Merge remote-tracking branch 'qatar/master'

* qatar/master: (23 commits)
  aacenc: Fix identification padding when the bitstream is already aligned.
  aacenc: Write correct length for long identification strings.
  aud: remove unneeded field, audio_stream_index from context
  aud: fix time stamp calculation for ADPCM IMA WS
  aud: simplify header parsing
  aud: set pts_wrap_bits to 64.
  cosmetics: indentation
  aud: support Westwood SND1 audio in AUD files.
  adpcm_ima_ws: fix stereo decoding
  avcodec: add a new codec_id for CRYO APC IMA ADPCM.
  vqa: remove unused context fields, audio_samplerate and audio_bits
  vqa: clean up audio header parsing
  vqa: set time base to frame rate as coded in the header.
  vqa: set packet duration.
  vqa: use 1/sample_rate as the audio stream time base
  vqa: set stream start_time to 0.
  lavc: postpone the removal of AVCodecContext.request_channels.
  lavf: postpone removing av_close_input_file().
  lavc: postpone removing old audio encoding and decoding API
  avplay: remove the -er option.
  ...

Conflicts:
	Changelog
	libavcodec/version.h
	libavdevice/v4l.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Changelog

 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-
 version next:
 - v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder
 - SBaGen (SBG) binaural beats script demuxer

ffplay.c

@@ -265,7 +265,6 @@ static int idct = FF_IDCT_AUTO;
 static enum AVDiscard skip_frame       = AVDISCARD_DEFAULT;
 static enum AVDiscard skip_idct        = AVDISCARD_DEFAULT;
 static enum AVDiscard skip_loop_filter = AVDISCARD_DEFAULT;
-static int error_recognition = FF_ER_CAREFUL;
 static int error_concealment = 3;
 static int decoder_reorder_pts = -1;
 static int autoexit;
@@ -2236,7 +2235,6 @@ static int stream_component_open(VideoState *is, int stream_index)
     avctx->skip_frame        = skip_frame;
     avctx->skip_idct         = skip_idct;
     avctx->skip_loop_filter  = skip_loop_filter;
-    avctx->error_recognition = error_recognition;
     avctx->error_concealment = error_concealment;
 
     if(avctx->lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
@@ -3071,7 +3069,6 @@ static const OptionDef options[] = {
     { "skipframe", OPT_INT | HAS_ARG | OPT_EXPERT, { (void*)&skip_frame }, "", "" },
     { "skipidct", OPT_INT | HAS_ARG | OPT_EXPERT, { (void*)&skip_idct }, "", "" },
     { "idct", OPT_INT | HAS_ARG | OPT_EXPERT, { (void*)&idct }, "set idct algo",  "algo" },
-    { "er", OPT_INT | HAS_ARG | OPT_EXPERT, { (void*)&error_recognition }, "set error detection threshold (0-4)",  "threshold" },
     { "ec", OPT_INT | HAS_ARG | OPT_EXPERT, { (void*)&error_concealment }, "set error concealment options",  "bit_mask" },
     { "sync", HAS_ARG | OPT_EXPERT, { (void*)opt_sync }, "set audio-video sync. type (type=audio/video/ext)", "type" },
     { "autoexit", OPT_BOOL | OPT_EXPERT, { (void*)&autoexit }, "exit at the end", "" },

libavcodec/Makefile

@@ -548,6 +548,7 @@ OBJS-$(CONFIG_ADPCM_G722_ENCODER)         += g722.o g722enc.o
 OBJS-$(CONFIG_ADPCM_G726_DECODER)         += g726.o
 OBJS-$(CONFIG_ADPCM_G726_ENCODER)         += g726.o
 OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER)      += adpcm.o adpcm_data.o
+OBJS-$(CONFIG_ADPCM_IMA_APC_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_DK4_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_EA_EACS_DECODER)  += adpcm.o adpcm_data.o

libavcodec/aacenc.c

@@ -464,7 +464,7 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
     if (namelen >= 15)
         put_bits(&s->pb, 8, namelen - 14);
     put_bits(&s->pb, 4, 0); //extension type - filler
-    padbits = 8 - (put_bits_count(&s->pb) & 7);
+    padbits = -put_bits_count(&s->pb) & 7;
     avpriv_align_put_bits(&s->pb);
     for (i = 0; i < namelen - 2; i++)
         put_bits(&s->pb, 8, name[i]);

