BeOS Audio ouput patch by (François Revol <revol at free dot fr>)

Originally committed as revision 1163 to svn://svn.ffmpeg.org/ffmpeg/trunk

6 files changed, 346 insertions, 2 deletions

diff --git a/Makefile b/Makefile
index 4c5c739a0a..e49c061939 100644
--- a/Makefile
+++ b/Makefile
@@ -28,6 +28,9 @@ EXT=
 PROG=ffmpeg ffplay ffserver
 endif
 
+ifeq ($(CONFIG_AUDIO_BEOS),yes)
+EXTRALIBS+=-lmedia -lbe
+endif
 
 ifeq ($(BUILD_SHARED),yes)
 DEP_LIBS=libavcodec/libavcodec.so libav/libavformat.a
diff --git a/configure b/configure
index d4e6134f9d..b41fd4d2c1 100755
--- a/configure
+++ b/configure
@@ -51,6 +51,7 @@ esac
 gprof="no"
 v4l="yes"
 audio_oss="yes"
+audio_beos="no"
 network="yes"
 zlib="yes"
 mp3lame="no"
@@ -89,6 +90,8 @@ SHFLAGS=-nostart
 # disable linux things
 audio_oss="no"
 v4l="no"
+# enable beos things
+audio_beos="yes"
 # no need for libm, but the inet stuff
 # Check for BONE
 if (echo $BEINCLUDES|grep 'headers/be/bone' >/dev/null); then
@@ -186,6 +189,8 @@ for opt do
   ;;
   --disable-audio-oss) audio_oss="no"
   ;;
+  --disable-audio-beos) audio_beos="no"
+  ;;
   --disable-network) network="no"
   ;;
   --disable-zlib) zlib="no"
@@ -411,6 +416,7 @@ echo "  --cpu=CPU                force cpu to CPU  [$cpu]"
 echo "  --disable-mmx            disable mmx usage"
 echo "  --disable-altivec        disable AltiVec usage"
 echo "  --disable-audio-oss      disable OSS audio support [default=no]"
+echo "  --disable-audio-beos     disable BeOS audio support [default=no]"
 echo "  --disable-v4l            disable video4linux grabbing [default=no]"
 echo "  --disable-network        disable network support [default=no]"
 echo "  --disable-zlib           disable zlib [default=no]"
@@ -559,6 +565,11 @@ if test "$audio_oss" = "yes" ; then
   echo "CONFIG_AUDIO_OSS=yes" >> config.mak
 fi
 
+if test "$audio_beos" = "yes" ; then
+  echo "#define CONFIG_AUDIO_BEOS 1" >> $TMPH
+  echo "CONFIG_AUDIO_BEOS=yes" >> config.mak
+fi
+
 if test "$network" = "yes" ; then
   echo "#define CONFIG_NETWORK 1" >> $TMPH
   echo "CONFIG_NETWORK=yes" >> config.mak
diff --git a/ffmpeg.c b/ffmpeg.c
index a969340118..53bcc4781f 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -145,7 +145,7 @@ static int video_stream_copy = 0;
 
 #define DEFAULT_PASS_LOGFILENAME "ffmpeg2pass"
 
-#ifndef CONFIG_AUDIO_OSS
+#if !defined(CONFIG_AUDIO_OSS) && !defined(CONFIG_AUDIO_BEOS)
 const char *audio_device = "none";
 #endif
 #ifndef CONFIG_VIDEO4LINUX
diff --git a/libav/Makefile b/libav/Makefile
index e3a8ace27b..2889fd31f5 100644
--- a/libav/Makefile
+++ b/libav/Makefile
@@ -29,6 +29,10 @@ ifeq ($(CONFIG_AUDIO_OSS),yes)
 OBJS+= audio.o 
 endif
 
+ifeq ($(CONFIG_AUDIO_BEOS),yes)
+OBJS+= beosaudio.o
+endif
+
 ifeq ($(CONFIG_NETWORK),yes)
 OBJS+= udp.o tcp.o http.o rtsp.o rtp.o rtpproto.o
 # BeOS network stuff
@@ -61,5 +65,9 @@ installlib: all
 %.o: %.c
 	$(CC) $(CFLAGS) -c -o $@ $< 
 
