractx->lpc_coef[0] = ractx->lpc_tables[0];

     ractx->lpc_coef[1] = ractx->lpc_tables[1];

-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;

+    avctx->channels       = 1;

+    avctx->channel_layout = AV_CH_LAYOUT_MONO;

+    avctx->sample_fmt     = AV_SAMPLE_FMT_S16;

     avcodec_get_frame_defaults(&ractx->frame);

     avctx->coded_frame = &ractx->frame;

@@ -19,6 +19,7 @@

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

  */

+#include "libavutil/audioconvert.h"

 #include "libavutil/float_dsp.h"

 #include "avcodec.h"

 #define BITSTREAM_READER_LE

@@ -61,7 +62,11 @@ typedef struct {

 static av_cold int ra288_decode_init(AVCodecContext *avctx)

 {

     RA288Context *ractx = avctx->priv_data;

-    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

+

+    avctx->channels       = 1;

+    avctx->channel_layout = AV_CH_LAYOUT_MONO;

+    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;

+

     avpriv_float_dsp_init(&ractx->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

     avcodec_get_frame_defaults(&ractx->frame);

@@ -340,7 +340,7 @@ static int read_header(ShortenContext *s)

     s->internal_ftype = get_uint(s, TYPESIZE);

     s->channels = get_uint(s, CHANSIZE);

-    if (s->channels > MAX_CHANNELS) {

+    if (s->channels <= 0 || s->channels > MAX_CHANNELS) {

         av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels);

         return -1;

     }

@@ -25,6 +25,7 @@

 #include <stdint.h>

 #include <string.h>

+#include "libavutil/audioconvert.h"

 #include "libavutil/mathematics.h"

 #include "avcodec.h"

 #define BITSTREAM_READER_LE

@@ -509,7 +510,9 @@ static av_cold int sipr_decoder_init(AVCodecContext * avctx)

     for (i = 0; i < 4; i++)

         ctx->energy_history[i] = -14;

-    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

+    avctx->channels       = 1;

+    avctx->channel_layout = AV_CH_LAYOUT_MONO;

+    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;

     avcodec_get_frame_defaults(&ctx->frame);

     avctx->coded_frame = &ctx->frame;

@@ -19,6 +19,7 @@

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

  */

+#include "libavutil/audioconvert.h"

 #include "libavutil/intreadwrite.h"

 #include "avcodec.h"

 #include "dsputil.h"

@@ -66,7 +67,8 @@ static av_cold int truespeech_decode_init(AVCodecContext * avctx)

         return AVERROR(EINVAL);

     }

-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;

+    avctx->channel_layout = AV_CH_LAYOUT_MONO;

+    avctx->sample_fmt     = AV_SAMPLE_FMT_S16;

     ff_dsputil_init(&c->dsp, avctx);

@@ -19,6 +19,7 @@

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

  */

+#include "libavutil/audioconvert.h"

 #include "libavutil/float_dsp.h"

 #include "avcodec.h"

 #include "get_bits.h"

@@ -1119,6 +1120,11 @@ static av_cold int twin_decode_init(AVCodecContext *avctx)

     avctx->channels = AV_RB32(avctx->extradata    ) + 1;

     avctx->bit_rate = AV_RB32(avctx->extradata + 4) * 1000;

     isampf          = AV_RB32(avctx->extradata + 8);

+

+    if (isampf < 8 || isampf > 44) {

+        av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate\n");

+        return AVERROR_INVALIDDATA;

+    }

     switch (isampf) {

     case 44: avctx->sample_rate = 44100;         break;

     case 22: avctx->sample_rate = 22050;         break;

@@ -1126,11 +1132,14 @@ static av_cold int twin_decode_init(AVCodecContext *avctx)

     default: avctx->sample_rate = isampf * 1000; break;

     }

-    if (avctx->channels > CHANNELS_MAX) {

+    if (avctx->channels <= 0 || avctx->channels > CHANNELS_MAX) {

         av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %i\n",

                avctx->channels);

         return -1;

     }

+    avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :

+                                                   AV_CH_LAYOUT_STEREO;

+

     ibps = avctx->bit_rate / (1000 * avctx->channels);

     switch ((isampf << 8) +  ibps) {

@@ -43,6 +43,7 @@

 #include <stdlib.h>

 #include <string.h>

+#include "libavutil/audioconvert.h"

 #include "libavutil/common.h"

 #include "libavutil/intreadwrite.h"

 #include "avcodec.h"

@@ -501,6 +502,9 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)

         return AVERROR(EINVAL);

     }

+    avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :

+                                                   AV_CH_LAYOUT_STEREO;

+

     if (avctx->bits_per_coded_sample == 16)

         avctx->sample_fmt = AV_SAMPLE_FMT_S16;

     else