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>

avconv.c

@@ -155,8 +155,6 @@ static uint8_t *audio_buf;
 static uint8_t *audio_out;
 static unsigned int allocated_audio_out_size, allocated_audio_buf_size;
 
-static void *samples;
-
 #define DEFAULT_PASS_LOGFILENAME_PREFIX "av2pass"
 
 typedef struct InputStream {
@@ -165,6 +163,8 @@ typedef struct InputStream {
     int discard;             /* true if stream data should be discarded */
     int decoding_needed;     /* true if the packets must be decoded in 'raw_fifo' */
     AVCodec *dec;
+    AVFrame *decoded_frame;
+    AVFrame *filtered_frame;
 
     int64_t       start;     /* time when read started */
     int64_t       next_pts;  /* synthetic pts for cases where pkt.pts
@@ -612,8 +612,11 @@ void exit_program(int ret)
     for(i=0;i<nb_input_files;i++) {
         av_close_input_file(input_files[i].ctx);
     }
-    for (i = 0; i < nb_input_streams; i++)
+    for (i = 0; i < nb_input_streams; i++) {
+        av_freep(&input_streams[i].decoded_frame);
+        av_freep(&input_streams[i].filtered_frame);
         av_dict_free(&input_streams[i].opts);
+    }
 
     if (vstats_file)
         fclose(vstats_file);
@@ -628,7 +631,6 @@ void exit_program(int ret)
     av_free(audio_buf);
     av_free(audio_out);
     allocated_audio_buf_size= allocated_audio_out_size= 0;
-    av_free(samples);
 
 #if CONFIG_AVFILTER
     avfilter_uninit();
@@ -787,14 +789,11 @@ static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_
     memset(buf, fill_char, size);
 }
 
-static void do_audio_out(AVFormatContext *s,
-                         OutputStream *ost,
-                         InputStream *ist,
-                         unsigned char *buf, int size)
+static void do_audio_out(AVFormatContext *s, OutputStream *ost,
+                         InputStream *ist, AVFrame *decoded_frame)
 {
     uint8_t *buftmp;
     int64_t audio_out_size, audio_buf_size;
-    int64_t allocated_for_size= size;
 
     int size_out, frame_bytes, ret, resample_changed;
     AVCodecContext *enc= ost->st->codec;
@@ -802,6 +801,9 @@ static void do_audio_out(AVFormatContext *s,
     int osize = av_get_bytes_per_sample(enc->sample_fmt);
     int isize = av_get_bytes_per_sample(dec->sample_fmt);
     const int coded_bps = av_get_bits_per_sample(enc->codec->id);
+    uint8_t *buf = decoded_frame->data[0];
+    int size     = decoded_frame->nb_samples * dec->channels * isize;
+    int64_t allocated_for_size = size;
 
 need_realloc:
     audio_buf_size= (allocated_for_size + isize*dec->channels - 1) / (isize*dec->channels);
@@ -1697,39 +1699,42 @@ static void rate_emu_sleep(InputStream *ist)
 
 static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
 {
-    static unsigned int samples_size = 0;
+    AVFrame *decoded_frame;
+    AVCodecContext *avctx = ist->st->codec;
     int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
-    uint8_t *decoded_data_buf  = NULL;
-    int      decoded_data_size = 0;
     int i, ret;
 
-    if (pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) {
-        av_free(samples);
-        samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE);
-        samples      = av_malloc(samples_size);
-    }
-    decoded_data_size = samples_size;
+    if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))
+        return AVERROR(ENOMEM);
+    else
+        avcodec_get_frame_defaults(ist->decoded_frame);
+    decoded_frame = ist->decoded_frame;
 
-    ret = avcodec_decode_audio3(ist->st->codec, samples, &decoded_data_size,
-                                pkt);
-    if (ret < 0)
+    ret = avcodec_decode_audio4(avctx, decoded_frame, got_output, pkt);
+    if (ret < 0) {
         return ret;
-    *got_output  = decoded_data_size > 0;
+    }
 
-    /* Some bug in mpeg audio decoder gives */
-    /* decoded_data_size < 0, it seems they are overflows */
     if (!*got_output) {
         /* no audio frame */
         return ret;
     }
 
-    decoded_data_buf = (uint8_t *)samples;
-    ist->next_pts   += ((int64_t)AV_TIME_BASE/bps * decoded_data_size) /
-                       (ist->st->codec->sample_rate * ist->st->codec->channels);
+    /* if the decoder provides a pts, use it instead of the last packet pts.
+       the decoder could be delaying output by a packet or more. */
+    if (decoded_frame->pts != AV_NOPTS_VALUE)
+        ist->next_pts = decoded_frame->pts;
+
+    /* increment next_pts to use for the case where the input stream does not
+       have timestamps or there are multiple frames in the packet */
+    ist->next_pts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) /
+                     avctx->sample_rate;
 
     // preprocess audio (volume)
     if (audio_volume != 256) {
-        switch (ist->st->codec->sample_fmt) {
+        int decoded_data_size = decoded_frame->nb_samples * avctx->channels * bps;
+        void *samples = decoded_frame->data[0];
+        switch (avctx->sample_fmt) {
         case AV_SAMPLE_FMT_U8:
         {
             uint8_t *volp = samples;
@@ -1790,9 +1795,9 @@ static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
 
         if (!check_output_constraints(ist, ost) || !ost->encoding_needed)
             continue;
-        do_audio_out(output_files[ost->file_index].ctx, ost, ist,
-                     decoded_data_buf, decoded_data_size);
+        do_audio_out(output_files[ost->file_index].ctx, ost, ist, decoded_frame);
     }
+
     return ret;
 }
 
@@ -1806,8 +1811,11 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
     int frame_available = 1;
 #endif
 
-    if (!(decoded_frame = avcodec_alloc_frame()))
+    if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))
         return AVERROR(ENOMEM);
+    else
+        avcodec_get_frame_defaults(ist->decoded_frame);
+    decoded_frame = ist->decoded_frame;
     pkt->pts  = *pkt_pts;
     pkt->dts  = ist->pts;
     *pkt_pts  = AV_NOPTS_VALUE;
@@ -1815,12 +1823,11 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
     ret = avcodec_decode_video2(ist->st->codec,
                                 decoded_frame, got_output, pkt);
     if (ret < 0)
-        goto fail;
+        return ret;
 
     quality = same_quant ? decoded_frame->quality : 0;
     if (!*got_output) {
         /* no picture yet */
-        av_freep(&decoded_frame);
         return ret;
     }
     ist->next_pts = ist->pts = decoded_frame->best_effort_timestamp;
@@ -1852,10 +1859,12 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
             decoded_frame->pts = ist->pts;
 
             av_vsrc_buffer_add_frame(ost->input_video_filter, decoded_frame, AV_VSRC_BUF_FLAG_OVERWRITE);
-            if (!(filtered_frame = avcodec_alloc_frame())) {
-                ret = AVERROR(ENOMEM);
-                goto fail;
-            }
+            if (!ist->filtered_frame && !(ist->filtered_frame = avcodec_alloc_frame())) {
+                av_free(buffer_to_free);
+                return AVERROR(ENOMEM);
+            } else
+                avcodec_get_frame_defaults(ist->filtered_frame);
+            filtered_frame = ist->filtered_frame;
             frame_available = avfilter_poll_frame(ost->output_video_filter->inputs[0]);
         }
         while (frame_available) {
@@ -1884,13 +1893,10 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
             if (ost->picref)
                 avfilter_unref_buffer(ost->picref);
         }
-        av_freep(&filtered_frame);
 #endif
     }
 
