Merge remote-tracking branch 'qatar/master'

* qatar/master:
  mov: Don't av_malloc(0).
  avconv: only allocate 1 AVFrame per input stream
  avconv: fix memleaks due to not freeing the AVFrame for audio
  h264-fate: remove -strict 1 except where necessary (mr4/5-tandberg).
  misc Doxygen markup improvements
  doxygen: eliminate Qt-style doxygen syntax
  g722: Add a regression test for muxing/demuxing in wav
  g722: Change bits per sample to 4
  g722dec: Signal skipping the lower bits via AVOptions instead of bits_per_coded_sample
  api-example: update to use avcodec_decode_audio4()
  avplay: use avcodec_decode_audio4()
  avplay: use a separate buffer for playing silence
  avformat: use avcodec_decode_audio4() in avformat_find_stream_info()
  avconv: use avcodec_decode_audio4() instead of avcodec_decode_audio3()
  mov: Allow empty stts atom.
  doc: document preferred Doxygen syntax and make patcheck detect it

Conflicts:
	avconv.c
	ffplay.c
	libavcodec/mlpdec.c
	libavcodec/version.h
	libavformat/mov.c
	tests/codec-regression.sh
	tests/fate/h264.mak

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

13 files changed, 69 insertions, 64 deletions

diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c
index 0a4a7e6dda..57c8ae9ae5 100644
--- a/libavcodec/amrnbdec.c
+++ b/libavcodec/amrnbdec.c
@@ -653,7 +653,7 @@ static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
 static void apply_ir_filter(float *out, const AMRFixed *in,
                             const float *filter)
 {
-    float filter1[AMR_SUBFRAME_SIZE],     //!< filters at pitch lag*1 and *2
+    float filter1[AMR_SUBFRAME_SIZE],     ///< filters at pitch lag*1 and *2
           filter2[AMR_SUBFRAME_SIZE];
     int   lag = in->pitch_lag;
     float fac = in->pitch_fac;
diff --git a/libavcodec/cinepak.c b/libavcodec/cinepak.c
index ebdbb5eaec..1181c99e7e 100644
--- a/libavcodec/cinepak.c
+++ b/libavcodec/cinepak.c
@@ -22,10 +22,11 @@
 /**
  * @file
  * Cinepak video decoder
- * by Ewald Snel <ewald@rambo.its.tudelft.nl>
- * For more information on the Cinepak algorithm, visit:
+ * @author Ewald Snel <ewald@rambo.its.tudelft.nl>
+ *
+ * @see For more information on the Cinepak algorithm, visit:
  *   http://www.csse.monash.edu.au/~timf/
- * For more information on the quirky data inside Sega FILM/CPK files, visit:
+ * @see For more information on the quirky data inside Sega FILM/CPK files, visit:
  *   http://wiki.multimedia.cx/index.php?title=Sega_FILM
  */
 
diff --git a/libavcodec/eamad.c b/libavcodec/eamad.c
index ebad3b336c..1f6282394e 100644
--- a/libavcodec/eamad.c
+++ b/libavcodec/eamad.c
@@ -22,9 +22,9 @@
 /**
  * @file
  * Electronic Arts Madcow Video Decoder
- * by Peter Ross <pross@xvid.org>
+ * @author Peter Ross <pross@xvid.org>
  *
- * Technical details here:
+ * @see technical details at
  * http://wiki.multimedia.cx/index.php?title=Electronic_Arts_MAD
  */
 
diff --git a/libavcodec/g722.h b/libavcodec/g722.h
index 69e7a86e25..eb3b9b872e 100644
--- a/libavcodec/g722.h
+++ b/libavcodec/g722.h
@@ -31,7 +31,9 @@
 #define PREV_SAMPLES_BUF_SIZE 1024
 
 typedef struct {
+    const AVClass *class;
     AVFrame frame;
+    int     bits_per_codeword;
     int16_t prev_samples[PREV_SAMPLES_BUF_SIZE]; ///< memory of past decoded samples
     int     prev_samples_pos;        ///< the number of values in prev_samples
 
diff --git a/libavcodec/g722dec.c b/libavcodec/g722dec.c
index 652a1aa4ae..50a224ba10 100644
--- a/libavcodec/g722dec.c
+++ b/libavcodec/g722dec.c
@@ -37,6 +37,21 @@
 #include "avcodec.h"
 #include "get_bits.h"
 #include "g722.h"
+#include "libavutil/opt.h"
+
+#define OFFSET(x) offsetof(G722Context, x)
+#define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
+static const AVOption options[] = {
+    { "bits_per_codeword", "Bits per G722 codeword", OFFSET(bits_per_codeword), AV_OPT_TYPE_FLAGS, { 8 }, 6, 8, AD },
+    { NULL }
+};
+
+static const AVClass g722_decoder_class = {
+    .class_name = "g722 decoder",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
 
 static av_cold int g722_decode_init(AVCodecContext * avctx)
 {
@@ -48,20 +63,6 @@ static av_cold int g722_decode_init(AVCodecContext * avctx)
     }
     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
 
