Merge remote-tracking branch 'qatar/master'

* qatar/master: (23 commits)
  build: cosmetics: Reorder some lists in a more logical fashion
  x86: pngdsp: Fix assembly for OS/2
  fate: add test for RTjpeg in nuv with frameheader
  rtmp: send check_bw as notification
  g723_1: clip argument for 15-bit version of normalize_bits()
  g723_1: use all LPC vectors in formant postfilter
  id3v2: Support v2.2 PIC
  avplay: fix build with lavfi disabled.
  avconv: split configuring filter configuration to a separate file.
  avconv: split option parsing into a separate file.
  mpc8: do not leave padding after last frame in buffer for the next decode call
  mpegaudioenc: list supported channel layouts.
  mpegaudiodec: don't print an error on > 1 frame in a packet.
  api-example: update to new audio encoding API.
  configure: add --enable/disable-random option
  doc: cygwin: Update list of FATE package requirements
  build: Remove all installed headers and header directories on uninstall
  build: change checkheaders to use regular build rules
  rtmp: Add a new option 'rtmp_subscribe'
  rtmp: Add support for subscribing live streams
  ...

Conflicts:
	Makefile
	common.mak
	configure
	doc/examples/decoding_encoding.c
	ffmpeg.c
	libavcodec/g723_1.c
	libavcodec/mpegaudiodec.c
	libavcodec/x86/pngdsp.asm
	libavformat/version.h
	library.mak
	tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>

.gitignore

@@ -4,7 +4,7 @@
 *.def
 *.dll
 *.exe
-*.ho
+*.h.c
 *.lib
 *.pc
 *.so

Makefile

@@ -18,6 +18,7 @@ PROGS-$(CONFIG_FFSERVER) += ffserver
 PROGS      := $(PROGS-yes:%=%$(EXESUF))
 INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
 OBJS        = cmdutils.o
+OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
 TOOLS       = qt-faststart trasher
@@ -68,8 +69,9 @@ config.h: .config
 
 SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
                ARCH_HEADERS BUILT_HEADERS SKIPHEADERS                    \
-               ALTIVEC-OBJS ARMV5TE-OBJS ARMV6-OBJS ARMVFP-OBJS MMI-OBJS \
-               MMX-OBJS NEON-OBJS VIS-OBJS YASM-OBJS                     \
+               ARMV5TE-OBJS ARMV6-OBJS ARMVFP-OBJS NEON-OBJS             \
+               MMI-OBJS ALTIVEC-OBJS VIS-OBJS                            \
+               MMX-OBJS YASM-OBJS                                        \
                MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS  \
                OBJS TESTOBJS
 

common.mak

@@ -10,8 +10,9 @@ ifndef SUBDIR
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX AS YASM AR LD HOSTCC STRIP CP
-SILENT = DEPCC DEPAS DEPHOSTCC DEPYASM RM RANLIB
+BRIEF  = CC CXX HOSTCC AS YASM AR LD STRIP CP
+SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
+
 MSG    = $@
 M      = @$(call ECHO,$(TAG),$@);
 $(foreach VAR,$(BRIEF), \
@@ -98,8 +99,9 @@ DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
 SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
 SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
-HEADEROBJS  := $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))
-checkheaders:  $(HEADEROBJS)
+HOBJS        = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
+checkheaders: $(HOBJS)
+.SECONDARY:   $(HOBJS:.o=.c)
 
 alltools: $(TOOLS)
 
@@ -117,8 +119,8 @@ $(TOOLOBJS):   | tools
 
 OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS) $(HEADEROBJS))
 
-CLEANSUFFIXES     = *.d *.o *~ *.ho *.map *.ver *.gcno *.gcda
+CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
 
--include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d))
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))

configure

@@ -1211,6 +1211,7 @@ HAVE_LIST="
     alsa_asoundlib_h
     altivec_h
     arpa_inet_h
+    asm_mod_q
     asm_mod_y
     asm_types_h
     attribute_may_alias
@@ -3119,6 +3120,7 @@ EOF
     enabled neon    && check_asm neon    '"vadd.i16 q0, q0, q0"'
     enabled vfpv3   && check_asm vfpv3   '"vmov.f32 s0, #1.0"'
 
+    check_asm asm_mod_q '"add r0, %Q0, %R0" :: "r"((long long)0)'
     check_asm asm_mod_y '"vmul.i32 d0, d0, %y0" :: "x"(0)'
 
     enabled_all armv6t2 shared !pic && enable_pic

doc/examples/decoding_encoding.c

@@ -34,6 +34,7 @@
 #include <libavutil/imgutils.h>
 #include <libavutil/opt.h>
 #include <libavcodec/avcodec.h>
+#include <libavutil/audioconvert.h>
 #include <libavutil/mathematics.h>
 #include <libavutil/samplefmt.h>
 
@@ -41,6 +42,59 @@
 #define AUDIO_INBUF_SIZE 20480
 #define AUDIO_REFILL_THRESH 4096
 
+/* check that a given sample format is supported by the encoder */
+static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
+{
+    const enum AVSampleFormat *p = codec->sample_fmts;
+
+    while (*p != AV_SAMPLE_FMT_NONE) {
+        if (*p == sample_fmt)
+            return 1;
+        p++;
+    }
+    return 0;
+}
+
+/* just pick the highest supported samplerate */
+static int select_sample_rate(AVCodec *codec)
+{
+    const int *p;
+    int best_samplerate = 0;
+
+    if (!codec->supported_samplerates)
+        return 44100;
+
+    p = codec->supported_samplerates;
+    while (*p) {
+        best_samplerate = FFMAX(*p, best_samplerate);
+        p++;
+    }
+    return best_samplerate;
+}
+
+/* select layout with the highest channel count */
+static int select_channel_layout(AVCodec *codec)
+{
+    const uint64_t *p;
+    uint64_t best_ch_layout = 0;
+    int best_nb_channells   = 0;
+
+    if (!codec->channel_layouts)
+        return AV_CH_LAYOUT_STEREO;
+
+    p = codec->channel_layouts;
+    while (*p) {
+        int nb_channels = av_get_channel_layout_nb_channels(*p);
+
+        if (nb_channels > best_nb_channells) {
+            best_ch_layout    = *p;
+            best_nb_channells = nb_channels;
+        }
+        p++;
+    }
+    return best_ch_layout;
+}
+
 /*
  * Audio encoding example
  */
@@ -48,11 +102,13 @@ static void audio_encode_example(const char *filename)
 {
     AVCodec *codec;
     AVCodecContext *c= NULL;
-    int frame_size, i, j, out_size, outbuf_size;
+    AVFrame *frame;
+    AVPacket pkt;
+    int i, j, k, ret, got_output;
+    int buffer_size;
     FILE *f;
-    short *samples;
+    uint16_t *samples;
     float t, tincr;
-    uint8_t *outbuf;
 
     printf("Encode audio file %s\n", filename);
 
@@ -67,9 +123,19 @@ static void audio_encode_example(const char *filename)
 
     /* put sample parameters */
     c->bit_rate = 64000;
-    c->sample_rate = 44100;
-    c->channels = 2;
+
+    /* check that the encoder supports s16 pcm input */
     c->sample_fmt = AV_SAMPLE_FMT_S16;
+    if (!check_sample_fmt(codec, c->sample_fmt)) {
+        fprintf(stderr, "encoder does not support %s",
+                av_get_sample_fmt_name(c->sample_fmt));
+        exit(1);
+    }
+
+    /* select other audio parameters supported by the encoder */
+    c->sample_rate    = select_sample_rate(codec);
+    c->channel_layout = select_channel_layout(codec);
+    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
 
     /* open it */
     if (avcodec_open2(c, codec, NULL) < 0) {
@@ -77,35 +143,71 @@ static void audio_encode_example(const char *filename)
         exit(1);
     }
 
-    /* the codec gives us the frame size, in samples */
-    frame_size = c->frame_size;
-    samples = malloc(frame_size * 2 * c->channels);
-    outbuf_size = 10000;
-    outbuf = malloc(outbuf_size);
-
     f = fopen(filename, "wb");
     if (!f) {
         fprintf(stderr, "could not open %s\n", filename);
         exit(1);
     }
 
+    /* frame containing input raw audio */
+    frame = avcodec_alloc_frame();
+    if (!frame) {
+        fprintf(stderr, "could not allocate audio frame\n");
+        exit(1);
+    }
+
+    frame->nb_samples     = c->frame_size;
+    frame->format         = c->sample_fmt;
+    frame->channel_layout = c->channel_layout;
+
+    /* the codec gives us the frame size, in samples,
+     * we calculate the size of the samples buffer in bytes */
+    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
+                                             c->sample_fmt, 0);
+    samples = av_malloc(buffer_size);
+    if (!samples) {
+        fprintf(stderr, "could not allocate %d bytes for samples buffer\n",
+                buffer_size);
+        exit(1);
+    }
+    /* setup the data pointers in the AVFrame */
+    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
+                                   (const uint8_t*)samples, buffer_size, 0);
+    if (ret < 0) {
+        fprintf(stderr, "could not setup audio frame\n");
+        exit(1);
+    }
+
     /* encode a single tone sound */
     t = 0;
     tincr = 2 * M_PI * 440.0 / c->sample_rate;
     for(i=0;i<200;i++) {
-        for(j=0;j<frame_size;j++) {
+        av_init_packet(&pkt);
+        pkt.data = NULL; // packet data will be allocated by the encoder
+        pkt.size = 0;
+
+        for (j = 0; j < c->frame_size; j++) {
             samples[2*j] = (int)(sin(t) * 10000);
-            samples[2*j+1] = samples[2*j];
+
+            for (k = 1; k < c->channels; k++)
+                samples[2*j + k] = samples[2*j];
             t += tincr;
         }
         /* encode the samples */
-        out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
-        fwrite(outbuf, 1, out_size, f);
+        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "error encoding audio frame\n");
+            exit(1);
+        }
+        if (got_output) {
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_free_packet(&pkt);
+        }
     }
     fclose(f);
-    free(outbuf);
-    free(samples);
 
+    av_freep(&samples);
+    av_freep(&frame);
     avcodec_close(c);
     av_free(c);
 }

doc/platform.texi

@@ -318,9 +318,9 @@ following "Devel" ones:
 binutils, gcc4-core, make, git, mingw-runtime, texi2html
 @end example
 
-And the following "Utils" one:
+In order to run FATE you will also need the following "Utils" packages:
 @example
-diffutils
+bc, diffutils
 @end example
 
 Then run

doc/protocols.texi

@@ -267,6 +267,11 @@ value will be sent.
 Stream identifier to play or to publish. This option overrides the
 parameter specified in the URI.
 
+@item rtmp_subscribe
+Name of live stream to subscribe to. By default no value will be sent.
+It is only sent if the option is specified or if rtmp_live
+is set to live.
+
 @item rtmp_swfurl
 URL of the SWF player for the media. By default no value will be sent.
 

libavcodec/g723_1.c

@@ -264,6 +264,7 @@ static int scale_vector(int16_t *vector, int length)
     for (i = 0; i < length; i++)
         max = FFMAX(max, FFABS(vector[i]));
 
+    max   = FFMIN(max, 0x7FFF);
     bits  = normalize_bits(max, 15);
     scale = shift_table[bits];
 
@@ -913,6 +914,7 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf)
         }
         iir_filter(filter_coef[0], filter_coef[1], buf + i,
                    filter_signal + i, 1);
+        lpc += LPC_ORDER;
     }
 
     memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));

libavcodec/mpc8.c

@@ -419,6 +419,8 @@ static int mpc8_decode_frame(AVCodecContext * avctx, void *data,
     c->cur_frame++;
 
     c->last_bits_used = get_bits_count(gb);
+    if(get_bits_left(gb) < 8) // we have only padding left
+        c->last_bits_used = buf_size << 3;
     if(c->cur_frame >= c->frames)
         c->cur_frame = 0;
 

libavcodec/mpegaudiodec.c

@@ -1686,7 +1686,7 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
     if (s->frame_size <= 0 || s->frame_size > buf_size) {
         av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
         return AVERROR_INVALIDDATA;
-    }else if(s->frame_size < buf_size){
+    } else if (s->frame_size < buf_size) {
         av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");
         buf_size= s->frame_size;
     }

libavcodec/mpegaudioenc.c

@@ -24,6 +24,8 @@
  * The simplest mpeg audio layer 2 encoder.
  */
 
+#include "libavutil/audioconvert.h"
+
 #include "avcodec.h"
 #include "internal.h"
 #include "put_bits.h"
@@ -792,6 +794,9 @@ AVCodec ff_mp2_encoder = {
     .supported_samplerates = (const int[]){
         44100, 48000,  32000, 22050, 24000, 16000, 0
     },
+    .channel_layouts       = (const uint64_t[]){ AV_CH_LAYOUT_MONO,
+                                                 AV_CH_LAYOUT_STEREO,
+                                                 0 },
     .long_name             = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
     .defaults              = mp2_defaults,
 };

libavcodec/x86/pngdsp.asm

@@ -28,7 +28,7 @@ SECTION_RODATA
 
 cextern pw_255
 
-SECTION_TEXT 16
+SECTION_TEXT
 
 ; %1 = nr. of xmm registers used
 %macro ADD_BYTES_FN 1

libavformat/id3v2.c

@@ -131,6 +131,8 @@ const CodecMime ff_id3v2_mime_tags[] = {
     {"image/png" , AV_CODEC_ID_PNG},
     {"image/tiff", AV_CODEC_ID_TIFF},
     {"image/bmp",  AV_CODEC_ID_BMP},
+    {"JPG",        AV_CODEC_ID_MJPEG}, /* ID3v2.2  */
+    {"PNG" ,       AV_CODEC_ID_PNG},   /* ID3v2.2  */
     {"",           AV_CODEC_ID_NONE},
 };
 

libavformat/rtmp.h

@@ -32,6 +32,15 @@
 #define HMAC_IPAD_VAL 0x36
 #define HMAC_OPAD_VAL 0x5C
 
+/**
+ * A non-zero transaction id requires the server to send back
+ * a _result or _error response.
+ * Setting it to 0 marks the message as a notification not
+ * requiring feedback.
+ */
+
+#define RTMP_NOTIFICATION 0
+
 /**
  * emulated Flash client version - 9.0.124.2 on Linux
  * @{

libavformat/rtmpproto.c

@@ -91,6 +91,7 @@ typedef struct RTMPContext {
     char*         flashver;                   ///< version of the flash plugin
     char*         swfurl;                     ///< url of the swf player
     char*         pageurl;                    ///< url of the web page
+    char*         subscribe;                  ///< name of live stream to subscribe
     int           server_bw;                  ///< server bandwidth
     int           client_buffer_time;         ///< client buffer time in ms
     int           flush_interval;             ///< number of packets flushed in the same request (RTMPT only)
@@ -572,7 +573,7 @@ static int gen_check_bw(URLContext *s, RTMPContext *rt)
 
     p = pkt.data;
     ff_amf_write_string(&p, "_checkbw");
-    ff_amf_write_number(&p, ++rt->nb_invokes);
+    ff_amf_write_number(&p, RTMP_NOTIFICATION);
     ff_amf_write_null(&p);
 
     ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
@@ -604,6 +605,30 @@ static int gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
     return ret;
 }
 
+static int gen_fcsubscribe_stream(URLContext *s, RTMPContext *rt,
+                                  const char *subscribe)
+{
+    RTMPPacket pkt;
+    uint8_t *p;
+    int ret;
+
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
+                                     0, 27 + strlen(subscribe))) < 0)
+        return ret;
+
+    p = pkt.data;
+    ff_amf_write_string(&p, "FCSubscribe");
+    ff_amf_write_number(&p, ++rt->nb_invokes);
+    ff_amf_write_null(&p);
+    ff_amf_write_string(&p, subscribe);
+
+    ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
+                               rt->prev_pkt[1]);
+    ff_rtmp_packet_destroy(&pkt);
+
+    return ret;
+}
+
 int ff_rtmp_calc_digest(const uint8_t *src, int len, int gap,
                         const uint8_t *key, int keylen, uint8_t *dst)
 {
@@ -1011,6 +1036,20 @@ static int handle_invoke(URLContext *s, RTMPPacket *pkt)
                 }
                 if ((ret = gen_create_stream(s, rt)) < 0)
                     return ret;
+
+                if (rt->is_input) {
+                    /* Send the FCSubscribe command when the name of live
+                     * stream is defined by the user or if it's a live stream. */
+                    if (rt->subscribe) {
+                        if ((ret = gen_fcsubscribe_stream(s, rt,
+                                                          rt->subscribe)) < 0)
+                            return ret;
+                    } else if (rt->live == -1) {
+                        if ((ret = gen_fcsubscribe_stream(s, rt,
+                                                          rt->playpath)) < 0)
+                            return ret;
+                    }
+                }
                 break;
             case STATE_FCPUBLISH:
                 rt->state = STATE_CONNECTING;
@@ -1593,115 +1632,35 @@ static const AVOption rtmp_options[] = {
     {"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"},
     {"rtmp_pageurl", "URL of the web page in which the media was embedded. By default no value will be sent.", OFFSET(pageurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
     {"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
+    {"rtmp_subscribe", "Name of live stream to subscribe to. Defaults to rtmp_playpath.", OFFSET(subscribe), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
     {"rtmp_swfurl", "URL of the SWF player. By default no value will be sent", OFFSET(swfurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
     {"rtmp_tcurl", "URL of the target stream. Defaults to proto://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
     { NULL },
 };
 
-static const AVClass rtmp_class = {
-    .class_name = "rtmp",
-    .item_name  = av_default_item_name,
-    .option     = rtmp_options,
-    .version    = LIBAVUTIL_VERSION_INT,
-};
-
-URLProtocol ff_rtmp_protocol = {
-    .name           = "rtmp",
-    .url_open       = rtmp_open,
-    .url_read       = rtmp_read,
-    .url_write      = rtmp_write,
-    .url_close      = rtmp_close,
-    .priv_data_size = sizeof(RTMPContext),
-    .flags          = URL_PROTOCOL_FLAG_NETWORK,
-    .priv_data_class= &rtmp_class,
+#define RTMP_PROTOCOL(flavor)                    \
+static const AVClass flavor##_class = {          \
+    .class_name = #flavor,                       \
+    .item_name  = av_default_item_name,          \
+    .option     = rtmp_options,                  \
+    .version    = LIBAVUTIL_VERSION_INT,         \
+};                                               \
+                                                 \
+URLProtocol ff_##flavor##_protocol = {           \
+    .name           = #flavor,                   \
+    .url_open       = rtmp_open,                 \
+    .url_read       = rtmp_read,                 \
+    .url_write      = rtmp_write,                \
+    .url_close      = rtmp_close,                \
+    .priv_data_size = sizeof(RTMPContext),       \
+    .flags          = URL_PROTOCOL_FLAG_NETWORK, \
+    .priv_data_class= &flavor##_class,           \
 };
 
-static const AVClass rtmpe_class = {
-    .class_name = "rtmpe",
-    .item_name  = av_default_item_name,
-    .option     = rtmp_options,
-    .version    = LIBAVUTIL_VERSION_INT,
-};
-
-URLProtocol ff_rtmpe_protocol = {
-    .name            = "rtmpe",
-    .url_open        = rtmp_open,
-    .url_read        = rtmp_read,
-    .url_write       = rtmp_write,
-    .url_close       = rtmp_close,
-    .priv_data_size  = sizeof(RTMPContext),
-    .flags           = URL_PROTOCOL_FLAG_NETWORK,
-    .priv_data_class = &rtmpe_class,
-};
 
-static const AVClass rtmps_class = {
-    .class_name = "rtmps",
-    .item_name  = av_default_item_name,
-    .option     = rtmp_options,
-    .version    = LIBAVUTIL_VERSION_INT,
-};
-
-URLProtocol ff_rtmps_protocol = {
-    .name            = "rtmps",
-    .url_open        = rtmp_open,
-    .url_read        = rtmp_read,
-    .url_write       = rtmp_write,
-    .url_close       = rtmp_close,
-    .priv_data_size  = sizeof(RTMPContext),
-    .flags           = URL_PROTOCOL_FLAG_NETWORK,
-    .priv_data_class = &rtmps_class,
-};
-
-static const AVClass rtmpt_class = {
-    .class_name = "rtmpt",
-    .item_name  = av_default_item_name,
-    .option     = rtmp_options,
-    .version    = LIBAVUTIL_VERSION_INT,
-};
-
-URLProtocol ff_rtmpt_protocol = {
-    .name            = "rtmpt",
-    .url_open        = rtmp_open,
-    .url_read        = rtmp_read,
-    .url_write       = rtmp_write,
-    .url_close       = rtmp_close,
-    .priv_data_size  = sizeof(RTMPContext),
-    .flags           = URL_PROTOCOL_FLAG_NETWORK,
-    .priv_data_class = &rtmpt_class,
-};
-
-static const AVClass rtmpte_class = {
-    .class_name = "rtmpte",
-    .item_name  = av_default_item_name,
-    .option     = rtmp_options,
-    .version    = LIBAVUTIL_VERSION_INT,
-};
-
-URLProtocol ff_rtmpte_protocol = {
-    .name            = "rtmpte",
-    .url_open        = rtmp_open,
-    .url_read        = rtmp_read,
-    .url_write       = rtmp_write,
-    .url_close       = rtmp_close,
-    .priv_data_size  = sizeof(RTMPContext),
-    .flags           = URL_PROTOCOL_FLAG_NETWORK,
-    .priv_data_class = &rtmpte_class,
-};
-
-static const AVClass rtmpts_class = {
-    .class_name = "rtmpts",
-    .item_name  = av_default_item_name,
-    .option     = rtmp_options,
-    .version    = LIBAVUTIL_VERSION_INT,
-};
-
-URLProtocol ff_rtmpts_protocol = {
-    .name            = "rtmpts",
-    .url_open        = rtmp_open,
-    .url_read        = rtmp_read,
-    .url_write       = rtmp_write,
-    .url_close       = rtmp_close,
-    .priv_data_size  = sizeof(RTMPContext),
-    .flags           = URL_PROTOCOL_FLAG_NETWORK,
-    .priv_data_class = &rtmpts_class,
-};
+RTMP_PROTOCOL(rtmp)
+RTMP_PROTOCOL(rtmpe)
+RTMP_PROTOCOL(rtmps)
+RTMP_PROTOCOL(rtmpt)
+RTMP_PROTOCOL(rtmpte)
+RTMP_PROTOCOL(rtmpts)

libavformat/version.h

@@ -31,7 +31,7 @@
 
 #define LIBAVFORMAT_VERSION_MAJOR 54
 #define LIBAVFORMAT_VERSION_MINOR 22
-#define LIBAVFORMAT_VERSION_MICRO 102
+#define LIBAVFORMAT_VERSION_MICRO 103
 
 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
                                                LIBAVFORMAT_VERSION_MINOR, \

libavutil/arm/intmath.h

@@ -97,6 +97,8 @@ static av_always_inline av_const int FASTDIV(int a, int b)
 
 #endif /* HAVE_ARMV6 */
 
+#if HAVE_ASM_MOD_Q
+
 #define av_clipl_int32 av_clipl_int32_arm
 static av_always_inline av_const int32_t av_clipl_int32_arm(int64_t a)
 {
@@ -110,6 +112,8 @@ static av_always_inline av_const int32_t av_clipl_int32_arm(int64_t a)
     return x;
 }
 
+#endif /* HAVE_ASM_MOD_Q */
+
 #endif /* HAVE_INLINE_ASM */
 
 #endif /* AVUTIL_ARM_INTMATH_H */

libavutil/arm/intreadwrite.h

@@ -61,6 +61,8 @@ static av_always_inline void AV_WN32(void *p, uint32_t v)
     __asm__ ("str  %1, %0" : "=m"(*(uint32_t *)p) : "r"(v));
 }
 
+#if HAVE_ASM_MOD_Q
+
 #define AV_RN64 AV_RN64
 static av_always_inline uint64_t AV_RN64(const void *p)
 {
@@ -82,6 +84,8 @@ static av_always_inline void AV_WN64(void *p, uint64_t v)
              : "r"(v));
 }
 
+#endif /* HAVE_ASM_MOD_Q */
+
 #endif /* HAVE_INLINE_ASM */
 
 #endif /* AVUTIL_ARM_INTREADWRITE_H */

library.mak

@@ -20,7 +20,7 @@ $(SUBDIR)x86/%.o: $(SUBDIR)x86/%.asm
 	$(DEPYASM) $(YASMFLAGS) -I $(<D)/ -M -o $@ $< > $(@:.o=.d)
 	$(YASM) $(YASMFLAGS) -I $(<D)/ -o $@ $<
 
-$(OBJS) $(OBJS:.o=.s) $(SUBDIR)%.ho $(TESTOBJS): CPPFLAGS += -DHAVE_AV_CONFIG_H
+$(OBJS) $(OBJS:.o=.s) $(SUBDIR)%.h.o $(TESTOBJS): CPPFLAGS += -DHAVE_AV_CONFIG_H
 $(TESTOBJS): CPPFLAGS += -DTEST
 
 $(SUBDIR)$(LIBNAME): $(OBJS)
@@ -87,7 +87,7 @@ uninstall-libs::
 	-$(RM) "$(LIBDIR)/$(LIBNAME)"
 
 uninstall-headers::
-	$(RM) $(addprefix "$(INCINSTDIR)/",$(HEADERS)) $(addprefix "$(INCINSTDIR)/",$(BUILT_HEADERS))
+	$(RM) $(addprefix "$(INCINSTDIR)/",$(HEADERS) $(BUILT_HEADERS))
 	$(RM) "$(LIBDIR)/pkgconfig/lib$(NAME).pc"
 	-rmdir "$(INCINSTDIR)"
 endef

tests/fate/video.mak

@@ -167,8 +167,14 @@ FATE_VIDEO += fate-mpeg2-field-enc
 fate-mpeg2-field-enc: CMD = framecrc -flags +bitexact -dct fastint -idct simple -i $(SAMPLES)/mpeg2/mpeg2_field_encoding.ts -an
 
 # FIXME dropped frames in this test because of coarse timebase
-FATE_VIDEO += fate-nuv
-fate-nuv: CMD = framecrc -idct simple -i $(SAMPLES)/nuv/Today.nuv -an
+FATE_NUV += fate-nuv-rtjpeg
+fate-nuv-rtjpeg: CMD = framecrc -idct simple -i $(SAMPLES)/nuv/Today.nuv -an
+
+FATE_NUV += fate-nuv-rtjpeg-fh
+fate-nuv-rtjpeg-fh: CMD = framecrc -idct simple -i $(SAMPLES)/nuv/rtjpeg_frameheader.nuv -an
+
+FATE_VIDEO += $(FATE_NUV)
+fate-nuv: $(FATE_NUV)
 
