Merge remote-tracking branch 'qatar/master'

* qatar/master:
  rtmp: Support 'rtmp_live', an option which specifies if the media is a live stream.
  av_samples_fill_array: Mark unmodified function argument as const.
  lagarith: add YUY2 decoding support
  Support decoding unaligned rgb24 lagarith.
  dv: Split profile handling code into a separate file.
  flvenc: use AVFormatContext, not AVCodecContext for logging.
  mov: Remove write-only variable in mov_read_chan().
  fate: Change the probe-format refs to match the final text format committed.
  fate: Add avprobe as a make dependency
  Add probe fate tests to test for regressions in detecting media types.
  fate: Add oneline comparison method
  qdm2: clip array indices returned by qdm2_get_vlc().
  avplay: properly close/reopen AVAudioResampleContext on channel layout change
  avcodec: do not needlessly set packet size to 0 in avcodec_encode_audio2()
  avcodec: for audio encoding, reset output packet when it is not valid
  avcodec: refactor avcodec_encode_audio2() to merge common branches
  avcodec: remove fallbacks for AVCodec.encode() in avcodec_encode_audio2()

Conflicts:
	ffplay.c
	libavcodec/Makefile
	libavcodec/dvdata.c
	libavcodec/dvdata.h
	libavcodec/qdm2.c
	libavcodec/utils.c
	libavformat/flvenc.c
	libavformat/mov.c
	tests/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>

ffprobe.c

@@ -265,17 +265,21 @@ static inline void writer_print_chapter_footer(WriterContext *wctx,
 static inline void writer_print_section_header(WriterContext *wctx,
                                                const char *section)
 {
-    if (wctx->writer->print_section_header)
-        wctx->writer->print_section_header(wctx, section);
-    wctx->nb_item = 0;
+    if (!fmt_entries_to_show || (section && av_dict_get(fmt_entries_to_show, section, NULL, 0))) {
+        if (wctx->writer->print_section_header)
+            wctx->writer->print_section_header(wctx, section);
+        wctx->nb_item = 0;
+    }
 }
 
 static inline void writer_print_section_footer(WriterContext *wctx,
                                                const char *section)
 {
-    if (wctx->writer->print_section_footer)
-        wctx->writer->print_section_footer(wctx, section);
-    wctx->nb_section++;
+    if (!fmt_entries_to_show || (section && av_dict_get(fmt_entries_to_show, section, NULL, 0))) {
+        if (wctx->writer->print_section_footer)
+            wctx->writer->print_section_footer(wctx, section);
+        wctx->nb_section++;
+    }
 }
 
 static inline void writer_print_integer(WriterContext *wctx,
@@ -448,7 +452,8 @@ static void default_show_tags(WriterContext *wctx, AVDictionary *dict)
 {
     AVDictionaryEntry *tag = NULL;
     while ((tag = av_dict_get(dict, "", tag, AV_DICT_IGNORE_SUFFIX))) {
-        printf("TAG:");
+        if (!fmt_entries_to_show || (tag->key && av_dict_get(fmt_entries_to_show, tag->key, NULL, 0)))
+            printf("TAG:");
         writer_print_string(wctx, tag->key, tag->value, 0);
     }
 }

libavcodec/Makefile

@@ -146,8 +146,8 @@ OBJS-$(CONFIG_DVBSUB_DECODER)          += dvbsubdec.o
 OBJS-$(CONFIG_DVBSUB_ENCODER)          += dvbsub.o
 OBJS-$(CONFIG_DVDSUB_DECODER)          += dvdsubdec.o
 OBJS-$(CONFIG_DVDSUB_ENCODER)          += dvdsubenc.o
-OBJS-$(CONFIG_DVVIDEO_DECODER)         += dvdec.o dv.o dvdata.o
-OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dv.o dvdata.o
+OBJS-$(CONFIG_DVVIDEO_DECODER)         += dvdec.o dv.o dvdata.o dv_profile.o
+OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dv.o dvdata.o dv_profile.o
 OBJS-$(CONFIG_DXA_DECODER)             += dxa.o
 OBJS-$(CONFIG_DXTORY_DECODER)          += dxtory.o
 OBJS-$(CONFIG_EAC3_DECODER)            += eac3dec.o eac3_data.o
@@ -609,8 +609,8 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcmenc.o adpcm_data.o
 OBJS-$(CONFIG_ADTS_MUXER)              += mpeg4audio.o
 OBJS-$(CONFIG_ADX_DEMUXER)             += adx.o
 OBJS-$(CONFIG_CAF_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
-OBJS-$(CONFIG_DV_DEMUXER)              += dvdata.o
-OBJS-$(CONFIG_DV_MUXER)                += dvdata.o timecode.o
+OBJS-$(CONFIG_DV_DEMUXER)              += dv_profile.o
+OBJS-$(CONFIG_DV_MUXER)                += dv_profile.o timecode.o
 OBJS-$(CONFIG_FLAC_DEMUXER)            += flacdec.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLAC_MUXER)              += flacdec.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLV_DEMUXER)             += mpeg4audio.o