-fail:
     av_free(buffer_to_free);
-    av_freep(&decoded_frame);
     return ret;
 }
 

cmdutils.h

@@ -196,7 +196,7 @@ void parse_loglevel(int argc, char **argv, const OptionDef *options);
  *
  * @param s  Corresponding format context.
  * @param st Stream from s to be checked.
- * @param spec A stream specifier of the [v|a|s|d]:[<stream index>] form.
+ * @param spec A stream specifier of the [v|a|s|d]:[\<stream index\>] form.
  *
  * @return 1 if the stream matches, 0 if it doesn't, <0 on error
  */

doc/developer.texi

@@ -77,6 +77,11 @@ Use the JavaDoc/Doxygen  format (see examples below) so that code documentation
 can be generated automatically. All nontrivial functions should have a comment
 above them explaining what the function does, even if it is just one sentence.
 All structures and their member variables should be documented, too.
+
+Avoid Qt-style and similar Doxygen syntax with @code{!} in it, i.e. replace
+@code{//!} with @code{///} and similar.  Also @@ syntax should be employed
+for markup commands, i.e. use @code{@@param} and not @code{\param}.
+
 @example
 /**
  * @@file

doc/examples/decoding_encoding.c

@@ -33,6 +33,7 @@
 #include "libavutil/opt.h"
 #include "libavcodec/avcodec.h"
 #include "libavutil/mathematics.h"
+#include "libavutil/samplefmt.h"
 
 #define INBUF_SIZE 4096
 #define AUDIO_INBUF_SIZE 20480
@@ -114,11 +115,11 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 {
     AVCodec *codec;
     AVCodecContext *c= NULL;
-    int out_size, len;
+    int len;
     FILE *f, *outfile;
-    uint8_t *outbuf;
     uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
     AVPacket avpkt;
+    AVFrame *decoded_frame = NULL;
 
     av_init_packet(&avpkt);
 
@@ -139,8 +140,6 @@ static void audio_decode_example(const char *outfilename, const char *filename)
         exit(1);
     }
 
-    outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
-
     f = fopen(filename, "rb");
     if (!f) {
         fprintf(stderr, "could not open %s\n", filename);
@@ -157,15 +156,27 @@ static void audio_decode_example(const char *outfilename, const char *filename)
     avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
 
     while (avpkt.size > 0) {
-        out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
-        len = avcodec_decode_audio3(c, (short *)outbuf, &out_size, &avpkt);
+        int got_frame = 0;
+
+        if (!decoded_frame) {
+            if (!(decoded_frame = avcodec_alloc_frame())) {
+                fprintf(stderr, "out of memory\n");
+                exit(1);
+            }
+        } else
+            avcodec_get_frame_defaults(decoded_frame);
+
+        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
         if (len < 0) {
             fprintf(stderr, "Error while decoding\n");
             exit(1);
         }
-        if (out_size > 0) {
+        if (got_frame) {
             /* if a frame has been decoded, output it */
-            fwrite(outbuf, 1, out_size, outfile);
+            int data_size = av_samples_get_buffer_size(NULL, c->channels,
+                                                       decoded_frame->nb_samples,
+                                                       c->sample_fmt, 1);
+            fwrite(decoded_frame->data[0], 1, data_size, outfile);
         }
         avpkt.size -= len;
         avpkt.data += len;
@@ -185,10 +196,10 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
     fclose(outfile);
     fclose(f);
-    free(outbuf);
 
     avcodec_close(c);
     av_free(c);
+    av_free(decoded_frame);
 }
 
 /*

ffmpeg.c

@@ -168,7 +168,6 @@ static uint8_t *audio_buf;
 static uint8_t *audio_out;
 static unsigned int allocated_audio_out_size, allocated_audio_buf_size;
 
-static void *samples;
 static uint8_t *input_tmp= NULL;
 
 #define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass"
@@ -179,6 +178,8 @@ typedef struct InputStream {
     int discard;             /* true if stream data should be discarded */
     int decoding_needed;     /* true if the packets must be decoded in 'raw_fifo' */
     AVCodec *dec;
+    AVFrame *decoded_frame;
+    AVFrame *filtered_frame;
 
     int64_t       start;     /* time when read started */
     int64_t       next_pts;  /* synthetic pts for cases where pkt.pts
@@ -658,8 +659,11 @@ void av_noreturn exit_program(int ret)
     for(i=0;i<nb_input_files;i++) {
         av_close_input_file(input_files[i].ctx);
     }
-    for (i = 0; i < nb_input_streams; i++)
+    for (i = 0; i < nb_input_streams; i++) {
+        av_freep(&input_streams[i].decoded_frame);
+        av_freep(&input_streams[i].filtered_frame);
         av_dict_free(&input_streams[i].opts);
+    }
 
     if (vstats_file)
         fclose(vstats_file);
@@ -674,7 +678,6 @@ void av_noreturn exit_program(int ret)
     av_free(audio_buf);
     av_free(audio_out);
     allocated_audio_buf_size= allocated_audio_out_size= 0;
-    av_free(samples);
 
 #if CONFIG_AVFILTER
     avfilter_uninit();
@@ -838,14 +841,11 @@ static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_
     memset(buf, fill_char, size);
 }
 
-static void do_audio_out(AVFormatContext *s,
-                         OutputStream *ost,
-                         InputStream *ist,
-                         unsigned char *buf, int size)
+static void do_audio_out(AVFormatContext *s, OutputStream *ost,
+                         InputStream *ist, AVFrame *decoded_frame)
 {
     uint8_t *buftmp;
     int64_t audio_out_size, audio_buf_size;
-    int64_t allocated_for_size= size;
 
     int size_out, frame_bytes, ret, resample_changed;
     AVCodecContext *enc= ost->st->codec;
@@ -853,6 +853,9 @@ static void do_audio_out(AVFormatContext *s,
     int osize = av_get_bytes_per_sample(enc->sample_fmt);
     int isize = av_get_bytes_per_sample(dec->sample_fmt);
     const int coded_bps = av_get_bits_per_sample(enc->codec->id);
+    uint8_t *buf = decoded_frame->data[0];
+    int size     = decoded_frame->nb_samples * dec->channels * isize;
+    int64_t allocated_for_size = size;
 
 need_realloc:
     audio_buf_size= (allocated_for_size + isize*dec->channels - 1) / (isize*dec->channels);
@@ -1732,39 +1735,42 @@ static void rate_emu_sleep(InputStream *ist)
 
 static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
 {
-    static unsigned int samples_size = 0;
+    AVFrame *decoded_frame;
+    AVCodecContext *avctx = ist->st->codec;
     int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
-    uint8_t *decoded_data_buf  = NULL;
-    int      decoded_data_size = 0;
     int i, ret;
 