libavcodec/adpcm.c

@@ -86,6 +86,7 @@ static const int swf_index_tables[4][16] = {
 typedef struct ADPCMDecodeContext {
     AVFrame frame;
     ADPCMChannelStatus status[6];
+    int vqa_version;                /**< VQA version. Used for ADPCM_IMA_WS */
 } ADPCMDecodeContext;
 
 static av_cold int adpcm_decode_init(AVCodecContext * avctx)
@@ -120,12 +121,16 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
             return -1;
         }
         break;
-    case CODEC_ID_ADPCM_IMA_WS:
-        if (avctx->extradata && avctx->extradata_size == 2 * 4) {
+    case CODEC_ID_ADPCM_IMA_APC:
+        if (avctx->extradata && avctx->extradata_size >= 8) {
             c->status[0].predictor = AV_RL32(avctx->extradata);
             c->status[1].predictor = AV_RL32(avctx->extradata + 4);
         }
         break;
+    case CODEC_ID_ADPCM_IMA_WS:
+        if (avctx->extradata && avctx->extradata_size >= 42)
+            c->vqa_version = AV_RL16(avctx->extradata);
+        break;
     default:
         break;
     }
@@ -362,6 +367,7 @@ static int get_nb_samples(AVCodecContext *avctx, const uint8_t *buf,
         break;
     /* simple 4-bit adpcm */
     case CODEC_ID_ADPCM_CT:
+    case CODEC_ID_ADPCM_IMA_APC:
     case CODEC_ID_ADPCM_IMA_EA_SEAD:
     case CODEC_ID_ADPCM_IMA_WS:
     case CODEC_ID_ADPCM_YAMAHA:
@@ -776,13 +782,37 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
             *samples++ = adpcm_ima_expand_nibble(&c->status[st], v2, 3);
         }
         break;
-    case CODEC_ID_ADPCM_IMA_WS:
+    case CODEC_ID_ADPCM_IMA_APC:
         while (src < buf + buf_size) {
             uint8_t v = *src++;
             *samples++ = adpcm_ima_expand_nibble(&c->status[0],  v >> 4  , 3);
             *samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3);
         }
         break;
+    case CODEC_ID_ADPCM_IMA_WS:
+        for (channel = 0; channel < avctx->channels; channel++) {
+            const uint8_t *src0;
+            int src_stride;
+            int16_t *smp = samples + channel;
+
+            if (c->vqa_version == 3) {
+                src0 = src + channel * buf_size / 2;
+                src_stride = 1;
+            } else {
+                src0 = src + channel;
+                src_stride = avctx->channels;
+            }
+            for (n = nb_samples / 2; n > 0; n--) {
+                uint8_t v = *src0;
+                src0 += src_stride;
+                *smp = adpcm_ima_expand_nibble(&c->status[channel], v >> 4  , 3);
+                smp += avctx->channels;
+                *smp = adpcm_ima_expand_nibble(&c->status[channel], v & 0x0F, 3);
+                smp += avctx->channels;
+            }
+        }
+        src = buf + buf_size;
+        break;
     case CODEC_ID_ADPCM_XA:
         while (buf_size >= 128) {
             xa_decode(samples, src, &c->status[0], &c->status[1],
@@ -1226,6 +1256,7 @@ ADPCM_DECODER(CODEC_ID_ADPCM_EA_R2, adpcm_ea_r2, "ADPCM Electronic Arts R2");
 ADPCM_DECODER(CODEC_ID_ADPCM_EA_R3, adpcm_ea_r3, "ADPCM Electronic Arts R3");
 ADPCM_DECODER(CODEC_ID_ADPCM_EA_XAS, adpcm_ea_xas, "ADPCM Electronic Arts XAS");
 ADPCM_DECODER(CODEC_ID_ADPCM_IMA_AMV, adpcm_ima_amv, "ADPCM IMA AMV");
+ADPCM_DECODER(CODEC_ID_ADPCM_IMA_APC, adpcm_ima_apc, "ADPCM IMA CRYO APC");
 ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK3, adpcm_ima_dk3, "ADPCM IMA Duck DK3");
 ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK4, adpcm_ima_dk4, "ADPCM IMA Duck DK4");
 ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs, "ADPCM IMA Electronic Arts EACS");

