Merge remote-tracking branch 'qatar/master'

* qatar/master: (44 commits)
  replacement Indeo 3 decoder
  gsm demuxer: do not allocate packet twice.
  flvenc: use first packet delay as global delay.
  ac3enc: doxygen update.
  imc: return error codes instead of 0 for error conditions.
  imc: return meaningful error codes instead of -1
  imc: do not set channel layout for stereo
  imc: validate channel count
  imc: check for ff_fft_init() failure
  imc: check output buffer size before decoding
  imc: use DSPContext.bswap16_buf() to byte-swap packet data
  rtsp: add allowed_media_types option
  libgsm: add flush function to reset the decoder state when seeking
  libgsm: simplify decoding by using a loop
  gsm: log error message when packet is too small
  libgsmdec: do not needlessly set *data_size to 0
  gsmdec: do not needlessly set *data_size to 0
  gsmdec: add flush function to reset the decoder state when seeking
  libgsmdec: check output buffer size before decoding
  gsmdec: log error message when output buffer is too small.
  ...

Conflicts:
	Changelog
	ffplay.c
	libavcodec/indeo3.c
	libavcodec/mjpeg_parser.c
	libavcodec/vp3.c
	libavformat/cutils.c
	libavformat/id3v2.c
	libavutil/parseutils.c

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

1 files changed, 111 insertions, 49 deletions

diff --git a/libavcodec/g726.c b/libavcodec/g726.c
index 2ce113b24b..ae1b5a3001 100644
--- a/libavcodec/g726.c
+++ b/libavcodec/g726.c
@@ -22,7 +22,10 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 #include <limits.h>
+#include "libavutil/avassert.h"
+#include "libavutil/opt.h"
 #include "avcodec.h"
+#include "internal.h"
 #include "get_bits.h"
 #include "put_bits.h"
 
@@ -71,6 +74,7 @@ typedef struct G726Tables {
 } G726Tables;
 
 typedef struct G726Context {
+    AVClass *class;
     G726Tables tbls;    /**< static tables needed for computation */
 
     Float11 sr[2];      /**< prev. reconstructed samples */
@@ -266,11 +270,11 @@ static int16_t g726_decode(G726Context* c, int I)
     return av_clip(re_signal << 2, -0xffff, 0xffff);
 }
 
-static av_cold int g726_reset(G726Context* c, int index)
+static av_cold int g726_reset(G726Context *c)
 {
     int i;
 
-    c->tbls = G726Tables_pool[index];
+    c->tbls = G726Tables_pool[c->code_size - 2];
     for (i=0; i<2; i++) {
         c->sr[i].mant = 1<<5;
         c->pk[i] = 1;
@@ -295,65 +299,59 @@ static int16_t g726_encode(G726Context* c, int16_t sig)
     g726_decode(c, i);
     return i;
 }
-#endif
 
 /* Interfacing to the libavcodec */
 
-static av_cold int g726_init(AVCodecContext * avctx)
+static av_cold int g726_encode_init(AVCodecContext *avctx)
 {
     G726Context* c = avctx->priv_data;
-    unsigned int index;
 
-    if (avctx->sample_rate <= 0) {
-        av_log(avctx, AV_LOG_ERROR, "Samplerate is invalid\n");
-        return -1;
+    if (avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL &&
+        avctx->sample_rate != 8000) {
+        av_log(avctx, AV_LOG_ERROR, "Sample rates other than 8kHz are not "
+               "allowed when the compliance level is higher than unofficial. "
+               "Resample or reduce the compliance level.\n");
+        return AVERROR(EINVAL);
     }
+    av_assert0(avctx->sample_rate > 0);
 
-    index = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate - 2;
-
-    if (avctx->bit_rate % avctx->sample_rate && avctx->codec->encode) {
-        av_log(avctx, AV_LOG_ERROR, "Bitrate - Samplerate combination is invalid\n");
-        return -1;
-    }
     if(avctx->channels != 1){
         av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
-        return -1;
-    }
-    if(index>3){
-        av_log(avctx, AV_LOG_ERROR, "Unsupported number of bits %d\n", index+2);
-        return -1;
+        return AVERROR(EINVAL);
     }
-    g726_reset(c, index);
-    c->code_size = index+2;
+
+    if (avctx->bit_rate)
+        c->code_size = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate;
+
+    c->code_size = av_clip(c->code_size, 2, 5);
+    avctx->bit_rate = c->code_size * avctx->sample_rate;
+    avctx->bits_per_coded_sample = c->code_size;
+
+    g726_reset(c);
 
     avctx->coded_frame = avcodec_alloc_frame();
     if (!avctx->coded_frame)
         return AVERROR(ENOMEM);
     avctx->coded_frame->key_frame = 1;
 
