* Ogg/Vorbis patch by Mark Hills

[ffmpeg] / libav / audio.c

diff --git a/libav/audio.c b/libav/audio.c
index b32a3bc95787362f3140f14d284dcb0617c28c75..4fa155c85de4c54df410d3a69ea50b6bd0650f67 100644 (file)
--- a/libav/audio.c
+++ b/libav/audio.c
@@ -21,7 +21,7 @@
 #include <stdlib.h>
 #include <stdio.h>
 #include <string.h>
-#include <linux/soundcard.h>
+#include <sys/soundcard.h>
 #include <unistd.h>
 #include <fcntl.h>
 #include <sys/ioctl.h>
@@ -64,7 +64,8 @@ static int audio_open(AudioData *s, int is_output)
     }
 
     /* non blocking mode */
-    fcntl(audio_fd, F_SETFL, O_NONBLOCK);
+    if (!is_output)
+        fcntl(audio_fd, F_SETFL, O_NONBLOCK);
 
     s->frame_size = AUDIO_BLOCK_SIZE;
 #if 0
@@ -177,7 +178,7 @@ static int audio_write_packet(AVFormatContext *s1, int stream_index,
         if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
             for(;;) {
                 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
-                if (ret != 0)
+                if (ret > 0)
                     break;
                 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
                     return -EIO;
@@ -220,21 +221,25 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
     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;
     }
+
+    /* 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;
+
+    av_set_pts_info(s1, 48, 1, 1000000);  /* 48 bits pts in us */
+    return 0;
 }
 
 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
 {
     AudioData *s = s1->priv_data;
-    int ret;
-
+    int ret, bdelay;
+    int64_t cur_time;
+    struct audio_buf_info abufi;
+    
     if (av_new_packet(pkt, s->frame_size) < 0)
         return -EIO;
     for(;;) {
@@ -252,6 +257,19 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
         }
     }
     pkt->size = ret;
+
+    /* compute pts of the start of the packet */
+    cur_time = av_gettime();
+    bdelay = ret;
+    if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
+        bdelay += abufi.bytes;
+    }
+    /* substract time represented by the number of bytes in the audio fifo */
+    cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
+
+    /* convert to wanted units */
+    pkt->pts = cur_time & ((1LL << 48) - 1);
+
     if (s->flip_left && s->channels == 2) {
         int i;
         short *p = (short *) pkt->data;
@@ -272,7 +290,7 @@ static int audio_read_close(AVFormatContext *s1)
     return 0;
 }
 
-AVInputFormat audio_in_format = {
+static AVInputFormat audio_in_format = {
     "audio_device",
     "audio grab and output",
     sizeof(AudioData),
@@ -280,10 +298,10 @@ AVInputFormat audio_in_format = {
     audio_read_header,
     audio_read_packet,
     audio_read_close,
-    flags: AVFMT_NOFILE,
+    .flags = AVFMT_NOFILE,
 };
 
-AVOutputFormat audio_out_format = {
+static AVOutputFormat audio_out_format = {
     "audio_device",
     "audio grab and output",
     "",
@@ -301,7 +319,7 @@ AVOutputFormat audio_out_format = {
     audio_write_header,
     audio_write_packet,
     audio_write_trailer,
-    flags: AVFMT_NOFILE,
+    .flags = AVFMT_NOFILE,
 };
 
 int audio_init(void)