libavcodec/dv_profile.h

+/*
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_DV_PROFILE_H
+#define AVCODEC_DV_PROFILE_H
+
+#include <stdint.h>
+
+#include "libavutil/pixfmt.h"
+#include "libavutil/rational.h"
+#include "avcodec.h"
+
+/* minimum number of bytes to read from a DV stream in order to
+   determine the profile */
+#define DV_PROFILE_BYTES (6*80) /* 6 DIF blocks */
+
+typedef struct DVwork_chunk {
+    uint16_t  buf_offset;
+    uint16_t  mb_coordinates[5];
+} DVwork_chunk;
+
+/*
+ * DVprofile is used to express the differences between various
+ * DV flavors. For now it's primarily used for differentiating
+ * 525/60 and 625/50, but the plans are to use it for various
+ * DV specs as well (e.g. SMPTE314M vs. IEC 61834).
+ */
+typedef struct DVprofile {
+    int              dsf;                   /* value of the dsf in the DV header */
+    int              video_stype;           /* stype for VAUX source pack */
+    int              frame_size;            /* total size of one frame in bytes */
+    int              difseg_size;           /* number of DIF segments per DIF channel */
+    int              n_difchan;             /* number of DIF channels per frame */
+    AVRational       time_base;             /* 1/framerate */
+    int              ltc_divisor;           /* FPS from the LTS standpoint */
+    int              height;                /* picture height in pixels */
+    int              width;                 /* picture width in pixels */
+    AVRational       sar[2];                /* sample aspect ratios for 4:3 and 16:9 */
+    DVwork_chunk    *work_chunks;           /* each thread gets its own chunk of frame to work on */
+    uint32_t        *idct_factor;           /* set of iDCT factor tables */
+    enum PixelFormat pix_fmt;               /* picture pixel format */
+    int              bpm;                   /* blocks per macroblock */
+    const uint8_t   *block_sizes;           /* AC block sizes, in bits */
+    int              audio_stride;          /* size of audio_shuffle table */
+    int              audio_min_samples[3];  /* min amount of audio samples */
+                                            /* for 48kHz, 44.1kHz and 32kHz */
+    int              audio_samples_dist[5]; /* how many samples are supposed to be */
+                                            /* in each frame in a 5 frames window */
+    const uint8_t  (*audio_shuffle)[9];     /* PCM shuffling table */
+} DVprofile;
+
+const DVprofile* avpriv_dv_frame_profile(const DVprofile *sys,
+                                         const uint8_t* frame, unsigned buf_size);
+const DVprofile* avpriv_dv_frame_profile2(AVCodecContext* codec, const DVprofile *sys,
+                                  const uint8_t* frame, unsigned buf_size);
+const DVprofile* avpriv_dv_codec_profile(AVCodecContext* codec);
+
+/**
+ *  Print all allowed DV profiles into logctx at specified logging level.
+ */
+void ff_dv_print_profiles(void *logctx, int loglevel);
+
+#endif /* AVCODEC_DV_PROFILE_H */

