Merge remote-tracking branch 'qatar/master'

* qatar/master:
  lavc: remove "legacy" mpegvideo decoder.
  iv8: assemble packets to return complete frames
  pulse: documentation
  pulse: introduce pulseaudio input
  remove the zork pcm seek test

Conflicts:
	configure
	libavdevice/Makefile
	libavdevice/alldevices.c
	libavdevice/avdevice.h
	libavdevice/pulse.c

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

doc/indevs.texi

@@ -390,6 +390,89 @@ ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
 For more information about OSS see:
 @url{http://manuals.opensound.com/usersguide/dsp.html}
 
+@section pulse
+
+pulseaudio input device.
+
+To enable this input device during configuration you need libpulse-simple
+installed in your system.
+
+The filename to provide to the input device is a source device or the
+string "default"
+
+To list the pulse source devices and their properties you can invoke
+the command @file{pactl list sources}.
+
+@example
+avconv -f pulse -i default /tmp/pulse.wav
+@end example
+
+@subsection @var{server} AVOption
+
+The syntax is:
+@example
+-server @var{server name}
+@end example
+
+Connects to a specific server.
+
+@subsection @var{name} AVOption
+
+The syntax is:
+@example
+-name @var{application name}
+@end example
+
+Specify the application name pulse will use when showing active clients,
+by default it is "libav"
+
+@subsection @var{stream_name} AVOption
+
+The syntax is:
+@example
+-stream_name @var{stream name}
+@end example
+
+Specify the stream name pulse will use when showing active streams,
+by default it is "record"
+
+@subsection @var{sample_rate} AVOption
+
+The syntax is:
+@example
+-sample_rate @var{samplerate}
+@end example
+
+Specify the samplerate in Hz, by default 48kHz is used.
+
+@subsection @var{channels} AVOption
+
+The syntax is:
+@example
+-channels @var{N}
+@end example
+
+Specify the channels in use, by default 2 (stereo) is set.
+
+@subsection @var{frame_size} AVOption
+
+The syntax is:
+@example
+-frame_size @var{bytes}
+@end example
+
+Specify the number of byte per frame, by default it is set to 1024.
+
+@subsection @var{fragment_size} AVOption
+
+The syntax is:
+@example
+-fragment_size @var{bytes}
+@end example
+
+Specify the minimal buffering fragment in pulseaudio, it will affect the
+audio latency. By default it is unset.
+
 @section sndio
 
 sndio input device.

libavdevice/pulse.c

@@ -21,11 +21,9 @@
 
 /**
  * @file
- * Pulseaudio input
+ * PulseAudio input using the simple API.
  * @author Luca Barbato <lu_zero@gentoo.org>
  *
- * This avdevice decoder allows to capture audio from a Pulseaudio device using
- * the simple api.
  */
 
 #include <pulse/simple.h>
@@ -95,9 +93,9 @@ static av_cold int pulse_read_header(AVFormatContext *s,
         device = s->filename;
 
     pd->s = pa_simple_new(pd->server, pd->name,
-                      PA_STREAM_RECORD,
+                          PA_STREAM_RECORD,
-                      device, pd->stream_name, &ss,
+                          device, pd->stream_name, &ss,
-                      NULL, &attr, &ret);
+                          NULL, &attr, &ret);
 
     if (!pd->s) {
         av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
@@ -122,7 +120,7 @@ static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
     int res;
     pa_usec_t latency;
     uint64_t frame_duration =
-        (pd->frame_size*1000000LL)/(pd->sample_rate * pd->channels);
+        (pd->frame_size*1000000LL) / (pd->sample_rate * pd->channels);
 
     if (av_new_packet(pkt, pd->frame_size) < 0) {
         return AVERROR(ENOMEM);
@@ -145,10 +143,10 @@ static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
         pd->pts = -latency;
     }
 
-    pd->pts += frame_duration;
-
     pkt->pts = pd->pts;
 