libavcodec/allcodecs.c

@@ -365,6 +365,7 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC  (ADPCM_G722, adpcm_g722);
     REGISTER_ENCDEC  (ADPCM_G726, adpcm_g726);
     REGISTER_DECODER (ADPCM_IMA_AMV, adpcm_ima_amv);
+    REGISTER_DECODER (ADPCM_IMA_APC, adpcm_ima_apc);
     REGISTER_DECODER (ADPCM_IMA_DK3, adpcm_ima_dk3);
     REGISTER_DECODER (ADPCM_IMA_DK4, adpcm_ima_dk4);
     REGISTER_DECODER (ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs);

libavcodec/avcodec.h

@@ -326,6 +326,7 @@ enum CodecID {
     CODEC_ID_ADPCM_EA_MAXIS_XA,
     CODEC_ID_ADPCM_IMA_ISS,
     CODEC_ID_ADPCM_G722,
+    CODEC_ID_ADPCM_IMA_APC,
 
     /* AMR */
     CODEC_ID_AMR_NB = 0x12000,

libavcodec/rv10.c

@@ -666,9 +666,12 @@ static int rv10_decode_frame(AVCodecContext *avctx,
         slice_count = avctx->slice_count;
 
     for(i=0; i<slice_count; i++){
-        int offset= get_slice_offset(avctx, slices_hdr, i);
+        unsigned offset = get_slice_offset(avctx, slices_hdr, i);
         int size, size2;
 
+        if (offset >= buf_size)
+            return AVERROR_INVALIDDATA;
+
         if(i+1 == slice_count)
             size= buf_size - offset;
         else
@@ -679,6 +682,10 @@ static int rv10_decode_frame(AVCodecContext *avctx,
         else
             size2= get_slice_offset(avctx, slices_hdr, i+2) - offset;
 
+        if (size <= 0 || size2 <= 0 ||
+            offset + FFMAX(size, size2) > buf_size)
+            return AVERROR_INVALIDDATA;
+
         if(rv10_decode_packet(avctx, buf+offset, size, size2) > 8*size)
             i++;
     }

libavcodec/utils.c

@@ -1780,6 +1780,7 @@ int av_get_bits_per_sample(enum CodecID codec_id){
         return 3;
     case CODEC_ID_ADPCM_SBPRO_4:
     case CODEC_ID_ADPCM_CT:
+    case CODEC_ID_ADPCM_IMA_APC:
     case CODEC_ID_ADPCM_IMA_WAV:
     case CODEC_ID_ADPCM_IMA_QT:
     case CODEC_ID_ADPCM_SWF:

libavcodec/version.h

@@ -21,7 +21,7 @@
 #define AVCODEC_VERSION_H
 
 #define LIBAVCODEC_VERSION_MAJOR 53
-#define LIBAVCODEC_VERSION_MINOR 58
+#define LIBAVCODEC_VERSION_MINOR 59
 #define LIBAVCODEC_VERSION_MICRO 100
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
@@ -51,7 +51,7 @@
 #define FF_API_ANTIALIAS_ALGO   (LIBAVCODEC_VERSION_MAJOR < 54)
 #endif
 #ifndef FF_API_REQUEST_CHANNELS
-#define FF_API_REQUEST_CHANNELS (LIBAVCODEC_VERSION_MAJOR < 54)
+#define FF_API_REQUEST_CHANNELS (LIBAVCODEC_VERSION_MAJOR < 55)
 #endif
 #ifndef FF_API_OPT_H
 #define FF_API_OPT_H            (LIBAVCODEC_VERSION_MAJOR < 54)
@@ -114,7 +114,7 @@
 #define FF_API_DATA_POINTERS (LIBAVCODEC_VERSION_MAJOR < 54)
 #endif
 #ifndef FF_API_OLD_DECODE_AUDIO
-#define FF_API_OLD_DECODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 54)
+#define FF_API_OLD_DECODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 55)
 #endif
 #ifndef FF_API_OLD_TIMECODE
 #define FF_API_OLD_TIMECODE (LIBAVCODEC_VERSION_MAJOR < 54)
@@ -124,7 +124,7 @@
 #define FF_API_AVFRAME_AGE (LIBAVCODEC_VERSION_MAJOR < 54)
 #endif
 #ifndef FF_API_OLD_ENCODE_AUDIO
-#define FF_API_OLD_ENCODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 54)
+#define FF_API_OLD_ENCODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 55)
 #endif
 
