fix -a^b which was interpreted as (-a)^b

[ffmpeg] / libavformat / audio.c

diff --git a/libavformat/audio.c b/libavformat/audio.c
index e3e5888e9b6bff5b0d3e8e42ed90171e293c528d..1dfccccb8cebda2ebf711b037fe74ebde0ea690d 100644 (file)
--- a/libavformat/audio.c
+++ b/libavformat/audio.c
@@ -2,26 +2,32 @@
  * Linux audio play and grab interface
  * Copyright (c) 2000, 2001 Fabrice Bellard.
  *
- * This library is free software; you can redistribute it and/or
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This library is distributed in the hope that it will be useful,
+ * FFmpeg 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
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 #include "avformat.h"
 
 #include <stdlib.h>
 #include <stdio.h>
 #include <string.h>
+#ifdef __OpenBSD__
+#include <soundcard.h>
+#else
 #include <sys/soundcard.h>
+#endif
 #include <unistd.h>
 #include <fcntl.h>
 #include <sys/ioctl.h>
@@ -49,7 +55,11 @@ static int audio_open(AudioData *s, int is_output, const char *audio_device)
 
     /* open linux audio device */
     if (!audio_device)
+#ifdef __OpenBSD__
+        audio_device = "/dev/sound";
+#else
         audio_device = "/dev/dsp";
+#endif
 
     if (is_output)
         audio_fd = open(audio_device, O_WRONLY);
@@ -57,7 +67,7 @@ static int audio_open(AudioData *s, int is_output, const char *audio_device)
         audio_fd = open(audio_device, O_RDONLY);
     if (audio_fd < 0) {
         perror(audio_device);
-        return -EIO;
+        return AVERROR_IO;
     }
 
     if (flip && *flip == '1') {
@@ -79,7 +89,7 @@ static int audio_open(AudioData *s, int is_output, const char *audio_device)
 
     /* select format : favour native format */
     err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
-    
+
 #ifdef WORDS_BIGENDIAN
     if (tmp & AFMT_S16_BE) {
         tmp = AFMT_S16_BE;
@@ -108,14 +118,14 @@ static int audio_open(AudioData *s, int is_output, const char *audio_device)
     default:
         av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
         close(audio_fd);
-        return -EIO;
+        return AVERROR_IO;
     }
     err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
     if (err < 0) {
         perror("SNDCTL_DSP_SETFMT");
         goto fail;
     }
-    
+
     tmp = (s->channels == 2);
     err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
     if (err < 0) {
@@ -124,7 +134,7 @@ static int audio_open(AudioData *s, int is_output, const char *audio_device)
     }
     if (tmp)
         s->channels = 2;
-    
+
     tmp = s->sample_rate;
     err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
     if (err < 0) {
@@ -137,7 +147,7 @@ static int audio_open(AudioData *s, int is_output, const char *audio_device)
     return 0;
  fail:
     close(audio_fd);
-    return -EIO;
+    return AVERROR_IO;
 }
 
 static int audio_close(AudioData *s)
@@ -154,11 +164,11 @@ static int audio_write_header(AVFormatContext *s1)
     int ret;
 
     st = s1->streams[0];
-    s->sample_rate = st->codec.sample_rate;
-    s->channels = st->codec.channels;
+    s->sample_rate = st->codec->sample_rate;
+    s->channels = st->codec->channels;
     ret = audio_open(s, 1, NULL);
     if (ret < 0) {
-        return -EIO;
+        return AVERROR_IO;
     } else {
         return 0;
     }
@@ -183,7 +193,7 @@ static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
                 if (ret > 0)
                     break;
                 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
-                    return -EIO;
+                    return AVERROR_IO;
             }
             s->buffer_ptr = 0;
         }
@@ -209,7 +219,7 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
     AVStream *st;
     int ret;
 
-    if (!ap || ap->sample_rate <= 0 || ap->channels <= 0)
+    if (ap->sample_rate <= 0 || ap->channels <= 0)
         return -1;
 
     st = av_new_stream(s1, 0);
@@ -222,16 +232,16 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
     ret = audio_open(s, 0, ap->device);
     if (ret < 0) {
         av_free(st);
-        return -EIO;
+        return AVERROR_IO;
     }
 
     /* 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;
+    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;
 
-    av_set_pts_info(st, 48, 1, 1000000);  /* 48 bits pts in us */
+    av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
     return 0;
 }
 
@@ -241,9 +251,9 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
     int ret, bdelay;
     int64_t cur_time;
     struct audio_buf_info abufi;
-    
+
     if (av_new_packet(pkt, s->frame_size) < 0)
-        return -EIO;
+        return AVERROR_IO;
     for(;;) {
         struct timeval tv;
         fd_set fds;
@@ -263,11 +273,12 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
         if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
             av_free_packet(pkt);
             pkt->size = 0;
+            pkt->pts = av_gettime();
             return 0;
         }
         if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
             av_free_packet(pkt);
-            return -EIO;
+            return AVERROR_IO;
         }
     }
     pkt->size = ret;
@@ -282,7 +293,7 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
     cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
 
     /* convert to wanted units */
-    pkt->pts = cur_time & ((1LL << 48) - 1);
+    pkt->pts = cur_time;
 
     if (s->flip_left && s->channels == 2) {
         int i;
@@ -304,7 +315,8 @@ static int audio_read_close(AVFormatContext *s1)
     return 0;
 }
 
-static AVInputFormat audio_in_format = {
+#ifdef CONFIG_AUDIO_DEMUXER
+AVInputFormat audio_demuxer = {
     "audio_device",
     "audio grab and output",
     sizeof(AudioData),
@@ -314,8 +326,10 @@ static AVInputFormat audio_in_format = {
     audio_read_close,
     .flags = AVFMT_NOFILE,
 };
+#endif
 
-static AVOutputFormat audio_out_format = {
+#ifdef CONFIG_AUDIO_MUXER
+AVOutputFormat audio_muxer = {
     "audio_device",
     "audio grab and output",
     "",
@@ -335,10 +349,4 @@ static AVOutputFormat audio_out_format = {
     audio_write_trailer,
     .flags = AVFMT_NOFILE,
 };
-
-int audio_init(void)
-{
-    av_register_input_format(&audio_in_format);
-    av_register_output_format(&audio_out_format);
-    return 0;
-}
+#endif