-    if (pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) {
-        av_free(samples);
-        samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE);
-        samples      = av_malloc(samples_size);
-    }
-    decoded_data_size = samples_size;
+    if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))
+        return AVERROR(ENOMEM);
+    else
+        avcodec_get_frame_defaults(ist->decoded_frame);
+    decoded_frame = ist->decoded_frame;
 
-    ret = avcodec_decode_audio3(ist->st->codec, samples, &decoded_data_size,
-                                pkt);
-    if (ret < 0)
+    ret = avcodec_decode_audio4(avctx, decoded_frame, got_output, pkt);
+    if (ret < 0) {
         return ret;
-    *got_output  = decoded_data_size > 0;
+    }
 
-    /* Some bug in mpeg audio decoder gives */
-    /* decoded_data_size < 0, it seems they are overflows */
     if (!*got_output) {
         /* no audio frame */
         return ret;
     }
 
-    decoded_data_buf = (uint8_t *)samples;
-    ist->next_pts   += ((int64_t)AV_TIME_BASE/bps * decoded_data_size) /
-                       (ist->st->codec->sample_rate * ist->st->codec->channels);
+    /* if the decoder provides a pts, use it instead of the last packet pts.
+       the decoder could be delaying output by a packet or more. */
+    if (decoded_frame->pts != AV_NOPTS_VALUE)
+        ist->next_pts = decoded_frame->pts;
+
+    /* increment next_pts to use for the case where the input stream does not
+       have timestamps or there are multiple frames in the packet */
+    ist->next_pts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) /
+                     avctx->sample_rate;
 
     // preprocess audio (volume)
     if (audio_volume != 256) {
-        switch (ist->st->codec->sample_fmt) {
+        int decoded_data_size = decoded_frame->nb_samples * avctx->channels * bps;
+        void *samples = decoded_frame->data[0];
+        switch (avctx->sample_fmt) {
         case AV_SAMPLE_FMT_U8:
         {
             uint8_t *volp = samples;
@@ -1825,9 +1831,9 @@ static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
 
         if (!check_output_constraints(ist, ost) || !ost->encoding_needed)
             continue;
-        do_audio_out(output_files[ost->file_index].ctx, ost, ist,
-                     decoded_data_buf, decoded_data_size);
+        do_audio_out(output_files[ost->file_index].ctx, ost, ist, decoded_frame);
     }
+
     return ret;
 }
 
@@ -1844,8 +1850,11 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
     int64_t *best_effort_timestamp;
     AVRational *frame_sample_aspect;
 
-    if (!(decoded_frame = avcodec_alloc_frame()))
+    if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))
         return AVERROR(ENOMEM);
+    else
+        avcodec_get_frame_defaults(ist->decoded_frame);
+    decoded_frame = ist->decoded_frame;
     pkt->pts  = *pkt_pts;
     pkt->dts  = *pkt_dts;
     *pkt_pts  = AV_NOPTS_VALUE;
@@ -1867,12 +1876,11 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
     ret = avcodec_decode_video2(ist->st->codec,
                                 decoded_frame, got_output, pkt);
     if (ret < 0)
-        goto fail;
+        return ret;
 
     quality = same_quant ? decoded_frame->quality : 0;
     if (!*got_output) {
         /* no picture yet */
-        av_freep(&decoded_frame);
         return ret;
     }
 
@@ -1917,10 +1925,12 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
                 AVRational ist_pts_tb = ost->output_video_filter->inputs[0]->time_base;
                 if (av_buffersink_get_buffer_ref(ost->output_video_filter, &ost->picref, 0) < 0)
                     goto cont;
-                if (!filtered_frame && !(filtered_frame = avcodec_alloc_frame())) {
-                    ret = AVERROR(ENOMEM);
-                    goto fail;
-                }
+                if (!ist->filtered_frame && !(ist->filtered_frame = avcodec_alloc_frame())) {
+                    av_free(buffer_to_free);
+                    return AVERROR(ENOMEM);
+                } else
+                    avcodec_get_frame_defaults(ist->filtered_frame);
+                filtered_frame = ist->filtered_frame;
                 *filtered_frame= *decoded_frame; //for me_threshold
                 if (ost->picref) {
                     avfilter_fill_frame_from_video_buffer_ref(filtered_frame, ost->picref);
@@ -1942,13 +1952,10 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
             frame_available = ost->output_video_filter && avfilter_poll_frame(ost->output_video_filter->inputs[0]);
             avfilter_unref_buffer(ost->picref);
         }
-        av_freep(&filtered_frame);
 #endif
     }
 
-fail:
     av_free(buffer_to_free);
-    av_freep(&decoded_frame);
     return ret;
 }
 

ffplay.c

@@ -151,11 +151,10 @@ typedef struct VideoState {
     AVStream *audio_st;
     PacketQueue audioq;
     int audio_hw_buf_size;
-    /* samples output by the codec. we reserve more space for avsync
-       compensation, resampling and format conversion */
-    DECLARE_ALIGNED(16,uint8_t,audio_buf1)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
     DECLARE_ALIGNED(16,uint8_t,audio_buf2)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
+    uint8_t silence_buf[SDL_AUDIO_BUFFER_SIZE];
     uint8_t *audio_buf;
+    uint8_t *audio_buf1;
     unsigned int audio_buf_size; /* in bytes */
     int audio_buf_index; /* in bytes */
     int audio_write_buf_size;
@@ -174,6 +173,7 @@ typedef struct VideoState {
     double audio_current_pts_drift;
     int frame_drops_early;
     int frame_drops_late;
+    AVFrame *frame;
 
     enum ShowMode {
         SHOW_MODE_NONE = -1, SHOW_MODE_VIDEO = 0, SHOW_MODE_WAVES, SHOW_MODE_RDFT, SHOW_MODE_NB
@@ -1998,8 +1998,8 @@ static int synchronize_audio(VideoState *is, short *samples,
                     max_size = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n;
                     if (wanted_size < min_size)
                         wanted_size = min_size;
-                    else if (wanted_size > FFMIN3(max_size, sizeof(is->audio_buf1), sizeof(is->audio_buf2)))
-                        wanted_size = FFMIN3(max_size, sizeof(is->audio_buf1), sizeof(is->audio_buf2));
+                    else if (wanted_size > FFMIN3(max_size, samples_size, sizeof(is->audio_buf2)))
+                        wanted_size = FFMIN3(max_size, samples_size, sizeof(is->audio_buf2));
 