libavcodec/dvdata.h

@@ -27,45 +27,10 @@
 #ifndef AVCODEC_DVDATA_H
 #define AVCODEC_DVDATA_H
 
-#include "libavutil/rational.h"
 #include "avcodec.h"
 #include "dsputil.h"
 #include "get_bits.h"
-
-typedef struct DVwork_chunk {
-    uint16_t  buf_offset;
-    uint16_t  mb_coordinates[5];
-} DVwork_chunk;
-
-/*
- * DVprofile is used to express the differences between various
- * DV flavors. For now it's primarily used for differentiating
- * 525/60 and 625/50, but the plans are to use it for various
- * DV specs as well (e.g. SMPTE314M vs. IEC 61834).
- */
-typedef struct DVprofile {
-    int              dsf;                   /* value of the dsf in the DV header */
-    int              video_stype;           /* stype for VAUX source pack */
-    int              frame_size;            /* total size of one frame in bytes */
-    int              difseg_size;           /* number of DIF segments per DIF channel */
-    int              n_difchan;             /* number of DIF channels per frame */
-    AVRational       time_base;             /* 1/framerate */
-    int              ltc_divisor;           /* FPS from the LTS standpoint */
-    int              height;                /* picture height in pixels */
-    int              width;                 /* picture width in pixels */
-    AVRational       sar[2];                /* sample aspect ratios for 4:3 and 16:9 */
-    DVwork_chunk    *work_chunks;           /* each thread gets its own chunk of frame to work on */
-    uint32_t        *idct_factor;           /* set of iDCT factor tables */
-    enum PixelFormat pix_fmt;               /* picture pixel format */
-    int              bpm;                   /* blocks per macroblock */
-    const uint8_t   *block_sizes;           /* AC block sizes, in bits */
-    int              audio_stride;          /* size of audio_shuffle table */
-    int              audio_min_samples[3];  /* min amount of audio samples */
-                                            /* for 48kHz, 44.1kHz and 32kHz */
-    int              audio_samples_dist[5]; /* how many samples are supposed to be */
-                                            /* in each frame in a 5 frames window */
-    const uint8_t  (*audio_shuffle)[9];     /* PCM shuffling table */
-} DVprofile;
+#include "dv_profile.h"
 
 typedef struct DVVideoContext {
     const DVprofile *sys;
@@ -118,10 +83,6 @@ extern const int ff_dv_iweight_720_c[64];
 #define DV_PROFILE_IS_1080i50(p) (((p)->video_stype == 0x14) && ((p)->dsf == 1))
 #define DV_PROFILE_IS_720p50(p)  (((p)->video_stype == 0x18) && ((p)->dsf == 1))
 
-/* minimum number of bytes to read from a DV stream in order to
-   determine the profile */
-#define DV_PROFILE_BYTES (6*80) /* 6 DIF blocks */
-
 /**
  * largest possible DV frame, in bytes (1080i50)
  */
@@ -136,12 +97,6 @@ extern const int ff_dv_iweight_720_c[64];
 
 extern RL_VLC_ELEM ff_dv_rl_vlc[1184];
 
-const DVprofile* avpriv_dv_frame_profile(const DVprofile *sys,
-                                  const uint8_t* frame, unsigned buf_size);
-const DVprofile* avpriv_dv_frame_profile2(AVCodecContext* codec, const DVprofile *sys,
-                                  const uint8_t* frame, unsigned buf_size);
-const DVprofile* avpriv_dv_codec_profile(AVCodecContext* codec);
-
 int ff_dv_init_dynamic_tables(const DVprofile *d);
 int ff_dvvideo_init(AVCodecContext *avctx);
 
@@ -166,9 +121,4 @@ static inline void dv_calculate_mb_xy(DVVideoContext *s, DVwork_chunk *work_chun
      }
 }
 
-/**
- *  Print all allowed DV profiles into logctx at specified logging level.
- */
-void ff_dv_print_profiles(void *logctx, int loglevel);
-
 #endif /* AVCODEC_DVDATA_H */

libavcodec/lagarith.c

@@ -269,6 +269,40 @@ static void lag_pred_line(LagarithContext *l, uint8_t *buf,
     }
 }
 