 #endif /* AVCODEC_VERSION_H */

libavdevice/Makefile

@@ -29,7 +29,6 @@ OBJS-$(CONFIG_SDL_OUTDEV)                += sdl.o
 OBJS-$(CONFIG_SNDIO_INDEV)               += sndio_common.o sndio_dec.o
 OBJS-$(CONFIG_SNDIO_OUTDEV)              += sndio_common.o sndio_enc.o
 OBJS-$(CONFIG_V4L2_INDEV)                += v4l2.o
-OBJS-$(CONFIG_V4L_INDEV)                 += v4l.o
 OBJS-$(CONFIG_VFWCAP_INDEV)              += vfwcap.o
 OBJS-$(CONFIG_X11_GRAB_DEVICE_INDEV)     += x11grab.o
 

libavdevice/alldevices.c

@@ -51,9 +51,7 @@ void avdevice_register_all(void)
     REGISTER_OUTDEV   (SDL, sdl);
     REGISTER_INOUTDEV (SNDIO, sndio);
     REGISTER_INDEV    (V4L2, v4l2);
-#if FF_API_V4L
     REGISTER_INDEV    (V4L, v4l);
-#endif
     REGISTER_INDEV    (VFWCAP, vfwcap);
     REGISTER_INDEV    (X11_GRAB_DEVICE, x11_grab_device);
 

libavdevice/avdevice.h

@@ -56,10 +56,6 @@
                                            LIBAVDEVICE_VERSION_MICRO)
 #define LIBAVDEVICE_BUILD       LIBAVDEVICE_VERSION_INT
 
-#ifndef FF_API_V4L
-#define FF_API_V4L              (LIBAVDEVICE_VERSION_MAJOR < 54)
-#endif
-
 /**
  * Return the LIBAVDEVICE_VERSION_INT constant.
  */

libavformat/apc.c

@@ -44,7 +44,7 @@ static int apc_read_header(AVFormatContext *s, AVFormatParameters *ap)
         return AVERROR(ENOMEM);
 
     st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-    st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS;
+    st->codec->codec_id = CODEC_ID_ADPCM_IMA_APC;
 
     avio_rl32(pb); /* number of samples */
     st->codec->sample_rate = avio_rl32(pb);

libavformat/version.h

@@ -123,7 +123,7 @@
 #define FF_API_SET_PTS_INFO            (LIBAVFORMAT_VERSION_MAJOR < 54)
 #endif
 #ifndef FF_API_CLOSE_INPUT_FILE
-#define FF_API_CLOSE_INPUT_FILE        (LIBAVFORMAT_VERSION_MAJOR < 54)
+#define FF_API_CLOSE_INPUT_FILE        (LIBAVFORMAT_VERSION_MAJOR < 55)
 #endif
 
 #endif /* AVFORMAT_VERSION_H */

libavformat/westwood_aud.c

@@ -41,15 +41,6 @@
 #define AUD_CHUNK_PREAMBLE_SIZE 8
 #define AUD_CHUNK_SIGNATURE 0x0000DEAF
 
-typedef struct WsAudDemuxContext {
-    int audio_samplerate;
-    int audio_channels;
-    int audio_bits;
-    enum CodecID audio_type;
-    int audio_stream_index;
-    int64_t audio_frame_counter;
-} WsAudDemuxContext;
-
 static int wsaud_probe(AVProbeData *p)
 {
     int field;
@@ -79,7 +70,7 @@ static int wsaud_probe(AVProbeData *p)
 
     /* note: only check for WS IMA (type 99) right now since there is no
      * support for type 1 */
-    if (p->buf[11] != 99)
+    if (p->buf[11] != 99 && p->buf[11] != 1)
         return 0;
 
     /* read ahead to the first audio chunk and validate the first header signature */
@@ -93,41 +84,44 @@ static int wsaud_probe(AVProbeData *p)
 static int wsaud_read_header(AVFormatContext *s,
                              AVFormatParameters *ap)
 {
-    WsAudDemuxContext *wsaud = s->priv_data;
     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);
-    wsaud->audio_samplerate = AV_RL16(&header[0]);
-    if (header[11] == 99)
-        wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS;
-    else
-        return AVERROR_INVALIDDATA;
 
-    /* flag 0 indicates stereo */
-    wsaud->audio_channels = (header[10] & 0x1) + 1;
-    /* flag 1 indicates 16 bit audio */
-    wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8;
+    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);
-    avpriv_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
-    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-    st->codec->codec_id = wsaud->audio_type;
-    st->codec->codec_tag = 0;  /* no tag */
-    st->codec->channels = wsaud->audio_channels;
-    st->codec->sample_rate = wsaud->audio_samplerate;
-    st->codec->bits_per_coded_sample = wsaud->audio_bits;
-    st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-        st->codec->bits_per_coded_sample / 4;
-    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
-
-    wsaud->audio_stream_index = st->index;
-    wsaud->audio_frame_counter = 0;
+
+    switch (codec) {
+    case  1:
+        if (channels != 1) {
+            av_log_ask_for_sample(s, "Stereo WS-SND1 is not supported.\n");
+            return AVERROR_PATCHWELCOME;
+        }
+        st->codec->codec_id = CODEC_ID_WESTWOOD_SND1;
+        break;
+    case 99:
+        st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS;
+        st->codec->bits_per_coded_sample = 4;
+        st->codec->bit_rate = channels * sample_rate * 4;
+        break;
+    default:
+        av_log_ask_for_sample(s, "Unknown codec: %d\n", 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->sample_rate = sample_rate;
 
     return 0;
 }
@@ -135,11 +129,11 @@ static int wsaud_read_header(AVFormatContext *s,
 static int wsaud_read_packet(AVFormatContext *s,
                              AVPacket *pkt)
 {
-    WsAudDemuxContext *wsaud = s->priv_data;
     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)
@@ -150,15 +144,30 @@ static int wsaud_read_packet(AVFormatContext *s,
         return AVERROR_INVALIDDATA;
 
     chunk_size = AV_RL16(&preamble[0]);
-    ret= av_get_packet(pb, pkt, chunk_size);
-    if (ret != chunk_size)
-        return AVERROR(EIO);
-    pkt->stream_index = wsaud->audio_stream_index;
-    pkt->pts = wsaud->audio_frame_counter;
-    pkt->pts /= wsaud->audio_samplerate;
 
-    /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
-    wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;
+    if (st->codec->codec_id == 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;
 }
@@ -166,7 +175,6 @@ static int wsaud_read_packet(AVFormatContext *s,
 AVInputFormat ff_wsaud_demuxer = {
     .name           = "wsaud",
     .long_name      = NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
-    .priv_data_size = sizeof(WsAudDemuxContext),
     .read_probe     = wsaud_probe,
     .read_header    = wsaud_read_header,
     .read_packet    = wsaud_read_packet,

libavformat/westwood_vqa.c

@@ -51,18 +51,12 @@
 #define CMDS_TAG MKBETAG('C', 'M', 'D', 'S')
 
 #define VQA_HEADER_SIZE 0x2A
-#define VQA_FRAMERATE 15
 #define VQA_PREAMBLE_SIZE 8
 
 typedef struct WsVqaDemuxContext {
-    int audio_samplerate;
     int audio_channels;
-    int audio_bits;
-
     int audio_stream_index;
     int video_stream_index;
-
-    int64_t audio_frame_counter;
 } WsVqaDemuxContext;
 
 static int wsvqa_probe(AVProbeData *p)
@@ -89,12 +83,13 @@ static int wsvqa_read_header(AVFormatContext *s,
     unsigned char scratch[VQA_PREAMBLE_SIZE];
     unsigned int chunk_tag;
     unsigned int chunk_size;
+    int fps, version, flags, sample_rate, channels;
 
     /* initialize the video decoder stream */
     st = avformat_new_stream(s, NULL);
     if (!st)
         return AVERROR(ENOMEM);
-    avpriv_set_pts_info(st, 33, 1, VQA_FRAMERATE);
+    st->start_time = 0;
     wsvqa->video_stream_index = st->index;
     st->codec->codec_type = AVMEDIA_TYPE_VIDEO;
     st->codec->codec_id = CODEC_ID_WS_VQA;
@@ -114,34 +109,59 @@ static int wsvqa_read_header(AVFormatContext *s,
     }
     st->codec->width = AV_RL16(&header[6]);
     st->codec->height = AV_RL16(&header[8]);
+    fps = header[12];
+    if (fps < 1 || fps > 30) {
+        av_log(s, AV_LOG_ERROR, "invalid fps: %d\n", fps);
+        return AVERROR_INVALIDDATA;
+    }
+    avpriv_set_pts_info(st, 64, 1, fps);
 