                     /* add or remove samples to correction the synchro */
                     if (wanted_size < samples_size) {
@@ -2043,7 +2043,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
     AVPacket *pkt = &is->audio_pkt;
     AVCodecContext *dec= is->audio_st->codec;
     int len1, len2, data_size, resampled_data_size;
-    int64_t dec_channel_layout;
+    int64_t dec_channel_layout, got_frame;
     double pts;
     int new_packet = 0;
     int flush_complete = 0;
@@ -2051,13 +2051,16 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
     for(;;) {
         /* NOTE: the audio packet can contain several frames */
         while (pkt_temp->size > 0 || (!pkt_temp->data && new_packet)) {
+            if (!is->frame) {
+                if (!(is->frame = avcodec_alloc_frame()))
+                    return AVERROR(ENOMEM);
+            } else
+                avcodec_get_frame_defaults(is->frame);
+
             if (flush_complete)
                 break;
             new_packet = 0;
-            data_size = sizeof(is->audio_buf1);
-            len1 = avcodec_decode_audio3(dec,
-                                        (int16_t *)is->audio_buf1, &data_size,
-                                        pkt_temp);
+            len1 = avcodec_decode_audio4(dec, is->frame, &got_frame, pkt_temp);
             if (len1 < 0) {
                 /* if error, we skip the frame */
                 pkt_temp->size = 0;
@@ -2067,12 +2070,15 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
             pkt_temp->data += len1;
             pkt_temp->size -= len1;
 
-            if (data_size <= 0) {
+            if (!got_frame) {
                 /* stop sending empty packets if the decoder is finished */
                 if (!pkt_temp->data && dec->codec->capabilities & CODEC_CAP_DELAY)
                     flush_complete = 1;
                 continue;
             }
+            data_size = av_samples_get_buffer_size(NULL, dec->channels,
+                                                   is->frame->nb_samples,
+                                                   dec->sample_fmt, 1);
 
             dec_channel_layout = (dec->channel_layout && dec->channels == av_get_channel_layout_nb_channels(dec->channel_layout)) ? dec->channel_layout : av_get_default_channel_layout(dec->channels);
 
@@ -2101,7 +2107,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
 
             resampled_data_size = data_size;
             if (is->swr_ctx) {
-                const uint8_t *in[] = {is->audio_buf1};
+                const uint8_t *in[] = { is->frame->data[0] };
                 uint8_t *out[] = {is->audio_buf2};
                 len2 = swr_convert(is->swr_ctx, out, sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt),
                                                 in, data_size / dec->channels / av_get_bytes_per_sample(dec->sample_fmt));
@@ -2116,7 +2122,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
                 is->audio_buf = is->audio_buf2;
                 resampled_data_size = len2 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
             } else {
-                is->audio_buf= is->audio_buf1;
+                is->audio_buf = is->frame->data[0];
             }
 
             /* if no pts, then compute it */
@@ -2150,11 +2156,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
         if (pkt->data == flush_pkt.data)
             avcodec_flush_buffers(dec);
 
-        pkt_temp->data = pkt->data;
-        pkt_temp->size = pkt->size;
-        pkt_temp->flags           = pkt->flags;
-        pkt_temp->side_data       = pkt->side_data;
-        pkt_temp->side_data_elems = pkt->side_data_elems;
+        *pkt_temp = *pkt;
 
         /* if update the audio clock with the pts */
         if (pkt->pts != AV_NOPTS_VALUE) {
@@ -2178,9 +2180,8 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
            audio_size = audio_decode_frame(is, &pts);
            if (audio_size < 0) {
                 /* if error, just output silence */
-               is->audio_buf = is->audio_buf1;
-               is->audio_buf_size = 256 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
-               memset(is->audio_buf, 0, is->audio_buf_size);
+               is->audio_buf      = is->silence_buf;
+               is->audio_buf_size = sizeof(is->silence_buf);
            } else {
                if (is->show_mode != SHOW_MODE_VIDEO)
                    update_sample_display(is, (int16_t *)is->audio_buf, audio_size);
@@ -2356,6 +2357,9 @@ static void stream_component_close(VideoState *is, int stream_index)
         if (is->swr_ctx)
             swr_free(&is->swr_ctx);
         av_free_packet(&is->audio_pkt);
+        av_freep(&is->audio_buf1);
+        is->audio_buf = NULL;
+        av_freep(&is->frame);
 
         if (is->rdft) {
             av_rdft_end(is->rdft);

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;

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
  */
 

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
  */
 

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
 

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,
 };

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);

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;

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)

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];

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),

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);}

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:

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, \

libavdevice/pulse.c

@@ -23,7 +23,6 @@
  * @file
  * PulseAudio input using the simple API.
  * @author Luca Barbato <lu_zero@gentoo.org>
- *
  */
 
 #include <pulse/simple.h>

libavdevice/x11grab.c

@@ -31,8 +31,9 @@
 
 /**
  * @file
- * X11 frame device demuxer by Clemens Fruhwirth <clemens@endorphin.org>
- * and Edouard Gomez <ed.gomez@free.fr>.
+ * X11 frame device demuxer
+ * @author Clemens Fruhwirth <clemens@endorphin.org>
+ * @author Edouard Gomez <ed.gomez@free.fr>
  */
 
 #include "config.h"

libavformat/avformat.h

@@ -293,7 +293,7 @@ typedef struct AVFormatParameters {
 #endif
 } AVFormatParameters;
 
-//! Demuxer will use avio_open, no opened file should be provided by the caller.
+/// Demuxer will use avio_open, no opened file should be provided by the caller.
 #define AVFMT_NOFILE        0x0001
 #define AVFMT_NEEDNUMBER    0x0002 /**< Needs '%d' in filename. */
 #define AVFMT_SHOW_IDS      0x0008 /**< Show format stream IDs numbers. */

libavformat/matroskadec.c

@@ -22,10 +22,10 @@
 /**
  * @file
  * Matroska file demuxer
- * by Ronald Bultje <rbultje@ronald.bitfreak.net>
- * with a little help from Moritz Bunkus <moritz@bunkus.org>
- * totally reworked by Aurelien Jacobs <aurel@gnuage.org>
- * Specs available on the Matroska project page: http://www.matroska.org/.
+ * @author Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * @author with a little help from Moritz Bunkus <moritz@bunkus.org>
+ * @author totally reworked by Aurelien Jacobs <aurel@gnuage.org>
+ * @see specs available on the Matroska project page: http://www.matroska.org/
  */
 
 #include <stdio.h>

libavformat/mov.c

@@ -957,6 +957,8 @@ static int mov_read_stco(MOVContext *c, AVIOContext *pb, MOVAtom atom)
 
     entries = avio_rb32(pb);
 
+    if (!entries)
+        return 0;
     if (entries >= UINT_MAX/sizeof(int64_t))
         return -1;
 
@@ -1398,6 +1400,8 @@ static int mov_read_stsc(MOVContext *c, AVIOContext *pb, MOVAtom atom)
 
     av_dlog(c->fc, "track[%i].stsc.entries = %i\n", c->fc->nb_streams-1, entries);
 