+static void lag_pred_line_yuy2(LagarithContext *l, uint8_t *buf,
+                               int width, int stride, int line,
+                               int is_luma)
+{
+    int L, TL;
+
+    if (!line) {
+        if (is_luma) {
+            buf++;
+            width--;
+        }
+        l->dsp.add_hfyu_left_prediction(buf + 1, buf + 1, width - 1, buf[0]);
+        return;
+    }
+    if (line == 1) {
+        const int HEAD = is_luma ? 4 : 2;
+        int i;
+
+        L  = buf[width - stride - 1];
+        TL = buf[HEAD  - stride - 1];
+        for (i = 0; i < HEAD; i++) {
+            L += buf[i];
+            buf[i] = L;
+        }
+        buf   += HEAD;
+        width -= HEAD;
+    } else {
+        TL = buf[width - (2 * stride) - 1];
+        L  = buf[width - stride - 1];
+    }
+    l->dsp.add_hfyu_median_prediction(buf, buf - stride, buf, width,
+                                      &L, &TL);
+}
+
 static int lag_decode_line(LagarithContext *l, lag_rac *rac,
                            uint8_t *dst, int width, int stride,
                            int esc_count)
@@ -442,9 +476,17 @@ static int lag_decode_arith_plane(LagarithContext *l, uint8_t *dst,
         return -1;
     }
 
-    for (i = 0; i < height; i++) {
-        lag_pred_line(l, dst, width, stride, i);
-        dst += stride;
+    if (l->avctx->pix_fmt != PIX_FMT_YUV422P) {
+        for (i = 0; i < height; i++) {
+            lag_pred_line(l, dst, width, stride, i);
+            dst += stride;
+        }
+    } else {
+        for (i = 0; i < height; i++) {
+            lag_pred_line_yuy2(l, dst, width, stride, i,
+                               width == l->avctx->width);
+            dst += stride;
+        }
     }
 
     return 0;
@@ -566,6 +608,32 @@ static int lag_decode_frame(AVCodecContext *avctx,
                 srcs[i] += l->rgb_stride;
         }
         break;
+    case FRAME_ARITH_YUY2:
+        avctx->pix_fmt = PIX_FMT_YUV422P;
+
+        if (avctx->get_buffer(avctx, p) < 0) {
+            av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+            return -1;
+        }
+
+        if (offset_ry >= buf_size ||
+            offset_gu >= buf_size ||
+            offset_bv >= buf_size) {
+            av_log(avctx, AV_LOG_ERROR,
+                   "Invalid frame offsets\n");
+            return AVERROR_INVALIDDATA;
+        }
+
+        lag_decode_arith_plane(l, p->data[0], avctx->width, avctx->height,
+                               p->linesize[0], buf + offset_ry,
+                               buf_size - offset_ry);
+        lag_decode_arith_plane(l, p->data[2], avctx->width / 2,
+                               avctx->height, p->linesize[2],
+                               buf + offset_gu, buf_size - offset_gu);
+        lag_decode_arith_plane(l, p->data[1], avctx->width / 2,
+                               avctx->height, p->linesize[1],
+                               buf + offset_bv, buf_size - offset_bv);
+        break;
     case FRAME_ARITH_YV12:
         avctx->pix_fmt = PIX_FMT_YUV420P;
 

libavcodec/qdm2.c

@@ -887,10 +887,10 @@ static int synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int le
 
                     case 30:
                         if (get_bits_left(gb) >= 4) {
-                            unsigned v = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
-                            if (v >= FF_ARRAY_ELEMS(type30_dequant))
+                            unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
+                            if (index >= FF_ARRAY_ELEMS(type30_dequant))
                                 return AVERROR_INVALIDDATA;
-                            samples[0] = type30_dequant[v];
+                            samples[0] = type30_dequant[index];
                         } else
                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
 
@@ -905,10 +905,10 @@ static int synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int le
                                 type34_predictor = samples[0];
                                 type34_first = 0;
                             } else {
-                                unsigned v = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
-                                if (v >= FF_ARRAY_ELEMS(type34_delta))
+                                unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
+                                if (index >= FF_ARRAY_ELEMS(type34_delta))
                                     return AVERROR_INVALIDDATA;
-                                samples[0] = type34_delta[v] / type34_div + type34_predictor;
+                                samples[0] = type34_delta[index] / type34_div + type34_predictor;
                                 type34_predictor = samples[0];
                             }
                         } else {

libavcodec/utils.c

@@ -1008,7 +1008,6 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
 {
     int ret;
     AVPacket user_pkt = *avpkt;
-    int nb_samples;
     int needs_realloc = !user_pkt.data;
 
