Merge commit '50a65e7a540ce6747f81d6dbf6a602ad35be77ff'

* commit '50a65e7a540ce6747f81d6dbf6a602ad35be77ff': (24 commits)
  vmdaudio: set channel layout
  twinvq: validate sample rate code
  twinvq: set channel layout
  twinvq: validate that channels is not <= 0
  truespeech: set channel layout
  sipr: set channel layout
  shorten: validate that the channel count in the header is not <= 0
  ra288dec: set channel layout
  ra144dec: set channel layout
  qdm2: remove unneeded checks for channel count
  qdm2: make sure channels is not <= 0 and set channel layout
  qcelpdec: set channel layout
  nellymoserdec: set channels to 1
  libopencore-amr: set channel layout for amr-nb or if not set by the user
  libilbc: set channel layout
  dpcm: use AVCodecContext.channels instead of keeping a private copy
  imc: set channels to 1 instead of validating it
  gsmdec: always set channel layout and sample rate at initialization
  libgsmdec: always set channel layout and sample rate at initialization
  g726dec: do not validate sample rate
  ...

Conflicts:
	libavcodec/dpcm.c
	libavcodec/qdm2.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

libavcodec/dpcm.c

@@ -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;

libavcodec/flac.c

@@ -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);
 

libavcodec/flac.h

@@ -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 */

libavcodec/flac_parser.c

@@ -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,

libavcodec/flacdec.c

@@ -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] = {
+static int allocate_buffers(FLACContext *s);
-    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 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++) {
+    buf_size = av_samples_get_buffer_size(NULL, s->channels, s->max_blocksize,
-        s->decoded[i] = av_malloc(sizeof(int32_t)*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) {
+    if (s->channels && fi.channels != s->channels && s->got_streaminfo) {
-        av_log(s->avctx, AV_LOG_ERROR, "switching channel layout mid-stream "
+        s->channels = s->avctx->channels = fi.channels;
-                                       "is not supported\n");
+        ff_flac_set_channel_layout(s->avctx);
-        return -1;
+        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_buffer);
-        av_freep(&s->decoded[i]);
-    }
 
     return 0;
 }

libavcodec/g726.c

@@ -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->channels       = 1;
-        avctx->sample_rate != 8000) {
+    avctx->channel_layout = AV_CH_LAYOUT_MONO;
-        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);
-    }
 
     c->code_size = avctx->bits_per_coded_sample;
     if (c->code_size < 2 || c->code_size > 5) {

libavcodec/gsmdec.c

@@ -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;
+    avctx->channels       = 1;
-    if (!avctx->sample_rate)
+    avctx->channel_layout = AV_CH_LAYOUT_MONO;
-        avctx->sample_rate = 8000;
+    avctx->sample_rate    = 8000;
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+    avctx->sample_fmt     = AV_SAMPLE_FMT_S16;
 
     switch (avctx->codec_id) {
     case AV_CODEC_ID_GSM:

libavcodec/imc.c

@@ -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)
+    if (avctx->codec_id == AV_CODEC_ID_IMC)
-        || (avctx->codec_id == AV_CODEC_ID_IAC && avctx->channels > 2)) {
+        avctx->channels = 1;
+
+    if (avctx->channels > 2) {
         av_log_ask_for_sample(avctx, "Number of channels is not supported\n");
         return AVERROR_PATCHWELCOME;
     }

libavcodec/libgsm.c

@@ -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) {
+    avctx->channels       = 1;
-        av_log(avctx, AV_LOG_ERROR, "Mono required for GSM, got %d channels\n",
+    avctx->channel_layout = AV_CH_LAYOUT_MONO;
-               avctx->channels);
+    avctx->sample_rate    = 8000;
-        return -1;
+    avctx->sample_fmt     = AV_SAMPLE_FMT_S16;
-    }
-
-    if (!avctx->channels)
-        avctx->channels = 1;
-
-    if (!avctx->sample_rate)
-        avctx->sample_rate = 8000;
-
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
 
     s->state = gsm_create();
 

libavcodec/libilbc.c

@@ -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->channels       = 1;
-    avctx->sample_rate = 8000;
+    avctx->channel_layout = AV_CH_LAYOUT_MONO;
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+    avctx->sample_rate    = 8000;
+    avctx->sample_fmt     = AV_SAMPLE_FMT_S16;
 
     return 0;
 }

libavcodec/libopencore-amr.c

@@ -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)
+    if (avctx->channels > 1) {
-        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

libavcodec/nellymoserdec.c

@@ -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);

libavcodec/qcelpdec.c

@@ -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.;

libavcodec/qdm2.c

@@ -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) {
+    if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
-        av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
+        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;

libavcodec/ra144dec.c

@@ -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;

libavcodec/ra288.c

@@ -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);

libavcodec/shorten.c

@@ -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;
     }

libavcodec/sipr.c

@@ -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;

libavcodec/truespeech.c

@@ -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);
 

libavcodec/twinvq.c

@@ -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) {

libavcodec/vmdav.c

@@ -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