+    if (!entries)
+        return 0;
     if (entries >= UINT_MAX / sizeof(*sc->stsc_data))
         return -1;
     sc->stsc_data = av_malloc(entries * sizeof(*sc->stsc_data));
@@ -1513,6 +1517,8 @@ static int mov_read_stsz(MOVContext *c, AVIOContext *pb, MOVAtom atom)
         return -1;
     }
 
+    if (!entries)
+        return 0;
     if (entries >= UINT_MAX / sizeof(int) || entries >= (UINT_MAX - 4) / field_size)
         return -1;
     sc->sample_sizes = av_malloc(entries * sizeof(int));
@@ -1615,6 +1621,8 @@ static int mov_read_ctts(MOVContext *c, AVIOContext *pb, MOVAtom atom)
 
     av_dlog(c->fc, "track[%i].ctts.entries = %i\n", c->fc->nb_streams-1, entries);
 
+    if (!entries)
+        return 0;
     if (entries >= UINT_MAX / sizeof(*sc->ctts_data))
         return -1;
     sc->ctts_data = av_malloc(entries * sizeof(*sc->ctts_data));
@@ -1675,6 +1683,8 @@ static void mov_build_index(MOVContext *mov, AVStream *st)
 
         current_dts -= sc->dts_shift;
 
+        if (!sc->sample_count)
+            return;
         if (sc->sample_count >= UINT_MAX / sizeof(*st->index_entries))
             return;
         st->index_entries = av_malloc(sc->sample_count*sizeof(*st->index_entries));

libavformat/nuv.c

@@ -47,7 +47,7 @@ static int nuv_probe(AVProbeData *p) {
     return 0;
 }
 
-//! little macro to sanitize packet size
+/// little macro to sanitize packet size
 #define PKTSIZE(s) (s &  0xffffff)
 
 /**

libavformat/oggdec.c

@@ -2,10 +2,9 @@
  * Ogg bitstream support
  * Luca Barbato <lu_zero@gentoo.org>
  * Based on tcvp implementation
- *
  */
 
-/**
+/*
     Copyright (C) 2005  Michael Ahlberg, Måns Rullgård
 
     Permission is hereby granted, free of charge, to any person
@@ -27,7 +26,7 @@
     WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
     OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
     DEALINGS IN THE SOFTWARE.
-**/
+ */
 
 #include <stdio.h>
 #include "oggdec.h"

libavformat/utils.c

@@ -2209,10 +2209,10 @@ static int has_decode_delay_been_guessed(AVStream *st)
 
 static int try_decode_frame(AVStream *st, AVPacket *avpkt, AVDictionary **options)
 {
-    int16_t *samples;
     AVCodec *codec;
-    int got_picture, data_size, ret=0;
+    int got_picture, ret = 0;
     AVFrame picture;
+    AVPacket pkt = *avpkt;
 
     if(!st->codec->codec){
         codec = avcodec_find_decoder(st->codec->codec_id);
@@ -2223,28 +2223,29 @@ static int try_decode_frame(AVStream *st, AVPacket *avpkt, AVDictionary **option
             return ret;
     }
 
-    if(!has_codec_parameters(st->codec) || !has_decode_delay_been_guessed(st) ||
-       (!st->codec_info_nb_frames && st->codec->codec->capabilities & CODEC_CAP_CHANNEL_CONF)) {
+    while (pkt.size > 0 && ret >= 0 &&
+           (!has_codec_parameters(st->codec)  ||
+           !has_decode_delay_been_guessed(st) ||
+           (!st->codec_info_nb_frames && st->codec->codec->capabilities & CODEC_CAP_CHANNEL_CONF))) {
+        got_picture = 0;
+        avcodec_get_frame_defaults(&picture);
         switch(st->codec->codec_type) {
         case AVMEDIA_TYPE_VIDEO:
-            avcodec_get_frame_defaults(&picture);
             ret = avcodec_decode_video2(st->codec, &picture,
-                                        &got_picture, avpkt);
-            if (got_picture)
-                st->info->nb_decoded_frames++;
+                                        &got_picture, &pkt);
             break;
         case AVMEDIA_TYPE_AUDIO:
-            data_size = FFMAX(avpkt->size, AVCODEC_MAX_AUDIO_FRAME_SIZE);
-            samples = av_malloc(data_size);
-            if (!samples)
-                goto fail;
-            ret = avcodec_decode_audio3(st->codec, samples,
-                                        &data_size, avpkt);
-            av_free(samples);
+            ret = avcodec_decode_audio4(st->codec, &picture, &got_picture, &pkt);
             break;
         default:
             break;
         }
+        if (ret >= 0) {
+            if (got_picture)
+                st->info->nb_decoded_frames++;
+            pkt.data += ret;
+            pkt.size -= ret;
+        }
     }
  fail:
     return ret;

libavutil/lzo.c

@@ -21,14 +21,14 @@
 
 #include "avutil.h"
 #include "common.h"
-//! Avoid e.g. MPlayers fast_memcpy, it slows things down here.
+/// Avoid e.g. MPlayers fast_memcpy, it slows things down here.
 #undef memcpy
 #include <string.h>
 #include "lzo.h"
 
-//! Define if we may write up to 12 bytes beyond the output buffer.
+/// Define if we may write up to 12 bytes beyond the output buffer.
 #define OUTBUF_PADDED 1
-//! Define if we may read up to 8 bytes beyond the input buffer.
+/// Define if we may read up to 8 bytes beyond the input buffer.
 #define INBUF_PADDED 1
 typedef struct LZOContext {
     const uint8_t *in, *in_end;

libavutil/lzo.h

@@ -33,13 +33,13 @@
 
 /** @name Error flags returned by av_lzo1x_decode
   * \{ */
-//! end of the input buffer reached before decoding finished
+/// end of the input buffer reached before decoding finished
 #define AV_LZO_INPUT_DEPLETED 1
-//! decoded data did not fit into output buffer
+/// decoded data did not fit into output buffer
 #define AV_LZO_OUTPUT_FULL 2
-//! a reference to previously decoded data was wrong
+/// a reference to previously decoded data was wrong
 #define AV_LZO_INVALID_BACKPTR 4
-//! a non-specific error in the compressed bitstream
+/// a non-specific error in the compressed bitstream
 #define AV_LZO_ERROR 8
 /** \} */
 

libpostproc/postprocess.c

@@ -644,7 +644,7 @@ static inline void postProcess(const uint8_t src[], int srcStride, uint8_t dst[]
 #endif
             postProcess_C(src, srcStride, dst, dstStride, width, height, QPs, QPStride, isColor, c);
 #endif
-#else //CONFIG_RUNTIME_CPUDETECT
+#else /* CONFIG_RUNTIME_CPUDETECT */
 #if   HAVE_MMX2
             postProcess_MMX2(src, srcStride, dst, dstStride, width, height, QPs, QPStride, isColor, c);
 #elif HAVE_AMD3DNOW
@@ -656,7 +656,7 @@ static inline void postProcess(const uint8_t src[], int srcStride, uint8_t dst[]
 #else
             postProcess_C(src, srcStride, dst, dstStride, width, height, QPs, QPStride, isColor, c);
 #endif
-#endif //!CONFIG_RUNTIME_CPUDETECT
+#endif /* !CONFIG_RUNTIME_CPUDETECT */
 }
 