-    switch (avctx->bits_per_coded_sample) {
-    case 8:
-    case 7:
-    case 6:
-        break;
-    default:
-        av_log(avctx, AV_LOG_WARNING, "Unsupported bits_per_coded_sample [%d], "
-                                      "assuming 8\n",
-                                      avctx->bits_per_coded_sample);
-    case 0:
-        avctx->bits_per_coded_sample = 8;
-        break;
-    }
-
     c->band[0].scale_factor = 8;
     c->band[1].scale_factor = 2;
     c->prev_samples_pos = 22;
@@ -89,7 +90,7 @@ static int g722_decode_frame(AVCodecContext *avctx, void *data,
     G722Context *c = avctx->priv_data;
     int16_t *out_buf;
     int j, ret;
-    const int skip = 8 - avctx->bits_per_coded_sample;
+    const int skip = 8 - c->bits_per_codeword;
     const int16_t *quantizer_table = low_inv_quants[skip];
     GetBitContext gb;
 
@@ -149,4 +150,5 @@ AVCodec ff_adpcm_g722_decoder = {
     .decode         = g722_decode_frame,
     .capabilities   = CODEC_CAP_DR1,
     .long_name      = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
+    .priv_class     = &g722_decoder_class,
 };
diff --git a/libavcodec/g722enc.c b/libavcodec/g722enc.c
index bdc30d5d07..f8db49aba8 100644
--- a/libavcodec/g722enc.c
+++ b/libavcodec/g722enc.c
@@ -139,7 +139,7 @@ static int g722_encode_trellis(AVCodecContext *avctx,
         nodes[i][0]->state = c->band[i];
     }
 
-    for (i = 0; i < buf_size >> 1; i++) {
+    for (i = 0; i < buf_size; i++) {
         int xlow, xhigh;
         struct TrellisNode *next[2];
         int heap_pos[2] = {0, 0};
@@ -285,7 +285,7 @@ static int g722_encode_frame(AVCodecContext *avctx,
     if (avctx->trellis)
         return g722_encode_trellis(avctx, dst, buf_size, data);
 
-    for (i = 0; i < buf_size >> 1; i++) {
+    for (i = 0; i < buf_size; i++) {
         int xlow, xhigh, ihigh, ilow;
         filter_samples(c, &samples[2*i], &xlow, &xhigh);
         ihigh = encode_high(&c->band[1], xhigh);
diff --git a/libavcodec/ivi_common.h b/libavcodec/ivi_common.h
index 10cca26045..c4c7c3da4d 100644
--- a/libavcodec/ivi_common.h
+++ b/libavcodec/ivi_common.h
@@ -51,7 +51,7 @@ typedef struct {
                             /// or "7" for custom one
     VLC         *tab;       /// pointer to the table associated with tab_sel
 
-    //! the following are used only when tab_sel == 7
+    /// the following are used only when tab_sel == 7
     IVIHuffDesc cust_desc;  /// custom Huffman codebook descriptor
     VLC         cust_tab;   /// vlc table for custom codebook
 } IVIHuffTab;
diff --git a/libavcodec/lsp.c b/libavcodec/lsp.c
index 374539a029..7fda12ee62 100644
--- a/libavcodec/lsp.c
+++ b/libavcodec/lsp.c
@@ -75,7 +75,7 @@ void ff_acelp_lsf2lspd(double *lsp, const float *lsf, int lp_order)
 
 /**
  * @brief decodes polynomial coefficients from LSP
- * @param f [out] decoded polynomial coefficients (-0x20000000 <= (3.22) <= 0x1fffffff)
+ * @param[out] f decoded polynomial coefficients (-0x20000000 <= (3.22) <= 0x1fffffff)
  * @param lsp LSP coefficients (-0x8000 <= (0.15) <= 0x7fff)
  */
 static void lsp2poly(int* f, const int16_t* lsp, int lp_half_order)
diff --git a/libavcodec/mlpdec.c b/libavcodec/mlpdec.c
index 4b439ddb67..e91295084c 100644
--- a/libavcodec/mlpdec.c
+++ b/libavcodec/mlpdec.c
@@ -45,35 +45,35 @@ static const char* sample_message =
     "a sample of this file.";
 
 typedef struct SubStream {
-    //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
+    /// Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
     uint8_t     restart_seen;
 