+    pd->pts += frame_duration;
+
     return 0;
 }
 
@@ -163,20 +161,13 @@ static av_cold int pulse_close(AVFormatContext *s)
 #define D AV_OPT_FLAG_DECODING_PARAM
 
 static const AVOption options[] = {
-    { "server",        "pulse server name",
+    { "server",        "pulse server name",                              OFFSET(server),        AV_OPT_TYPE_STRING, {.str = NULL},     0, 0, D },
-        OFFSET(server),        AV_OPT_TYPE_STRING, {.str = NULL},     0, 0, D },
+    { "name",          "application name",                               OFFSET(name),          AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT},  0, 0, D },
-    { "name",          "application name",
+    { "stream_name",   "stream description",                             OFFSET(stream_name),   AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
-        OFFSET(name),          AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT},  0, 0, D },
+    { "sample_rate",   "sample rate in Hz",                              OFFSET(sample_rate),   AV_OPT_TYPE_INT,    {.dbl = 48000},    1, INT_MAX, D },
-    { "stream_name",   "stream description",
+    { "channels",      "number of audio channels",                       OFFSET(channels),      AV_OPT_TYPE_INT,    {.dbl = 2},        1, INT_MAX, D },
-        OFFSET(stream_name),   AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
+    { "frame_size",    "number of bytes per frame",                      OFFSET(frame_size),    AV_OPT_TYPE_INT,    {.dbl = 1024},     1, INT_MAX, D },
-    { "sample_rate",   "",
+    { "fragment_size", "buffering size, affects latency and cpu usage",  OFFSET(fragment_size), AV_OPT_TYPE_INT,    {.dbl = -1},      -1, INT_MAX, D },
-        OFFSET(sample_rate),   AV_OPT_TYPE_INT,    {.dbl = 48000},    1, INT_MAX, D },
-    { "channels",      "",
-        OFFSET(channels),      AV_OPT_TYPE_INT,    {.dbl = 2},        1, INT_MAX, D },
-    { "frame_size",    "",
-        OFFSET(frame_size),    AV_OPT_TYPE_INT,    {.dbl = 1024},     1, INT_MAX, D },
-    { "fragment_size", "buffering size, affects latency and cpu usage",
-        OFFSET(fragment_size), AV_OPT_TYPE_INT,    {.dbl = -1},      -1, INT_MAX, D },
     { NULL },
 };
 

libavformat/iv8.c

@@ -55,33 +55,56 @@ static int read_header(AVFormatContext *s, AVFormatParameters *ap)
 
 static int read_packet(AVFormatContext *s, AVPacket *pkt)
 {
-    int ret, size, pts, type;
+    int ret, size, pts, type, flags;
-retry:
+    int first_pkt      = 0;
-    type= avio_rb16(s->pb); // 257 or 258
+    int frame_complete = 0;
-    size= avio_rb16(s->pb);
+
-
+    while (!frame_complete) {
-    avio_rb16(s->pb); //some flags, 0x80 indicates end of frame
+
-    avio_rb16(s->pb); //packet number
+        type  = avio_rb16(s->pb); // 257 or 258
-    pts=avio_rb32(s->pb);
+        size  = avio_rb16(s->pb);
-    avio_rb32(s->pb); //6A 13 E3 88
+        flags = avio_rb16(s->pb); //some flags, 0x80 indicates end of frame
-
+                avio_rb16(s->pb); //packet number
-    size -= 12;
+        pts   = avio_rb32(s->pb);
-    if(size<1)
+                avio_rb32(s->pb); //6A 13 E3 88
-        return -1;
+
-
+        frame_complete = flags & 0x80;
-    if(type==258){
+
-        avio_skip(s->pb, size);
+        size -= 12;
-        goto retry;
+        if (size < 1)
+            return -1;
+
+        if (type == 258) {
+            avio_skip(s->pb, size);
+            frame_complete = 0;
+            continue;
+        }
+
+        if (!first_pkt) {
+            ret = av_get_packet(s->pb, pkt, size);
+            if (ret < 0)
+                return ret;
+            first_pkt = 1;
+            pkt->pts  = pts;
+            pkt->pos -= 16;
+        } else {
+            ret = av_append_packet(s->pb, pkt, size);
+            if (ret < 0) {
+                av_log(s, AV_LOG_ERROR, "failed to grow packet\n");
+                av_free_packet(pkt);
+                return ret;
+            }
+        }
+        if (ret < size) {
+            av_log(s, AV_LOG_ERROR, "Truncated packet! Read %d of %d bytes\n",
+                   ret, size);
+            pkt->flags |= AV_PKT_FLAG_CORRUPT;
+            break;
+        }
     }
-
-    ret= av_get_packet(s->pb, pkt, size);
-
-    pkt->pts= pts;
-    pkt->pos-=16;
-
     pkt->stream_index = 0;
 
-    return ret;
+    return 0;
 }
 
 AVInputFormat ff_iv8_demuxer = {