 //static void postProcess(uint8_t src[], int srcStride, uint8_t dst[], int dstStride, int width, int height,

tests/codec-regression.sh

@@ -352,6 +352,11 @@ do_audio_encoding g723_1.tco "-b:a 6.3k -ac 1 -ar 8000 -acodec g723_1"
 do_audio_decoding
 fi
 
+if [ -n "$do_g722" ] ; then
+do_audio_encoding g722.wav "-b 64k -ac 1 -ar 16000 -acodec g722"
+do_audio_decoding
+fi
+
 if [ -n "$do_g726" ] ; then
 do_audio_encoding g726.wav "-b:a 32k -ac 1 -ar 8000 -acodec g726"
 do_audio_decoding

tests/fate/h264.mak

@@ -194,7 +194,7 @@ fate-h264-conformance-aud_mw_e: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-confo
 fate-h264-conformance-ba1_ft_c: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/BA1_FT_C.264
 fate-h264-conformance-ba1_sony_d: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/BA1_Sony_D.jsv
 fate-h264-conformance-ba2_sony_f: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/BA2_Sony_F.jsv
-fate-h264-conformance-ba3_sva_c: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/BA3_SVA_C.264
+fate-h264-conformance-ba3_sva_c: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/BA3_SVA_C.264
 fate-h264-conformance-ba_mw_d: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/BA_MW_D.264
 fate-h264-conformance-bamq1_jvc_c: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/BAMQ1_JVC_C.264
 fate-h264-conformance-bamq2_jvc_c: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/BAMQ2_JVC_C.264
@@ -204,80 +204,80 @@ fate-h264-conformance-caba1_sony_d: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-c
 fate-h264-conformance-caba1_sva_b: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CABA1_SVA_B.264
 fate-h264-conformance-caba2_sony_e: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CABA2_Sony_E.jsv
 fate-h264-conformance-caba2_sva_b: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CABA2_SVA_B.264
-fate-h264-conformance-caba3_sony_c: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CABA3_Sony_C.jsv
-fate-h264-conformance-caba3_sva_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CABA3_SVA_B.264
+fate-h264-conformance-caba3_sony_c: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CABA3_Sony_C.jsv
+fate-h264-conformance-caba3_sva_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CABA3_SVA_B.264
 fate-h264-conformance-caba3_toshiba_e: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CABA3_TOSHIBA_E.264
-fate-h264-conformance-cabac_mot_fld0_full: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/camp_mot_fld0_full.26l
-fate-h264-conformance-cabac_mot_frm0_full: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/camp_mot_frm0_full.26l
-fate-h264-conformance-cabac_mot_mbaff0_full: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/camp_mot_mbaff0_full.26l
-fate-h264-conformance-cabac_mot_picaff0_full: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/camp_mot_picaff0_full.26l
-fate-h264-conformance-cabaci3_sony_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CABACI3_Sony_B.jsv
-fate-h264-conformance-cabast3_sony_e: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CABAST3_Sony_E.jsv
-fate-h264-conformance-cabastbr3_sony_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CABASTBR3_Sony_B.jsv
-fate-h264-conformance-cabref3_sand_d: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CABREF3_Sand_D.264
-fate-h264-conformance-cacqp3_sony_d: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CACQP3_Sony_D.jsv
-fate-h264-conformance-cafi1_sva_c: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAFI1_SVA_C.264
+fate-h264-conformance-cabac_mot_fld0_full: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/camp_mot_fld0_full.26l
+fate-h264-conformance-cabac_mot_frm0_full: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/camp_mot_frm0_full.26l
+fate-h264-conformance-cabac_mot_mbaff0_full: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/camp_mot_mbaff0_full.26l
+fate-h264-conformance-cabac_mot_picaff0_full: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/camp_mot_picaff0_full.26l
+fate-h264-conformance-cabaci3_sony_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CABACI3_Sony_B.jsv
+fate-h264-conformance-cabast3_sony_e: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CABAST3_Sony_E.jsv
+fate-h264-conformance-cabastbr3_sony_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CABASTBR3_Sony_B.jsv
+fate-h264-conformance-cabref3_sand_d: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CABREF3_Sand_D.264
+fate-h264-conformance-cacqp3_sony_d: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CACQP3_Sony_D.jsv
+fate-h264-conformance-cafi1_sva_c: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAFI1_SVA_C.264
 fate-h264-conformance-cama1_sony_c: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CAMA1_Sony_C.jsv
-fate-h264-conformance-cama1_toshiba_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAMA1_TOSHIBA_B.264
-fate-h264-conformance-cama1_vtc_c: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/cama1_vtc_c.avc
+fate-h264-conformance-cama1_toshiba_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAMA1_TOSHIBA_B.264
+fate-h264-conformance-cama1_vtc_c: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/cama1_vtc_c.avc
 fate-h264-conformance-cama2_vtc_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/cama2_vtc_b.avc
-fate-h264-conformance-cama3_sand_e: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAMA3_Sand_E.264
-fate-h264-conformance-cama3_vtc_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/cama3_vtc_b.avc
-fate-h264-conformance-camaci3_sony_c: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAMACI3_Sony_C.jsv
-fate-h264-conformance-camanl1_toshiba_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAMANL1_TOSHIBA_B.264
-fate-h264-conformance-camanl2_toshiba_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAMANL2_TOSHIBA_B.264
-fate-h264-conformance-camanl3_sand_e: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAMANL3_Sand_E.264
-fate-h264-conformance-camasl3_sony_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAMASL3_Sony_B.jsv
-fate-h264-conformance-camp_mot_mbaff_l30: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAMP_MOT_MBAFF_L30.26l
-fate-h264-conformance-camp_mot_mbaff_l31: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAMP_MOT_MBAFF_L31.26l
+fate-h264-conformance-cama3_sand_e: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAMA3_Sand_E.264
+fate-h264-conformance-cama3_vtc_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/cama3_vtc_b.avc
+fate-h264-conformance-camaci3_sony_c: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAMACI3_Sony_C.jsv
+fate-h264-conformance-camanl1_toshiba_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAMANL1_TOSHIBA_B.264
+fate-h264-conformance-camanl2_toshiba_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAMANL2_TOSHIBA_B.264
+fate-h264-conformance-camanl3_sand_e: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAMANL3_Sand_E.264
+fate-h264-conformance-camasl3_sony_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAMASL3_Sony_B.jsv
+fate-h264-conformance-camp_mot_mbaff_l30: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAMP_MOT_MBAFF_L30.26l
+fate-h264-conformance-camp_mot_mbaff_l31: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAMP_MOT_MBAFF_L31.26l
 fate-h264-conformance-canl1_sony_e: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CANL1_Sony_E.jsv
 fate-h264-conformance-canl1_sva_b: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CANL1_SVA_B.264
 fate-h264-conformance-canl1_toshiba_g: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CANL1_TOSHIBA_G.264
 fate-h264-conformance-canl2_sony_e: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CANL2_Sony_E.jsv
 fate-h264-conformance-canl2_sva_b: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CANL2_SVA_B.264
-fate-h264-conformance-canl3_sony_c: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CANL3_Sony_C.jsv
+fate-h264-conformance-canl3_sony_c: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CANL3_Sony_C.jsv
 fate-h264-conformance-canl3_sva_b: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CANL3_SVA_B.264
 fate-h264-conformance-canl4_sva_b: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CANL4_SVA_B.264
 fate-h264-conformance-canlma2_sony_c: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CANLMA2_Sony_C.jsv
 fate-h264-conformance-canlma3_sony_c: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CANLMA3_Sony_C.jsv
-fate-h264-conformance-capa1_toshiba_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAPA1_TOSHIBA_B.264
-fate-h264-conformance-capama3_sand_f: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAPAMA3_Sand_F.264
+fate-h264-conformance-capa1_toshiba_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAPA1_TOSHIBA_B.264
+fate-h264-conformance-capama3_sand_f: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAPAMA3_Sand_F.264
 fate-h264-conformance-capcm1_sand_e: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CAPCM1_Sand_E.264
 fate-h264-conformance-capcmnl1_sand_e: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CAPCMNL1_Sand_E.264
-fate-h264-conformance-capm3_sony_d: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAPM3_Sony_D.jsv
+fate-h264-conformance-capm3_sony_d: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAPM3_Sony_D.jsv
 fate-h264-conformance-caqp1_sony_b: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CAQP1_Sony_B.jsv
-fate-h264-conformance-cavlc_mot_fld0_full_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/cvmp_mot_fld0_full_B.26l
-fate-h264-conformance-cavlc_mot_frm0_full_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/cvmp_mot_frm0_full_B.26l
-fate-h264-conformance-cavlc_mot_mbaff0_full_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/cvmp_mot_mbaff0_full_B.26l
-fate-h264-conformance-cavlc_mot_picaff0_full_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/cvmp_mot_picaff0_full_B.26l
+fate-h264-conformance-cavlc_mot_fld0_full_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/cvmp_mot_fld0_full_B.26l
+fate-h264-conformance-cavlc_mot_frm0_full_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/cvmp_mot_frm0_full_B.26l
+fate-h264-conformance-cavlc_mot_mbaff0_full_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/cvmp_mot_mbaff0_full_B.26l