+# BeOS: remove -Wall to get rid of all the "multibyte constant" warnings
+%.o: %.cpp
+	g++ $(subst -Wall,,$(CFLAGS)) -c -o $@ $< 
+
 clean: 
 	rm -f *.o *~ *.a 
diff --git a/libav/allformats.c b/libav/allformats.c
index e4102157b8..8a65638e88 100644
--- a/libav/allformats.c
+++ b/libav/allformats.c
@@ -56,7 +56,7 @@ void av_register_all(void)
 #ifdef CONFIG_VIDEO4LINUX
     video_grab_init();
 #endif
-#ifdef CONFIG_AUDIO_OSS
+#if defined(CONFIG_AUDIO_OSS) || defined(CONFIG_AUDIO_BEOS)
     audio_init();
 #endif
 
diff --git a/libav/beosaudio.cpp b/libav/beosaudio.cpp
new file mode 100644
index 0000000000..127454606e
--- /dev/null
+++ b/libav/beosaudio.cpp
@@ -0,0 +1,322 @@
+/*
+ * BeOS audio play interface
+ * Copyright (c) 2000, 2001 Fabrice Bellard.
+ *
+ * This library 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 of the License, or (at your option) any later version.
+ *
+ * This library 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 this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include <signal.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <sys/time.h>
+
+#include <Application.h>
+#include <SoundPlayer.h>
+
+extern "C" {
+#include "avformat.h"
+}
+
+
+//const char *audio_device = "/dev/dsp";
+const char *audio_device = "beosaudio:";
+
+#define AUDIO_BLOCK_SIZE 4096
+
+typedef struct {
+    int fd;
+    int sample_rate;
+    int channels;
+    int frame_size; /* in bytes ! */
+    CodecID codec_id;
+    int flip_left : 1;
+    UINT8 buffer[AUDIO_BLOCK_SIZE];
+    int buffer_ptr;
+    int pipefd; /* the other end of the pipe */
+    BSoundPlayer *player;
+    int has_quit; /* signal callbacks not to wait */
+} AudioData;
+
+static int own_BApp_created = 0;
+
+/* create the BApplication and Run() it */
+static int32 bapp_thread(void *arg)
+{
+    new BApplication("application/x-vnd.ffmpeg");
+    own_BApp_created = 1;
+    be_app->Run();
+    /* kill the process group */
+    kill(0, SIGINT);
+    return B_OK;
+}
+
+/* called back by BSoundPlayer */
+static void audioplay_callback(void *cookie, void *buffer, size_t bufferSize, const media_raw_audio_format &format)
+{
+    AudioData *s;
+    size_t amount;
+    unsigned char *buf = (unsigned char *)buffer;
+
+    s = (AudioData *)cookie;
+    if (s->has_quit)
+        return;
+    while (bufferSize > 0) {
+        if ((amount = read(s->pipefd, buf, bufferSize)) < B_OK) {
+            puts("EPIPE");
+            snooze(100000);
+            s->player->SetHasData(false);
+            return;
+        }
+        if (amount == 0) {
+            snooze(100000);
+            s->player->SetHasData(false);
+            return;
+        }
+        buf += amount;
+        bufferSize -= amount;
+    }
+}
+
+static int audio_open(AudioData *s, int is_output)
+{
+    int p[2];
+    int ret;
+    media_raw_audio_format format;
+
+    if (!is_output)
+        return -EIO; /* not for now */
+    ret = pipe(p);
+    if (ret < 0)
+        return -EIO;
+    s->fd = p[is_output?1:0];
+    s->pipefd = p[is_output?0:1];
+    if (s->fd < 0) {
+        perror(is_output?"audio out":"audio in");
+        return -EIO;
+    }
+    /* non blocking mode */
+//    fcntl(s->fd, F_SETFL, O_NONBLOCK);
+//    fcntl(s->pipefd, F_SETFL, O_NONBLOCK);
+    s->frame_size = AUDIO_BLOCK_SIZE;
+    format = media_raw_audio_format::wildcard;
+    format.format = media_raw_audio_format::B_AUDIO_SHORT;
+    format.byte_order = B_HOST_IS_LENDIAN ? B_MEDIA_LITTLE_ENDIAN : B_MEDIA_BIG_ENDIAN;
+    format.channel_count = s->channels;
+    format.buffer_size = s->frame_size;
+    format.frame_rate = s->sample_rate;
+    s->player = new BSoundPlayer(&format, "ffmpeg output", audioplay_callback);
+    if (s->player->InitCheck() != B_OK) {
+        delete s->player;
+        s->player = NULL;
+        close(s->fd);
+        close(s->pipefd);
+        return -EIO;
+    }
+    s->player->SetCookie(s);
+    s->player->SetVolume(1.0);
+    s->player->Start();
+    s->player->SetHasData(true);
+    return 0;
+}
+
+static int audio_close(AudioData *s)
+{
+    s->has_quit = 1;
+    if (s->player) {
+        s->player->Stop();
+    }
+    if (s->player)
+        delete s->player;