     /* initialize the audio decoder stream for VQA v1 or nonzero samplerate */
-    if (AV_RL16(&header[24]) || (AV_RL16(&header[0]) == 1 && AV_RL16(&header[2]) == 1)) {
+    version     = AV_RL16(&header[ 0]);
+    flags       = AV_RL16(&header[ 2]);
+    sample_rate = AV_RL16(&header[24]);
+    channels    =          header[26];
+    if (sample_rate || (version == 1 && flags == 1)) {
         st = avformat_new_stream(s, NULL);
         if (!st)
             return AVERROR(ENOMEM);
-        avpriv_set_pts_info(st, 33, 1, VQA_FRAMERATE);
+        st->start_time = 0;
         st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-        if (AV_RL16(&header[0]) == 1)
+
+        st->codec->extradata_size = VQA_HEADER_SIZE;
+        st->codec->extradata = av_mallocz(VQA_HEADER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
+        if (!st->codec->extradata)
+            return AVERROR(ENOMEM);
+        memcpy(st->codec->extradata, header, VQA_HEADER_SIZE);
+
+        if (!sample_rate)
+            sample_rate = 22050;
+        st->codec->sample_rate = sample_rate;
+        avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
+
+        if (!channels)
+            channels = 1;
+        st->codec->channels = channels;
+
+        switch (version) {
+        case 1:
             st->codec->codec_id = CODEC_ID_WESTWOOD_SND1;
-        else
+            break;
+        case 2:
+        case 3:
             st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS;
-        st->codec->codec_tag = 0;  /* no tag */
-        st->codec->sample_rate = AV_RL16(&header[24]);
-        if (!st->codec->sample_rate)
-            st->codec->sample_rate = 22050;
-        st->codec->channels = header[26];
-        if (!st->codec->channels)
-            st->codec->channels = 1;
-        st->codec->bits_per_coded_sample = 16;
-        st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-            st->codec->bits_per_coded_sample / 4;
-        st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
+            st->codec->bits_per_coded_sample = 4;
+            st->codec->bit_rate = channels * sample_rate * 4;
+            break;
+        default:
+            /* NOTE: version 0 is supposedly raw pcm_u8 or pcm_s16le, but we do
+                     not have any samples to validate this */
+            av_log_ask_for_sample(s, "VQA version %d audio\n", version);
+            return AVERROR_PATCHWELCOME;
+        }
 
         wsvqa->audio_stream_index = st->index;
-        wsvqa->audio_samplerate = st->codec->sample_rate;
         wsvqa->audio_channels = st->codec->channels;
-        wsvqa->audio_frame_counter = 0;
     }
 
     /* there are 0 or more chunks before the FINF chunk; iterate until
@@ -208,13 +228,14 @@ static int wsvqa_read_packet(AVFormatContext *s,
             if (chunk_type == SND2_TAG) {
                 pkt->stream_index = wsvqa->audio_stream_index;
                 /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
-                wsvqa->audio_frame_counter += (chunk_size * 2) / wsvqa->audio_channels;
+                pkt->duration = (chunk_size * 2) / wsvqa->audio_channels;
             } else if(chunk_type == SND1_TAG) {
                 pkt->stream_index = wsvqa->audio_stream_index;
                 /* unpacked size is stored in header */
-                wsvqa->audio_frame_counter += AV_RL16(pkt->data) / wsvqa->audio_channels;
+                pkt->duration = AV_RL16(pkt->data) / wsvqa->audio_channels;
             } else {
                 pkt->stream_index = wsvqa->video_stream_index;
+                pkt->duration = 1;
             }
             /* stay on 16-bit alignment */
             if (skip_byte)

libavformat/yop.c

@@ -72,7 +72,7 @@ static int yop_read_header(AVFormatContext *s, AVFormatParameters *ap)
     // Audio
     audio_dec               = audio_stream->codec;
     audio_dec->codec_type   = AVMEDIA_TYPE_AUDIO;
-    audio_dec->codec_id     = CODEC_ID_ADPCM_IMA_WS;
+    audio_dec->codec_id     = CODEC_ID_ADPCM_IMA_APC;
     audio_dec->channels     = 1;
     audio_dec->sample_rate  = 22050;