+fate-h264-conformance-cavlc_mot_picaff0_full_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/cvmp_mot_picaff0_full_B.26l
 fate-h264-conformance-cawp1_toshiba_e: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CAWP1_TOSHIBA_E.264
-fate-h264-conformance-cawp5_toshiba_e: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CAWP5_TOSHIBA_E.264
+fate-h264-conformance-cawp5_toshiba_e: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CAWP5_TOSHIBA_E.264
 fate-h264-conformance-ci1_ft_b: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CI1_FT_B.264
 fate-h264-conformance-ci_mw_d: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CI_MW_D.264
-fate-h264-conformance-cvbs3_sony_c: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVBS3_Sony_C.jsv
+fate-h264-conformance-cvbs3_sony_c: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVBS3_Sony_C.jsv
 fate-h264-conformance-cvcanlma2_sony_c: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CVCANLMA2_Sony_C.jsv
-fate-h264-conformance-cvfi1_sony_d: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVFI1_Sony_D.jsv
-fate-h264-conformance-cvfi1_sva_c: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVFI1_SVA_C.264
-fate-h264-conformance-cvfi2_sony_h: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVFI2_Sony_H.jsv
-fate-h264-conformance-cvfi2_sva_c: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVFI2_SVA_C.264
+fate-h264-conformance-cvfi1_sony_d: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVFI1_Sony_D.jsv
+fate-h264-conformance-cvfi1_sva_c: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVFI1_SVA_C.264
+fate-h264-conformance-cvfi2_sony_h: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVFI2_Sony_H.jsv
+fate-h264-conformance-cvfi2_sva_c: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVFI2_SVA_C.264
 fate-h264-conformance-cvma1_sony_d: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CVMA1_Sony_D.jsv
-fate-h264-conformance-cvma1_toshiba_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVMA1_TOSHIBA_B.264
-fate-h264-conformance-cvmanl1_toshiba_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVMANL1_TOSHIBA_B.264
-fate-h264-conformance-cvmanl2_toshiba_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVMANL2_TOSHIBA_B.264
-fate-h264-conformance-cvmapaqp3_sony_e: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVMAPAQP3_Sony_E.jsv
-fate-h264-conformance-cvmaqp2_sony_g: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVMAQP2_Sony_G.jsv
-fate-h264-conformance-cvmaqp3_sony_d: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVMAQP3_Sony_D.jsv
-fate-h264-conformance-cvmp_mot_fld_l30_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVMP_MOT_FLD_L30_B.26l
-fate-h264-conformance-cvmp_mot_frm_l31_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVMP_MOT_FRM_L31_B.26l
+fate-h264-conformance-cvma1_toshiba_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVMA1_TOSHIBA_B.264
+fate-h264-conformance-cvmanl1_toshiba_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVMANL1_TOSHIBA_B.264
+fate-h264-conformance-cvmanl2_toshiba_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVMANL2_TOSHIBA_B.264
+fate-h264-conformance-cvmapaqp3_sony_e: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVMAPAQP3_Sony_E.jsv
+fate-h264-conformance-cvmaqp2_sony_g: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVMAQP2_Sony_G.jsv
+fate-h264-conformance-cvmaqp3_sony_d: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVMAQP3_Sony_D.jsv
+fate-h264-conformance-cvmp_mot_fld_l30_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVMP_MOT_FLD_L30_B.26l
+fate-h264-conformance-cvmp_mot_frm_l31_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVMP_MOT_FRM_L31_B.26l
 fate-h264-conformance-cvnlfi1_sony_c: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVNLFI1_Sony_C.jsv
-fate-h264-conformance-cvnlfi2_sony_h: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVNLFI2_Sony_H.jsv
-fate-h264-conformance-cvpa1_toshiba_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVPA1_TOSHIBA_B.264
+fate-h264-conformance-cvnlfi2_sony_h: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVNLFI2_Sony_H.jsv
+fate-h264-conformance-cvpa1_toshiba_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVPA1_TOSHIBA_B.264
 fate-h264-conformance-cvpcmnl1_sva_c: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CVPCMNL1_SVA_C.264
 fate-h264-conformance-cvpcmnl2_sva_c: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CVPCMNL2_SVA_C.264
 fate-h264-conformance-cvwp1_toshiba_e: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/CVWP1_TOSHIBA_E.264
-fate-h264-conformance-cvwp2_toshiba_e: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVWP2_TOSHIBA_E.264
-fate-h264-conformance-cvwp3_toshiba_e: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVWP3_TOSHIBA_E.264
-fate-h264-conformance-cvwp5_toshiba_e: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/CVWP5_TOSHIBA_E.264
+fate-h264-conformance-cvwp2_toshiba_e: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVWP2_TOSHIBA_E.264
+fate-h264-conformance-cvwp3_toshiba_e: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVWP3_TOSHIBA_E.264
+fate-h264-conformance-cvwp5_toshiba_e: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/CVWP5_TOSHIBA_E.264
 fate-h264-conformance-fi1_sony_e: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/FI1_Sony_E.jsv
 fate-h264-conformance-frext-alphaconformanceg: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/FRext/test8b43.264
 fate-h264-conformance-frext-bcrm_freh10: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/FRext/freh10.264 -vsync 0
@@ -337,8 +337,8 @@ fate-h264-conformance-frext-pph422i4_panasonic_a: CMD = framecrc -vsync 0 -i $(S
 fate-h264-conformance-frext-pph422i5_panasonic_a: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/FRext/PPH422I5_Panasonic_A.264 -pix_fmt yuv422p10le
 fate-h264-conformance-frext-pph422i6_panasonic_a: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/FRext/PPH422I6_Panasonic_A.264 -pix_fmt yuv422p10le
 fate-h264-conformance-frext-pph422i7_panasonic_a: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/FRext/PPH422I7_Panasonic_A.264 -pix_fmt yuv422p10le
-fate-h264-conformance-hcbp2_hhi_a: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/HCBP2_HHI_A.264
-fate-h264-conformance-hcmp1_hhi_a: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/HCMP1_HHI_A.264
+fate-h264-conformance-hcbp2_hhi_a: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/HCBP2_HHI_A.264
+fate-h264-conformance-hcmp1_hhi_a: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/HCMP1_HHI_A.264
 fate-h264-conformance-ls_sva_d: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/LS_SVA_D.264
 fate-h264-conformance-midr_mw_d: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/MIDR_MW_D.264
 fate-h264-conformance-mps_mw_a: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/MPS_MW_A.264
@@ -349,11 +349,11 @@ fate-h264-conformance-mr2_tandberg_e: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h26
 fate-h264-conformance-mr3_tandberg_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/MR3_TANDBERG_B.264
 fate-h264-conformance-mr4_tandberg_c: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/MR4_TANDBERG_C.264
 fate-h264-conformance-mr5_tandberg_c: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/MR5_TANDBERG_C.264
-fate-h264-conformance-mr6_bt_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/MR6_BT_B.h264
-fate-h264-conformance-mr7_bt_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/MR7_BT_B.h264
-fate-h264-conformance-mr8_bt_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/MR8_BT_B.h264
-fate-h264-conformance-mr9_bt_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/MR9_BT_B.h264
-fate-h264-conformance-mv1_brcm_d: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/src19td.IBP.264
+fate-h264-conformance-mr6_bt_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/MR6_BT_B.h264
+fate-h264-conformance-mr7_bt_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/MR7_BT_B.h264
+fate-h264-conformance-mr8_bt_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/MR8_BT_B.h264
+fate-h264-conformance-mr9_bt_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/MR9_BT_B.h264
+fate-h264-conformance-mv1_brcm_d: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/src19td.IBP.264
 fate-h264-conformance-nl1_sony_d: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/NL1_Sony_D.jsv
 fate-h264-conformance-nl2_sony_h: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/NL2_Sony_H.jsv
 fate-h264-conformance-nl3_sva_e: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/NL3_SVA_E.264
@@ -363,9 +363,9 @@ fate-h264-conformance-nrf_mw_e: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-confo
 fate-h264-conformance-sharp_mp_field_1_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/Sharp_MP_Field_1_B.jvt
 fate-h264-conformance-sharp_mp_field_2_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/Sharp_MP_Field_2_B.jvt
 fate-h264-conformance-sharp_mp_field_3_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/Sharp_MP_Field_3_B.jvt
-fate-h264-conformance-sharp_mp_paff_1r2: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/Sharp_MP_PAFF_1r2.jvt
-fate-h264-conformance-sharp_mp_paff_2r: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/Sharp_MP_PAFF_2.jvt
-fate-h264-conformance-sl1_sva_b: CMD = framecrc  -vsync 0 -strict 1 -i $(SAMPLES)/h264-conformance/SL1_SVA_B.264
+fate-h264-conformance-sharp_mp_paff_1r2: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/Sharp_MP_PAFF_1r2.jvt
+fate-h264-conformance-sharp_mp_paff_2r: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/Sharp_MP_PAFF_2.jvt
+fate-h264-conformance-sl1_sva_b: CMD = framecrc  -vsync 0 -i $(SAMPLES)/h264-conformance/SL1_SVA_B.264
 fate-h264-conformance-sva_ba1_b: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/SVA_BA1_B.264
 fate-h264-conformance-sva_ba2_d: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/SVA_BA2_D.264
 fate-h264-conformance-sva_base_b: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conformance/SVA_Base_B.264
@@ -376,4 +376,4 @@ fate-h264-conformance-sva_nl2_e: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264-conf
 