     //@{
     /** restart header data */
-    //! The type of noise to be used in the rematrix stage.
+    /// The type of noise to be used in the rematrix stage.
     uint16_t    noise_type;
 
-    //! The index of the first channel coded in this substream.
+    /// The index of the first channel coded in this substream.
     uint8_t     min_channel;
-    //! The index of the last channel coded in this substream.
+    /// The index of the last channel coded in this substream.
     uint8_t     max_channel;
-    //! The number of channels input into the rematrix stage.
+    /// The number of channels input into the rematrix stage.
     uint8_t     max_matrix_channel;
-    //! For each channel output by the matrix, the output channel to map it to
+    /// For each channel output by the matrix, the output channel to map it to
     uint8_t     ch_assign[MAX_CHANNELS];
 
-    //! Channel coding parameters for channels in the substream
+    /// Channel coding parameters for channels in the substream
     ChannelParams channel_params[MAX_CHANNELS];
 
-    //! The left shift applied to random noise in 0x31ea substreams.
+    /// The left shift applied to random noise in 0x31ea substreams.
     uint8_t     noise_shift;
-    //! The current seed value for the pseudorandom noise generator(s).
+    /// The current seed value for the pseudorandom noise generator(s).
     uint32_t    noisegen_seed;
 
-    //! Set if the substream contains extra info to check the size of VLC blocks.
+    /// Set if the substream contains extra info to check the size of VLC blocks.
     uint8_t     data_check_present;
 
-    //! Bitmask of which parameter sets are conveyed in a decoding parameter block.
+    /// Bitmask of which parameter sets are conveyed in a decoding parameter block.
     uint8_t     param_presence_flags;
 #define PARAM_BLOCKSIZE     (1 << 7)
 #define PARAM_MATRIX        (1 << 6)
@@ -88,32 +88,32 @@ typedef struct SubStream {
     //@{
     /** matrix data */
 
-    //! Number of matrices to be applied.
+    /// Number of matrices to be applied.
     uint8_t     num_primitive_matrices;
 
-    //! matrix output channel
+    /// matrix output channel
     uint8_t     matrix_out_ch[MAX_MATRICES];
 
-    //! Whether the LSBs of the matrix output are encoded in the bitstream.
+    /// Whether the LSBs of the matrix output are encoded in the bitstream.
     uint8_t     lsb_bypass[MAX_MATRICES];
-    //! Matrix coefficients, stored as 2.14 fixed point.
+    /// Matrix coefficients, stored as 2.14 fixed point.
     int32_t     matrix_coeff[MAX_MATRICES][MAX_CHANNELS];
-    //! Left shift to apply to noise values in 0x31eb substreams.
+    /// Left shift to apply to noise values in 0x31eb substreams.
     uint8_t     matrix_noise_shift[MAX_MATRICES];
     //@}
 
-    //! Left shift to apply to Huffman-decoded residuals.
+    /// Left shift to apply to Huffman-decoded residuals.
     uint8_t     quant_step_size[MAX_CHANNELS];
 
-    //! number of PCM samples in current audio block
+    /// number of PCM samples in current audio block
     uint16_t    blocksize;
-    //! Number of PCM samples decoded so far in this frame.
+    /// Number of PCM samples decoded so far in this frame.
     uint16_t    blockpos;
 
-    //! Left shift to apply to decoded PCM values to get final 24-bit output.
+    /// Left shift to apply to decoded PCM values to get final 24-bit output.
     int8_t      output_shift[MAX_CHANNELS];
 
-    //! Running XOR of all output samples.
+    /// Running XOR of all output samples.
     int32_t     lossless_check_data;
 
 } SubStream;
@@ -122,24 +122,24 @@ typedef struct MLPDecodeContext {
     AVCodecContext *avctx;
     AVFrame     frame;
 
-    //! Current access unit being read has a major sync.
+    /// Current access unit being read has a major sync.
     int         is_major_sync_unit;
 
-    //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
+    /// Set if a valid major sync block has been read. Otherwise no decoding is possible.
     uint8_t     params_valid;
 
-    //! Number of substreams contained within this stream.
+    /// Number of substreams contained within this stream.
     uint8_t     num_substreams;
 
-    //! Index of the last substream to decode - further substreams are skipped.
+    /// Index of the last substream to decode - further substreams are skipped.
     uint8_t     max_decoded_substream;
 