     *got_packet_ptr = 0;
@@ -1016,27 +1015,23 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
     if (!(avctx->codec->capabilities & CODEC_CAP_DELAY) && !frame) {
         av_free_packet(avpkt);
         av_init_packet(avpkt);
-        avpkt->size = 0;
         return 0;
     }
 
     /* check for valid frame size */
     if (frame) {
-        nb_samples = frame->nb_samples;
         if (avctx->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME) {
-            if (nb_samples > avctx->frame_size)
+            if (frame->nb_samples > avctx->frame_size)
                 return AVERROR(EINVAL);
         } else if (!(avctx->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)) {
-            if (nb_samples != avctx->frame_size)
+            if (frame->nb_samples != avctx->frame_size)
                 return AVERROR(EINVAL);
         }
-    } else {
-        nb_samples = avctx->frame_size;
     }
 
-    if (avctx->codec->encode2) {
-        ret = avctx->codec->encode2(avctx, avpkt, frame, got_packet_ptr);
-        if (!ret && *got_packet_ptr) {
+    ret = avctx->codec->encode2(avctx, avpkt, frame, got_packet_ptr);
+    if (!ret) {
+        if (*got_packet_ptr) {
             if (!(avctx->codec->capabilities & CODEC_CAP_DELAY)) {
                 if (avpkt->pts == AV_NOPTS_VALUE)
                     avpkt->pts = frame->pts;
@@ -1048,69 +1043,6 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
         } else {
             avpkt->size = 0;
         }
-    } else {
-        /* for compatibility with encoders not supporting encode2(), we need to
-           allocate a packet buffer if the user has not provided one or check
-           the size otherwise */
-        int fs_tmp   = 0;
-        int buf_size = avpkt->size;
-        if (!user_pkt.data) {
-            if (avctx->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE) {
-                av_assert0(av_get_bits_per_sample(avctx->codec_id) != 0);
-                if (!frame)
-                    return AVERROR(EINVAL);
-                buf_size = nb_samples * avctx->channels *
-                           av_get_bits_per_sample(avctx->codec_id) / 8;
-            } else {
-                /* this is a guess as to the required size.
-                   if an encoder needs more than this, it should probably
-                   implement encode2() */
-                buf_size = 2 * avctx->frame_size * avctx->channels *
-                           av_get_bytes_per_sample(avctx->sample_fmt);
-                buf_size += 2*FF_MIN_BUFFER_SIZE;
-            }
-        }
-        if ((ret = ff_alloc_packet2(avctx, avpkt, buf_size)))
-            return ret;
-
-        /* Encoders using AVCodec.encode() that support
-           CODEC_CAP_SMALL_LAST_FRAME require avctx->frame_size to be set to
-           the smaller size when encoding the last frame.
-           This code can be removed once all encoders supporting
-           CODEC_CAP_SMALL_LAST_FRAME use encode2() */
-        if ((avctx->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME) &&
-            nb_samples < avctx->frame_size) {
-            fs_tmp = avctx->frame_size;
-            avctx->frame_size = nb_samples;
-        }
-
-        /* encode the frame */
-        ret = avctx->codec->encode(avctx, avpkt->data, avpkt->size,
-                                   frame ? frame->data[0] : NULL);
-        if (ret >= 0) {
-            if (!ret) {
-                /* no output. if the packet data was allocated by libavcodec,
-                   free it */
-                if (!user_pkt.data && avpkt->data != avctx->internal->byte_buffer)
-                    av_freep(&avpkt->data);
-            } else {
-                if (avctx->coded_frame)
-                    avpkt->pts = avpkt->dts = avctx->coded_frame->pts;
-                /* Set duration for final small packet. This can be removed
-                   once all encoders supporting CODEC_CAP_SMALL_LAST_FRAME use
-                   encode2() */
-                if (fs_tmp) {
-                    avpkt->duration = ff_samples_to_time_base(avctx,
-                                                              avctx->frame_size);
-                }
-            }
-            avpkt->size = ret;
-            *got_packet_ptr = (ret > 0);
-            ret = 0;
-        }
-
-        if (fs_tmp)
-            avctx->frame_size = fs_tmp;
     }
     if (avpkt->data && avpkt->data == avctx->internal->byte_buffer) {
         needs_realloc = 0;
@@ -1141,8 +1073,11 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
         avctx->frame_number++;
     }
 
