Merge remote-tracking branch 'qatar/master'

* qatar/master: (46 commits)
  mtv: Make sure audio_subsegments is not 0
  v4l2: use V4L2_FMT_FLAG_EMULATED only if it is defined
  avconv: add symbolic names for -vsync parameters
  flvdec: Fix compiler warning for uninitialized variables
  rtsp: Fix compiler warning for uninitialized variable
  ulti: convert to new bytestream API.
  swscale: Use standard multiple inclusion guards in ppc/ header files.
  Place some START_TIMER invocations in separate blocks.
  v4l2: list available formats
  v4l2: set the proper codec_tag
  v4l2: refactor device_open
  v4l2: simplify away io_method
  v4l2: cosmetics
  v4l2: uniform and format options
  v4l2: do not force interlaced mode
  avio: exit early in fill_buffer without read_packet
  vc1dec: fix invalid memory access for small video dimensions
  rv34: fix invalid memory access for small video dimensions
  rv34: joint coefficient decoding and dequantization
  avplay: Don't call avio_set_interrupt_cb(NULL)
  ...

Conflicts:
	Changelog
	avconv.c
	doc/APIchanges
	doc/indevs.texi
	libavcodec/adxenc.c
	libavcodec/dnxhdenc.c
	libavcodec/h264.c
	libavdevice/v4l2.c
	libavformat/flvdec.c
	libavformat/mtv.c
	libswscale/utils.c

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

1 files changed, 77 insertions, 108 deletions

diff --git a/libavcodec/adxenc.c b/libavcodec/adxenc.c
index 51545fa06c..b029e2eb78 100644
--- a/libavcodec/adxenc.c
+++ b/libavcodec/adxenc.c
@@ -19,9 +19,9 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include "libavutil/intreadwrite.h"
 #include "avcodec.h"
 #include "adx.h"
+#include "bytestream.h"
 #include "put_bits.h"
 
 /**
@@ -33,167 +33,135 @@
  * adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
  */
 
-/* 18 bytes <-> 32 samples */
-
-static void adx_encode(ADXContext *c, unsigned char *adx, const short *wav,
-                       ADXChannelState *prev)
+static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav,
+                       ADXChannelState *prev, int channels)
 {
     PutBitContext pb;
     int scale;
-    int i;
-    int s0,s1,s2,d;
-    int max=0;
-    int min=0;
-    int data[32];
+    int i, j;
+    int s0, s1, s2, d;
+    int max = 0;
+    int min = 0;
+    int data[BLOCK_SAMPLES];
 
     s1 = prev->s1;
     s2 = prev->s2;
-    for(i=0;i<32;i++) {
+    for (i = 0, j = 0; j < 32; i += channels, j++) {
         s0 = wav[i];
         d = ((s0 << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
-        data[i]=d;
-        if (max<d) max=d;
-        if (min>d) min=d;
+        data[j] = d;
+        if (max < d)
+            max = d;
+        if (min > d)
+            min = d;
         s2 = s1;
         s1 = s0;
     }
     prev->s1 = s1;
     prev->s2 = s2;
 
-    /* -8..+7 */
-
-    if (max==0 && min==0) {
-        memset(adx,0,18);
+    if (max == 0 && min == 0) {
+        memset(adx, 0, BLOCK_SIZE);
         return;
     }
 
-    if (max/7>-min/8) scale = max/7;
-    else scale = -min/8;
+    if (max / 7 > -min / 8)
+        scale = max / 7;
+    else
+        scale = -min / 8;
 
-    if (scale==0) scale=1;
+    if (scale == 0)
+        scale = 1;
 
     AV_WB16(adx, scale);
 
     init_put_bits(&pb, adx + 2, 16);
-    for (i = 0; i < 32; i++)
-        put_sbits(&pb, 4, av_clip(data[i]/scale, -8, 7));
+    for (i = 0; i < BLOCK_SAMPLES; i++)
+        put_sbits(&pb, 4, av_clip(data[i] / scale, -8, 7));
     flush_put_bits(&pb);
 }
 
-static int adx_encode_header(AVCodecContext *avctx,unsigned char *buf,size_t bufsize)
+#define HEADER_SIZE 36
+
+static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
 {
-#if 0
-    struct {
-        uint32_t offset; /* 0x80000000 + sample start - 4 */
-        unsigned char unknown1[3]; /* 03 12 04 */
-        unsigned char channel; /* 1 or 2 */
-        uint32_t freq;
-        uint32_t size;
-        uint32_t unknown2; /* 01 f4 03 00 */
-        uint32_t unknown3; /* 00 00 00 00 */
-        uint32_t unknown4; /* 00 00 00 00 */
-
-    /* if loop
-        unknown3 00 15 00 01
-        unknown4 00 00 00 01
-        long loop_start_sample;
-        long loop_start_byte;
-        long loop_end_sample;
-        long loop_end_byte;
-        long
-    */
-    } adxhdr; /* big endian */
-    /* offset-6 "(c)CRI" */
-#endif
     ADXContext *c = avctx->priv_data;
 
-    AV_WB32(buf+0x00,0x80000000|0x20);
-    AV_WB32(buf+0x04,0x03120400|avctx->channels);
-    AV_WB32(buf+0x08,avctx->sample_rate);
-    AV_WB32(buf+0x0c,0); /* FIXME: set after */
-    AV_WB16(buf + 0x10, c->cutoff);
-    AV_WB32(buf + 0x12, 0x03000000);
-    AV_WB32(buf + 0x16, 0x00000000);
-    AV_WB32(buf + 0x1a, 0x00000000);
-    memcpy (buf + 0x1e, "(c)CRI", 6);
-    return 0x20+4;
+    if (bufsize < HEADER_SIZE)
+        return AVERROR(EINVAL);
+
+    bytestream_put_be16(&buf, 0x8000);              /* header signature */
+    bytestream_put_be16(&buf, HEADER_SIZE - 4);     /* copyright offset */
+    bytestream_put_byte(&buf, 3);                   /* encoding */
+    bytestream_put_byte(&buf, BLOCK_SIZE);          /* block size */
+    bytestream_put_byte(&buf, 4);                   /* sample size */
+    bytestream_put_byte(&buf, avctx->channels);     /* channels */
+    bytestream_put_be32(&buf, avctx->sample_rate);  /* sample rate */
+    bytestream_put_be32(&buf, 0);                   /* total sample count */
+    bytestream_put_be16(&buf, c->cutoff);           /* cutoff frequency */
+    bytestream_put_byte(&buf, 3);                   /* version */
+    bytestream_put_byte(&buf, 0);                   /* flags */
+    bytestream_put_be32(&buf, 0);                   /* unknown */
+    bytestream_put_be32(&buf, 0);                   /* loop enabled */
+    bytestream_put_be16(&buf, 0);                   /* padding */
+    bytestream_put_buffer(&buf, "(c)CRI", 6);       /* copyright signature */
+
+    return HEADER_SIZE;
 }
 
 static av_cold int adx_encode_init(AVCodecContext *avctx)
 {
     ADXContext *c = avctx->priv_data;
 
-    if (avctx->channels > 2)
-        return -1; /* only stereo or mono =) */
-    avctx->frame_size = 32;
-
-    avctx->coded_frame= avcodec_alloc_frame();
-    avctx->coded_frame->key_frame= 1;
+    if (avctx->channels > 2) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
+        return AVERROR(EINVAL);
+    }
+    avctx->frame_size = BLOCK_SAMPLES;
 
-//    avctx->bit_rate = avctx->sample_rate*avctx->channels*18*8/32;
+    avctx->coded_frame = avcodec_alloc_frame();
 
     /* the cutoff can be adjusted, but this seems to work pretty well */
     c->cutoff = 500;
     ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
 
-    av_log(avctx, AV_LOG_DEBUG, "adx encode init\n");
-
     return 0;
 }
 
 static av_cold int adx_encode_close(AVCodecContext *avctx)
 {
     av_freep(&avctx->coded_frame);
-
     return 0;
 }
 
-static int adx_encode_frame(AVCodecContext *avctx,
-                uint8_t *frame, int buf_size, void *data)
+static int adx_encode_frame(AVCodecContext *avctx, uint8_t *frame,
+                            int buf_size, void *data)
 {
-    ADXContext *c = avctx->priv_data;
-    const short *samples = data;
-    unsigned char *dst = frame;
-    int rest = avctx->frame_size;
-
-/*
-    input data size =
-    ffmpeg.c: do_audio_out()
-    frame_bytes = enc->frame_size * 2 * enc->channels;
-*/
+    ADXContext *c          = avctx->priv_data;
+    const int16_t *samples = data;
+    uint8_t *dst           = frame;
+    int ch;
 
-//    printf("sz=%d ",buf_size); fflush(stdout);
     if (!c->header_parsed) {
-        int hdrsize = adx_encode_header(avctx,dst,buf_size);
-        dst+=hdrsize;
+        int hdrsize;
+        if ((hdrsize = adx_encode_header(avctx, dst, buf_size)) < 0) {
+            av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
+            return AVERROR(EINVAL);
+        }
+        dst      += hdrsize;
+        buf_size -= hdrsize;
         c->header_parsed = 1;
     }
+    if (buf_size < BLOCK_SIZE * avctx->channels) {
+        av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
+        return AVERROR(EINVAL);
+    }
 
-    if (avctx->channels==1) {
-        while(rest>=32) {
-            adx_encode(c, dst, samples, c->prev);
-            dst+=18;
-            samples+=32;
-            rest-=32;
-        }
-    } else {
-        while(rest>=32*2) {
-            short tmpbuf[32*2];
-            int i;
-
-            for(i=0;i<32;i++) {
-                tmpbuf[i] = samples[i*2];
-                tmpbuf[i+32] = samples[i*2+1];
-            }
-
-            adx_encode(c, dst,      tmpbuf,      c->prev);
-            adx_encode(c, dst + 18, tmpbuf + 32, c->prev + 1);
-            dst+=18*2;
-            samples+=32*2;
-            rest-=32*2;
-        }
+    for (ch = 0; ch < avctx->channels; ch++) {
+        adx_encode(c, dst, samples + ch, &c->prev[ch], avctx->channels);
+        dst += BLOCK_SIZE;
     }
-    return dst-frame;
+    return dst - frame;
 }
 
 AVCodec ff_adpcm_adx_encoder = {
@@ -204,6 +172,7 @@ AVCodec ff_adpcm_adx_encoder = {
     .init           = adx_encode_init,
     .encode         = adx_encode_frame,
     .close          = adx_encode_close,
-    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
-    .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
+                                                      AV_SAMPLE_FMT_NONE },
+    .long_name      = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
 };