 fate-h264-interlace-crop: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264/interlaced_crop.mp4 -vframes 3
 fate-h264-lossless: CMD = framecrc -vsync 0 -i $(SAMPLES)/h264/lossless.h264
-fate-h264-extreme-plane-pred: CMD = framemd5 -strict 1 -vsync 0 -i $(SAMPLES)/h264/extreme-plane-pred.h264
+fate-h264-extreme-plane-pred: CMD = framemd5 -vsync 0 -i $(SAMPLES)/h264/extreme-plane-pred.h264

tests/ref/acodec/g722

@@ -0,0 +1,4 @@
+156f63e3391b95020ae882dbae6eccf3 *./tests/data/acodec/g722.wav
+47991 ./tests/data/acodec/g722.wav
+8f65de513acc08b37a488d6a802b4f00 *./tests/data/g722.acodec.out.wav
+stddev: 8860.50 PSNR: 17.38 MAXDIFF:33814 bytes:   191732/  1058400

tools/patcheck

@@ -55,6 +55,7 @@ hiegrep 'INIT_VLC_USE_STATIC' 'forbidden ancient vlc type' $*
 hiegrep '=[-+\*\&] ' 'looks like compound assignment' $*
 hiegrep2 '/\*\* *[a-zA-Z0-9].*' '\*/' 'Inconsistently formatted doxygen comment' $*
 hiegrep '; */\*\*[^<]' 'Misformatted doxygen comment' $*
+hiegrep '//!|/\*!' 'inconsistent doxygen syntax' $*
 
 hiegrep2 '(int|unsigned|static|void)[a-zA-Z0-9 _]*(init|end)[a-zA-Z0-9 _]*\(.*[^;]$' '(av_cold|:\+[^a-zA-Z_])' 'These functions may need av_cold, please review the whole patch for similar functions needing av_cold' $*