-    if (ret < 0 || !*got_packet_ptr)
+    if (ret < 0 || !*got_packet_ptr) {
         av_free_packet(avpkt);
+        av_init_packet(avpkt);
+        return ret;
+    }
 
     /* NOTE: if we add any audio encoders which output non-keyframe packets,
              this needs to be moved to the encoders, but for now we can do it

libavformat/dv.c

@@ -31,6 +31,7 @@
 #include <time.h>
 #include "avformat.h"
 #include "internal.h"
+#include "libavcodec/dv_profile.h"
 #include "libavcodec/dvdata.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/mathematics.h"

libavformat/dvenc.c

@@ -32,6 +32,7 @@
 
 #include "avformat.h"
 #include "internal.h"
+#include "libavcodec/dv_profile.h"
 #include "libavcodec/dvdata.h"
 #include "dv.h"
 #include "libavutil/fifo.h"

libavformat/flvenc.c

@@ -68,18 +68,19 @@ typedef struct FLVStreamContext {
     int64_t last_ts;    ///< last timestamp for each stream
 } FLVStreamContext;
 
-static int get_audio_flags(AVCodecContext *enc){
+static int get_audio_flags(AVFormatContext *s, AVCodecContext *enc)
+{
     int flags = (enc->bits_per_coded_sample == 16) ? FLV_SAMPLESSIZE_16BIT : FLV_SAMPLESSIZE_8BIT;
 
     if (enc->codec_id == CODEC_ID_AAC) // specs force these parameters
         return FLV_CODECID_AAC | FLV_SAMPLERATE_44100HZ | FLV_SAMPLESSIZE_16BIT | FLV_STEREO;
     else if (enc->codec_id == CODEC_ID_SPEEX) {
         if (enc->sample_rate != 16000) {
-            av_log(enc, AV_LOG_ERROR, "flv only supports wideband (16kHz) Speex audio\n");
+            av_log(s, AV_LOG_ERROR, "flv only supports wideband (16kHz) Speex audio\n");
             return -1;
         }
         if (enc->channels != 1) {
-            av_log(enc, AV_LOG_ERROR, "flv only supports mono Speex audio\n");
+            av_log(s, AV_LOG_ERROR, "flv only supports mono Speex audio\n");
             return -1;
         }
         return FLV_CODECID_SPEEX | FLV_SAMPLERATE_11025HZ | FLV_SAMPLESSIZE_16BIT;
@@ -102,7 +103,7 @@ static int get_audio_flags(AVCodecContext *enc){
                 break;
             }
         default:
-            av_log(enc, AV_LOG_ERROR, "flv does not support that sample rate, choose from (44100, 22050, 11025).\n");
+            av_log(s, AV_LOG_ERROR, "flv does not support that sample rate, choose from (44100, 22050, 11025).\n");
             return -1;
     }
     }
@@ -140,7 +141,7 @@ static int get_audio_flags(AVCodecContext *enc){
         flags |= enc->codec_tag<<4;
         break;
     default:
-        av_log(enc, AV_LOG_ERROR, "codec not compatible with flv\n");
+        av_log(s, AV_LOG_ERROR, "codec not compatible with flv\n");
         return -1;
     }
 
@@ -198,12 +199,12 @@ static int flv_write_header(AVFormatContext *s)
             }
             video_enc = enc;
             if(enc->codec_tag == 0) {
-                av_log(enc, AV_LOG_ERROR, "video codec not compatible with flv\n");
+                av_log(s, AV_LOG_ERROR, "video codec not compatible with flv\n");
                 return -1;
             }
         } else if (enc->codec_type == AVMEDIA_TYPE_AUDIO) {
             audio_enc = enc;
-            if(get_audio_flags(enc)<0)
+            if (get_audio_flags(s, enc) < 0)
                 return -1;
         }
         avpriv_set_pts_info(s->streams[i], 32, 1, 1000); /* 32 bit pts in ms */
@@ -344,7 +345,7 @@ static int flv_write_header(AVFormatContext *s)
             avio_wb24(pb, 0); // streamid
             pos = avio_tell(pb);
             if (enc->codec_id == CODEC_ID_AAC) {
-                avio_w8(pb, get_audio_flags(enc));
+                avio_w8(pb, get_audio_flags(s, enc));
                 avio_w8(pb, 0); // AAC sequence header
                 avio_write(pb, enc->extradata, enc->extradata_size);
             } else {
@@ -420,13 +421,13 @@ static int flv_write_packet(AVFormatContext *s, AVPacket *pkt)
 