-    if (avctx->codec->decode)
-        avctx->sample_fmt = AV_SAMPLE_FMT_S16;
-
     /* select a frame size that will end on a byte boundary and have a size of
        approximately 1024 bytes */
-    if (avctx->codec->encode)
-        avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[index];
+    avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[c->code_size - 2];
 
     return 0;
 }
 
-static av_cold int g726_close(AVCodecContext *avctx)
+static av_cold int g726_encode_close(AVCodecContext *avctx)
 {
     av_freep(&avctx->coded_frame);
     return 0;
 }
 
-#if CONFIG_ADPCM_G726_ENCODER
 static int g726_encode_frame(AVCodecContext *avctx,
                             uint8_t *dst, int buf_size, void *data)
 {
     G726Context *c = avctx->priv_data;
-    const short *samples = data;
+    const int16_t *samples = data;
     PutBitContext pb;
     int i;
 
@@ -366,8 +364,72 @@ static int g726_encode_frame(AVCodecContext *avctx,
 
     return put_bits_count(&pb)>>3;
 }
+
+#define OFFSET(x) offsetof(G726Context, x)
+#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
+static const AVOption options[] = {
+    { "code_size", "Bits per code", OFFSET(code_size), AV_OPT_TYPE_INT, { 4 }, 2, 5, AE },
+    { NULL },
+};
+
+static const AVClass class = {
+    .class_name = "g726",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+static const AVCodecDefault defaults[] = {
+    { "b", "0" },
+    { NULL },
+};
+
+AVCodec ff_adpcm_g726_encoder = {
+    .name           = "g726",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_ADPCM_G726,
+    .priv_data_size = sizeof(G726Context),
+    .init           = g726_encode_init,
+    .encode         = g726_encode_frame,
+    .close          = g726_encode_close,
+    .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
+    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+    .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
+    .priv_class     = &class,
+    .defaults       = defaults,
+};
 #endif
 
+#if CONFIG_ADPCM_G726_DECODER
+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);
+    }
+
+    c->code_size = avctx->bits_per_coded_sample;
+    if (c->code_size < 2 || c->code_size > 5) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid number of bits %d\n", c->code_size);
+        return AVERROR(EINVAL);
+    }
+    g726_reset(c);
+
+    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+
+    return 0;
+}
+
 static int g726_decode_frame(AVCodecContext *avctx,
                              void *data, int *data_size,
                              AVPacket *avpkt)
@@ -375,43 +437,43 @@ static int g726_decode_frame(AVCodecContext *avctx,
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
     G726Context *c = avctx->priv_data;
-    short *samples = data;
+    int16_t *samples = data;
     GetBitContext gb;
+    int out_samples, out_size;
+
+    out_samples = buf_size * 8 / c->code_size;
+    out_size    = out_samples * av_get_bytes_per_sample(avctx->sample_fmt);
+    if (*data_size < out_size) {
+        av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
+        return AVERROR(EINVAL);
+    }
 
     init_get_bits(&gb, buf, buf_size * 8);
 
-    while (get_bits_count(&gb) + c->code_size <= buf_size*8)
+    while (out_samples--)
         *samples++ = g726_decode(c, get_bits(&gb, c->code_size));
 
-    if(buf_size*8 != get_bits_count(&gb))
+    if (get_bits_left(&gb) > 0)
         av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
 
-    *data_size = (uint8_t*)samples - (uint8_t*)data;
+    *data_size = out_size;
     return buf_size;
 }
 
-#if CONFIG_ADPCM_G726_ENCODER
-AVCodec ff_adpcm_g726_encoder = {
-    .name           = "g726",
-    .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = CODEC_ID_ADPCM_G726,
-    .priv_data_size = sizeof(G726Context),
-    .init           = g726_init,
-    .encode         = g726_encode_frame,
-    .close          = g726_close,
-    .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
-    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
-    .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
-};
-#endif
+static void g726_decode_flush(AVCodecContext *avctx)
+{
+    G726Context *c = avctx->priv_data;
+    g726_reset(c);
+}
 
 AVCodec ff_adpcm_g726_decoder = {
     .name           = "g726",
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = CODEC_ID_ADPCM_G726,
     .priv_data_size = sizeof(G726Context),
-    .init           = g726_init,
-    .close          = g726_close,
+    .init           = g726_decode_init,
     .decode         = g726_decode_frame,
+    .flush          = g726_decode_flush,
     .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
 };
+#endif