+    close(s->pipefd);
+    close(s->fd);
+    return 0;
+}
+
+/* sound output support */
+static int audio_write_header(AVFormatContext *s1)
+{
+    AudioData *s = (AudioData *)s1->priv_data;
+    AVStream *st;
+    int ret;
+
+    st = s1->streams[0];
+    s->sample_rate = st->codec.sample_rate;
+    s->channels = st->codec.channels;
+    ret = audio_open(s, 1);
+    if (ret < 0)
+        return -EIO;
+    return 0;
+}
+
+static int audio_write_packet(AVFormatContext *s1, int stream_index,
+                              UINT8 *buf, int size, int force_pts)
+{
+    AudioData *s = (AudioData *)s1->priv_data;
+    int len, ret;
+
+    while (size > 0) {
+        len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
+        if (len > size)
+            len = size;
+        memcpy(s->buffer + s->buffer_ptr, buf, len);
+        s->buffer_ptr += len;
+        if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
+            for(;;) {
+snooze(1000);
+                ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
+                if (ret != 0)
+                    break;
+                if (ret < 0 && (errno != EAGAIN && errno != EINTR))
+                    return -EIO;
+            }
+            s->buffer_ptr = 0;
+        }
+        buf += len;
+        size -= len;
+    }
+    return 0;
+}
+
+static int audio_write_trailer(AVFormatContext *s1)
+{
+    AudioData *s = (AudioData *)s1->priv_data;
+
+    audio_close(s);
+    return 0;
+}
+
+/* grab support */
+
+static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
+{
+    AudioData *s = (AudioData *)s1->priv_data;
+    AVStream *st;
+    int ret;
+
+    if (!ap || ap->sample_rate <= 0 || ap->channels <= 0)
+        return -1;
+
+    st = av_new_stream(s1, 0);
+    if (!st) {
+        return -ENOMEM;
+    }
+    s->sample_rate = ap->sample_rate;
+    s->channels = ap->channels;
+
+    ret = audio_open(s, 0);
+    if (ret < 0) {
+        av_free(st);
+        return -EIO;
+    } else {
+        /* take real parameters */
+        st->codec.codec_type = CODEC_TYPE_AUDIO;
+        st->codec.codec_id = s->codec_id;
+        st->codec.sample_rate = s->sample_rate;
+        st->codec.channels = s->channels;
+        return 0;
+    }
+}
+
+static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
+{
+    AudioData *s = (AudioData *)s1->priv_data;
+    int ret;
+
+    if (av_new_packet(pkt, s->frame_size) < 0)
+        return -EIO;
+    for(;;) {
+        ret = read(s->fd, pkt->data, pkt->size);
+        if (ret > 0)
+            break;
+        if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
+            av_free_packet(pkt);
+            pkt->size = 0;
+            return 0;
+        }
+        if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
+            av_free_packet(pkt);
+            return -EIO;
+        }
+    }
+    pkt->size = ret;
+    if (s->flip_left && s->channels == 2) {
+        int i;
+        short *p = (short *) pkt->data;
+
+        for (i = 0; i < ret; i += 4) {
+            *p = ~*p;
+            p += 2;
+        }
+    }
+    return 0;
+}
+
+static int audio_read_close(AVFormatContext *s1)
+{
+    AudioData *s = (AudioData *)s1->priv_data;
+
+    audio_close(s);
+    return 0;
+}
+
+AVInputFormat audio_in_format = {
+    "audio_device",
+    "audio grab and output",
+    sizeof(AudioData),
+    NULL,
+    audio_read_header,
+    audio_read_packet,
+    audio_read_close,
+    NULL,
+    AVFMT_NOFILE,
+};
+
+AVOutputFormat audio_out_format = {
+    "audio_device",
+    "audio grab and output",
+    "",
+    "",
+    sizeof(AudioData),
+#ifdef WORDS_BIGENDIAN
+    CODEC_ID_PCM_S16BE,
+#else
+    CODEC_ID_PCM_S16LE,
+#endif
+    CODEC_ID_NONE,
+    audio_write_header,
+    audio_write_packet,
+    audio_write_trailer,
+    AVFMT_NOFILE,
+};
+
+extern "C" {
+
+int audio_init(void)
+{
+    /* needed by libmedia */
+    if (be_app == NULL) {
+        resume_thread(spawn_thread(bapp_thread, "ffmpeg BApplication", B_NORMAL_PRIORITY, NULL));
+        while (!own_BApp_created)
+            snooze(50000);
+    }
+    av_register_input_format(&audio_in_format);
+    av_register_output_format(&audio_out_format);
+    return 0;
+}
+
+} // "C"
+