         flags = enc->codec_tag;
         if(flags == 0) {
-            av_log(enc, AV_LOG_ERROR, "video codec %s not compatible with flv\n", avcodec_get_name(enc->codec_id));
+            av_log(s, AV_LOG_ERROR, "video codec %s not compatible with flv\n", avcodec_get_name(enc->codec_id));
             return -1;
         }
 
         flags |= pkt->flags & AV_PKT_FLAG_KEY ? FLV_FRAME_KEY : FLV_FRAME_INTER;
     } else if (enc->codec_type == AVMEDIA_TYPE_AUDIO) {
-        flags = get_audio_flags(enc);
+        flags = get_audio_flags(s, enc);
 
         assert(size);
 

libavformat/mov.c

@@ -655,9 +655,9 @@ static int mov_read_chan(MOVContext *c, AVIOContext *pb, MOVAtom atom)
 
     label_mask = 0;
     for (i = 0; i < num_descr; i++) {
-        uint32_t av_unused label, cflags;
+        uint32_t label;
         label     = avio_rb32(pb);          // mChannelLabel
-        cflags    = avio_rb32(pb);          // mChannelFlags
+        avio_rb32(pb);                      // mChannelFlags
         avio_rl32(pb);                      // mCoordinates[0]
         avio_rl32(pb);                      // mCoordinates[1]
         avio_rl32(pb);                      // mCoordinates[2]

libavformat/rtmpproto.c

@@ -66,6 +66,7 @@ typedef struct RTMPContext {
     int           chunk_size;                 ///< size of the chunks RTMP packets are divided into
     int           is_input;                   ///< input/output flag
     char          *playpath;                  ///< stream identifier to play (with possible "mp4:" prefix)
+    int           live;                       ///< 0: recorded, -1: live, -2: both
     char          *app;                       ///< name of application
     ClientState   state;                      ///< current state
     int           main_channel_id;            ///< an additional channel ID which is used for some invocations
@@ -287,7 +288,7 @@ static void gen_play(URLContext *s, RTMPContext *rt)
 
     av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
     ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
-                          20 + strlen(rt->playpath));
+                          29 + strlen(rt->playpath));
     pkt.extra = rt->main_channel_id;
 
     p = pkt.data;
@@ -295,6 +296,7 @@ static void gen_play(URLContext *s, RTMPContext *rt)
     ff_amf_write_number(&p, ++rt->nb_invokes);
     ff_amf_write_null(&p);
     ff_amf_write_string(&p, rt->playpath);
+    ff_amf_write_number(&p, rt->live);
 
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
@@ -1050,6 +1052,10 @@ static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
 
 static const AVOption rtmp_options[] = {
     {"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
+    {"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {-2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
+    {"any", "both", 0, AV_OPT_TYPE_CONST, {-2}, 0, 0, DEC, "rtmp_live"},
+    {"live", "live stream", 0, AV_OPT_TYPE_CONST, {-1}, 0, 0, DEC, "rtmp_live"},
+    {"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"},
     {"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
     { NULL },
 };

libavutil/samplefmt.c

@@ -152,7 +152,7 @@ int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
 }
 
 int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
-                           uint8_t *buf, int nb_channels, int nb_samples,
+                           const uint8_t *buf, int nb_channels, int nb_samples,
                            enum AVSampleFormat sample_fmt, int align)
 {
     int ch, planar, buf_size, line_size;

libavutil/samplefmt.h

@@ -178,7 +178,8 @@ int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
  * @param align            buffer size alignment (0 = default, 1 = no alignment)
  * @return                 0 on success or a negative error code on failure
  */
-int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, uint8_t *buf,
+int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
+                           const uint8_t *buf,
                            int nb_channels, int nb_samples,
                            enum AVSampleFormat sample_fmt, int align);
 

tests/Makefile

@@ -62,6 +62,7 @@ include $(SRC_PATH)/tests/fate/microsoft.mak
 include $(SRC_PATH)/tests/fate/mp3.mak
 include $(SRC_PATH)/tests/fate/mpc.mak
 include $(SRC_PATH)/tests/fate/pcm.mak
+include $(SRC_PATH)/tests/fate/probe.mak
 include $(SRC_PATH)/tests/fate/prores.mak
 include $(SRC_PATH)/tests/fate/qt.mak
 include $(SRC_PATH)/tests/fate/qtrle.mak
@@ -138,7 +139,7 @@ TOOL = ffmpeg
 
 fate:: $(FATE)
 
-$(FATE) $(FATE_TESTS-no): $(TOOL)$(EXESUF) $(FATE_UTILS:%=tests/%$(HOSTEXESUF))
+$(FATE) $(FATE_TESTS-no): ffmpeg$(EXESUF) ffprobe$(EXESUF) $(FATE_UTILS:%=tests/%$(HOSTEXESUF))
 	@echo "TEST    $(@:fate-%=%)"
 	$(Q)$(SRC_PATH)/tests/fate-run.sh $@ "$(SAMPLES)" "$(TARGET_EXEC)" "$(TARGET_PATH)" '$(CMD)' '$(CMP)' '$(REF)' '$(FUZZ)' '$(THREADS)' '$(THREAD_TYPE)' '$(CPUFLAGS)' '$(CMP_SHIFT)' '$(CMP_TARGET)' '$(SIZE_TOLERANCE)'
 

tests/fate-run.sh

@@ -60,11 +60,21 @@ stddev(){
     do_tiny_psnr "$1" "$2" stddev
 }
 
+oneline(){
+    val=$(cat "$2")
+    test x"$val" = x"$1" || { r=$?; printf -- '-%s\n+%s\n' "$ref" "$val"; }
+    return ${r:-0}
+}
+
 run(){
     test "${V:-0}" -gt 0 && echo "$target_exec" $target_path/"$@" >&3
     $target_exec $target_path/"$@"
 }
 
+probefmt(){
+    run ffprobe -show_format_entry format_name -v 0 "$@"
+}
+
 avconv(){
     run ffmpeg -nostats -threads $threads -thread_type $thread_type -cpuflags $cpuflags "$@"
 }
@@ -151,11 +161,12 @@ if [ $err -gt 128 ]; then
     test "${sig}" = "${sig%[!A-Za-z]*}" || unset sig
 fi
 
-if test -e "$ref"; then
+if test -e "$ref" || test $cmp = "oneline" ; then
     case $cmp in
         diff)   diff -u -w "$ref" "$outfile"            >$cmpfile ;;
         oneoff) oneoff     "$ref" "$outfile"            >$cmpfile ;;
         stddev) stddev     "$ref" "$outfile"            >$cmpfile ;;
+        oneline)oneline    "$ref" "$outfile"            >$cmpfile ;;
         null)   cat               "$outfile"            >$cmpfile ;;
     esac
     cmperr=$?

tests/fate/probe.mak

+FATE_PROBE_FORMAT += fate-probe-format-roundup997
+fate-probe-format-roundup997:  REF = format_name=mpeg
+
+FATE_PROBE_FORMAT += fate-probe-format-roundup1383
+fate-probe-format-roundup1383: REF = format_name=mp3
+
+FATE_PROBE_FORMAT += fate-probe-format-roundup1414
+fate-probe-format-roundup1414: REF = format_name=mpeg
+
+FATE_PROBE_FORMAT += fate-probe-format-roundup2015
+fate-probe-format-roundup2015: REF = format_name=dv
+
+FATE_TESTS += $(FATE_PROBE_FORMAT)
+fate-probe-format: $(FATE_PROBE_FORMAT)
+
+$(FATE_PROBE_FORMAT): CMP = oneline
+fate-probe-format-%: CMD = probefmt $(SAMPLES)/probe-format/$(@:fate-probe-format-%=%)