-    //! Stream needs channel reordering to comply with FFmpeg's channel order
+    /// Stream needs channel reordering to comply with FFmpeg's channel order
     uint8_t     needs_reordering;
 
-    //! number of PCM samples contained in each frame
+    /// number of PCM samples contained in each frame
     int         access_unit_size;
-    //! next power of two above the number of samples in each frame
+    /// next power of two above the number of samples in each frame
     int         access_unit_size_pow2;
 
     SubStream   substream[MAX_SUBSTREAMS];
diff --git a/libavcodec/qcelpdata.h b/libavcodec/qcelpdata.h
index e71ee9fdb7..4a7dcdb38b 100644
--- a/libavcodec/qcelpdata.h
+++ b/libavcodec/qcelpdata.h
@@ -40,16 +40,16 @@
 typedef struct {
 /// @name QCELP excitation codebook parameters
 /// @{
-    uint8_t cbsign[16]; ///!< sign of the codebook gain for each codebook subframe
-    uint8_t cbgain[16]; ///!< unsigned codebook gain for each codebook subframe
-    uint8_t cindex[16]; ///!< codebook index for each codebook subframe
+    uint8_t cbsign[16]; ///< sign of the codebook gain for each codebook subframe
+    uint8_t cbgain[16]; ///< unsigned codebook gain for each codebook subframe
+    uint8_t cindex[16]; ///< codebook index for each codebook subframe
 /// @}
 
 /// @name QCELP pitch prediction parameters
 /// @{
-    uint8_t plag[4];    ///!< pitch lag for each pitch subframe
-    uint8_t pfrac[4];   ///!< fractional pitch lag for each pitch subframe
-    uint8_t pgain[4];   ///!< pitch gain for each pitch subframe
+    uint8_t plag[4];    ///< pitch lag for each pitch subframe
+    uint8_t pfrac[4];   ///< fractional pitch lag for each pitch subframe
+    uint8_t pgain[4];   ///< pitch gain for each pitch subframe
 /// @}
 
     /**
@@ -266,7 +266,7 @@ static const QCELPBitmap qcelp_rate_octave_bitmap[] = {
  * the QCELPContext
  */
 static const QCELPBitmap * const qcelp_unpacking_bitmaps_per_rate[5] = {
-    NULL,                     ///!< for SILENCE rate
+    NULL,                     ///< for SILENCE rate
     qcelp_rate_octave_bitmap,
     qcelp_rate_quarter_bitmap,
     qcelp_rate_half_bitmap,
@@ -274,7 +274,7 @@ static const QCELPBitmap * const qcelp_unpacking_bitmaps_per_rate[5] = {
 };
 
 static const uint16_t qcelp_unpacking_bitmaps_lengths[5] = {
-    0, ///!< for SILENCE rate
+    0, ///< for SILENCE rate
     FF_ARRAY_ELEMS(qcelp_rate_octave_bitmap),
     FF_ARRAY_ELEMS(qcelp_rate_quarter_bitmap),
     FF_ARRAY_ELEMS(qcelp_rate_half_bitmap),
diff --git a/libavcodec/rtjpeg.c b/libavcodec/rtjpeg.c
index 07e4f02e67..e03282b412 100644
--- a/libavcodec/rtjpeg.c
+++ b/libavcodec/rtjpeg.c
@@ -27,7 +27,7 @@
     i = scan[coeff--]; \
     block[i] = (c) * quant[i];
 
-//! aligns the bitstream to the give power of two
+/// aligns the bitstream to the given power of two
 #define ALIGN(a) \
     n = (-get_bits_count(gb)) & (a - 1); \
     if (n) {skip_bits(gb, n);}
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index ae26590a69..1685b4d196 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -1468,8 +1468,8 @@ int av_get_bits_per_sample(enum CodecID codec_id){
     case CODEC_ID_ADPCM_SWF:
     case CODEC_ID_ADPCM_MS:
     case CODEC_ID_ADPCM_YAMAHA:
-        return 4;
     case CODEC_ID_ADPCM_G722:
+        return 4;
     case CODEC_ID_PCM_ALAW:
     case CODEC_ID_PCM_MULAW:
     case CODEC_ID_PCM_S8:
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 70dbd0001e..14b04bac01 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -21,7 +21,7 @@
 #define AVCODEC_VERSION_H
 
 #define LIBAVCODEC_VERSION_MAJOR 53
-#define LIBAVCODEC_VERSION_MINOR 40
+#define LIBAVCODEC_VERSION_MINOR 41
 #define LIBAVCODEC_VERSION_MICRO  0
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \