Merge remote-tracking branch 'qatar/master'

* qatar/master:
  rtmp: Return a proper error code instead of -1
  rtmp: Check malloc calls
  rtmp: Check ff_rtmp_packet_create calls
  lavfi: add audio mix filter
  flvdec: Make sure sample_rate is set to the updated value
  tqi: Pass errors from the MB decoder

Conflicts:
	Changelog
	doc/filters.texi
	libavcodec/eatqi.c
	libavfilter/Makefile
	libavfilter/allfilters.c
	libavfilter/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Changelog

@@ -38,6 +38,7 @@ version 0.11:
 - audio split filter
 - vorbis parser
 - png parser
+- audio mix filter
 
 
 version 0.10:

doc/filters.texi

@@ -216,6 +216,44 @@ amovie=input.mkv:si=5 [a5];
 [x3][a5] amerge" -c:a pcm_s16le output.mkv
 @end example
 
+@section amix
+
+Mixes multiple audio inputs into a single output.
+
+For example
+@example
+ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT
+@end example
+will mix 3 input audio streams to a single output with the same duration as the
+first input and a dropout transition time of 3 seconds.
+
+The filter accepts the following named parameters:
+@table @option
+
+@item inputs
+Number of inputs. If unspecified, it defaults to 2.
+
+@item duration
+How to determine the end-of-stream.
+@table @option
+
+@item longest
+Duration of longest input. (default)
+
+@item shortest
+Duration of shortest input.
+
+@item first
+Duration of first input.
+
+@end table
+
+@item dropout_transition
+Transition time, in seconds, for volume renormalization when an input
+stream ends. The default value is 2 seconds.
+
+@end table
+
 @section anull
 
 Pass the audio source unchanged to the output.

libavcodec/eatqi.c

@@ -62,7 +62,7 @@ static int tqi_decode_mb(MpegEncContext *s, DCTELEM (*block)[64])
     int n;
     s->dsp.clear_blocks(block[0]);
     for (n=0; n<6; n++)
-        if(ff_mpeg1_decode_block_intra(s, block[n], n)<0)
+        if (ff_mpeg1_decode_block_intra(s, block[n], n) < 0)
             return -1;
 
     return 0;
@@ -138,7 +138,7 @@ static int tqi_decode_frame(AVCodecContext *avctx,
     for (s->mb_y=0; s->mb_y<(avctx->height+15)/16; s->mb_y++)
     for (s->mb_x=0; s->mb_x<(avctx->width+15)/16; s->mb_x++)
     {
-        if(tqi_decode_mb(s, t->block) < 0)
+        if (tqi_decode_mb(s, t->block) < 0)
             goto end;
         tqi_idct_put(t, t->block);
     }

libavfilter/Makefile

@@ -47,6 +47,7 @@ OBJS-$(CONFIG_SWSCALE)                       += lswsutils.o
 OBJS-$(CONFIG_ACONVERT_FILTER)               += af_aconvert.o
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
 OBJS-$(CONFIG_AMERGE_FILTER)                 += af_amerge.o
+OBJS-$(CONFIG_AMIX_FILTER)                   += af_amix.o
 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
 OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o
 OBJS-$(CONFIG_ASHOWINFO_FILTER)              += af_ashowinfo.o

libavfilter/allfilters.c

@@ -37,6 +37,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER (ACONVERT,    aconvert,    af);
     REGISTER_FILTER (AFORMAT,     aformat,     af);
     REGISTER_FILTER (AMERGE,      amerge,      af);
+    REGISTER_FILTER (AMIX,        amix,        af);
     REGISTER_FILTER (ANULL,       anull,       af);
     REGISTER_FILTER (ARESAMPLE,   aresample,   af);
     REGISTER_FILTER (ASHOWINFO,   ashowinfo,   af);

libavfilter/version.h

@@ -29,7 +29,7 @@
 #include "libavutil/avutil.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  2
-#define LIBAVFILTER_VERSION_MINOR 76
+#define LIBAVFILTER_VERSION_MINOR 77
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \

libavformat/flvdec.c

@@ -591,8 +591,8 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
         }
         if(!st->codec->codec_id){
             flv_set_audio_codec(s, st, st->codec, flags & FLV_AUDIO_CODECID_MASK);
-            flv->last_sample_rate = st->codec->sample_rate;
-            flv->last_channels    = st->codec->channels;
+            flv->last_sample_rate = sample_rate = st->codec->sample_rate;
+            flv->last_channels    = channels    = st->codec->channels;
         } else {
             AVCodecContext ctx;
             ctx.sample_rate = sample_rate;

libavformat/rtmppkt.c

@@ -79,6 +79,7 @@ int ff_rtmp_packet_read(URLContext *h, RTMPPacket *p,
     uint32_t extra = 0;
     enum RTMPPacketType type;
     int size = 0;
+    int ret;
 
     if (ffurl_read(h, &hdr, 1) != 1)
         return AVERROR(EIO);
@@ -129,8 +130,9 @@ int ff_rtmp_packet_read(URLContext *h, RTMPPacket *p,
     if (hdr != RTMP_PS_TWELVEBYTES)
         timestamp += prev_pkt[channel_id].timestamp;
 
-    if (ff_rtmp_packet_create(p, channel_id, type, timestamp, data_size))
-        return -1;
+    if ((ret = ff_rtmp_packet_create(p, channel_id, type, timestamp,
+                                     data_size)) < 0)
+        return ret;
     p->extra = extra;
     // save history
     prev_pkt[channel_id].channel_id = channel_id;

tests/fate/audio.mak

@@ -39,8 +39,8 @@ fate-nellymoser-aref-encode: $(AREF)
 fate-nellymoser-aref-encode: CMD = enc_dec_pcm flv wav s16le $(REF) -c:a nellymoser
 fate-nellymoser-aref-encode: CMP = stddev
 fate-nellymoser-aref-encode: REF = ./tests/data/acodec-16000-1.ref.wav
-fate-nellymoser-aref-encode: CMP_SHIFT = -1172
-fate-nellymoser-aref-encode: CMP_TARGET = 9617
+fate-nellymoser-aref-encode: CMP_SHIFT = -244
+fate-nellymoser-aref-encode: CMP_TARGET = 9612
 fate-nellymoser-aref-encode: SIZE_TOLERANCE = 268
 
 FATE_SAMPLES_AUDIO += fate-sierra-vmd-audio