 FATE_VIDEO += fate-paf-video
 fate-paf-video: CMD = framecrc -i $(SAMPLES)/paf/hod1-partial.paf -pix_fmt rgb24 -an

tests/ref/fate/acodec-g723_1

 dec0deb2425e908d232d2471acff04a3 *tests/data/fate/acodec-g723_1.g723_1
 4800 tests/data/fate/acodec-g723_1.g723_1
-87fd529c9e41914f73a865d147cc9516 *tests/data/fate/acodec-g723_1.out.wav
-stddev: 8425.98 PSNR: 17.82 MAXDIFF:53268 bytes:    95992/    96000
+d70776846d77c652bceed281fcca9cc8 *tests/data/fate/acodec-g723_1.out.wav
+stddev: 8423.47 PSNR: 17.82 MAXDIFF:53292 bytes:    95992/    96000

tests/ref/fate/nuv-rtjpeg-fh

+#tb 0: 1/50
+0,          0,          0,        1,   221184, 0xf48c94f6
+0,          2,          2,        1,   221184, 0x89b625b2
+0,          3,          3,        1,   221184, 0x37e04714
+0,          4,          4,        1,   221184, 0x4f4c5224
+0,          5,          5,        1,   221184, 0x9193c9f1
+0,          6,          6,        1,   221184, 0x5d1a6197
+0,          7,          7,        1,   221184, 0x40cd51e7
+0,          8,          8,        1,   221184, 0xb2c1a729
+0,         10,         10,        1,   221184, 0x998d6144
+0,         11,         11,        1,   221184, 0xf5d52311
+0,         12,         12,        1,   221184, 0xea9dd6bf
+0,         13,         13,        1,   221184, 0x0e2ed854
+0,         14,         14,        1,   221184, 0xe295ba58
+0,         15,         15,        1,   221184, 0x8aedbb69
+0,         16,         16,        1,   221184, 0x253c9aaa
+0,         17,         17,        1,   221184, 0x5eaf9fb1
+0,         18,         18,        1,   221184, 0xcdb5a0cb
+0,         19,         19,        1,   221184, 0xcdb5a0cb
+0,         20,         20,        1,   221184, 0x23f89994
+0,         21,         21,        1,   221184, 0x23f89994
+0,         22,         22,        1,   221184, 0x10dc98d6
+0,         23,         23,        1,   221184, 0x799b9d98
+0,         24,         24,        1,   221184, 0xb226996c
+0,         25,         25,        1,   221184, 0x0ac59a42
+0,         26,         26,        1,   221184, 0x87c2a654
+0,         27,         27,        1,   221184, 0xf4c1a711
+0,         28,         28,        1,   221184, 0xf60fa72e
+0,         29,         29,        1,   221184, 0xc8f8b6fc
+0,         30,         30,        1,   221184, 0xd709b813
+0,         31,         31,        1,   221184, 0x5fdfb76b
+0,         32,         32,        1,   221184, 0x5798b0aa
+0,         33,         33,        1,   221184, 0xf572b1c3
+0,         34,         34,        1,   221184, 0x14b0afdf
+0,         35,         35,        1,   221184, 0x0a66b5b8
+0,         36,         36,        1,   221184, 0xe316c620
+0,         37,         37,        1,   221184, 0xbc76c5c2
+0,         38,         38,        1,   221184, 0x77c7c5e5
+0,         39,         39,        1,   221184, 0xfc7ac63e
+0,         40,         40,        1,   221184, 0x05a29ffe
+0,         41,         41,        1,   221184, 0x9bffbf6c
+0,         42,         42,        1,   221184, 0x3c55be40
+0,         43,         43,        1,   221184, 0x6f46c14e
+0,         44,         44,        1,   221184, 0x9cf4ae70
+0,         45,         45,        1,   221184, 0xf205b2f8
+0,         46,         46,        1,   221184, 0x7180aff8
+0,         47,         47,        1,   221184, 0x125eaffe
+0,         48,         48,        1,   221184, 0x6970a32d
+0,         49,         49,        1,   221184, 0xaea79f62
+0,         50,         50,        1,   221184, 0x48d2a093
+0,         51,         51,        1,   221184, 0x10a59eb5