avcodec: Switch AVCPBProperties to 64bits

[ffmpeg] / doc / examples / decode_audio.c

diff --git a/doc/examples/decode_audio.c b/doc/examples/decode_audio.c
index 19dcafd2c82c3ecd40837a75776b24faf55312b6..6c2a8ed5508a3c2a2b43ae4ced3b654943313cf3 100644 (file)
--- a/doc/examples/decode_audio.c
+++ b/doc/examples/decode_audio.c
@@ -39,6 +39,35 @@
 #define AUDIO_INBUF_SIZE 20480
 #define AUDIO_REFILL_THRESH 4096
 
+static int get_format_from_sample_fmt(const char **fmt,
+                                      enum AVSampleFormat sample_fmt)
+{
+    int i;
+    struct sample_fmt_entry {
+        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
+    } sample_fmt_entries[] = {
+        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
+        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
+        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
+        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
+        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
+    };
+    *fmt = NULL;
+
+    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
+        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
+        if (sample_fmt == entry->sample_fmt) {
+            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
+            return 0;
+        }
+    }
+
+    fprintf(stderr,
+            "sample format %s is not supported as output format\n",
+            av_get_sample_fmt_name(sample_fmt));
+    return -1;
+}
+
 static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
                    FILE *outfile)
 {
@@ -86,6 +115,9 @@ int main(int argc, char **argv)
     size_t   data_size;
     AVPacket *pkt;
     AVFrame *decoded_frame = NULL;
+    enum AVSampleFormat sfmt;
+    int n_channels = 0;
+    const char *fmt;
 
     if (argc <= 2) {
         fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
@@ -172,6 +204,26 @@ int main(int argc, char **argv)
     pkt->size = 0;
     decode(c, pkt, decoded_frame, outfile);
 
+    /* print output pcm infomations, because there have no metadata of pcm */
+    sfmt = c->sample_fmt;
+
+    if (av_sample_fmt_is_planar(sfmt)) {
+        const char *packed = av_get_sample_fmt_name(sfmt);
+        printf("Warning: the sample format the decoder produced is planar "
+               "(%s). This example will output the first channel only.\n",
+               packed ? packed : "?");
+        sfmt = av_get_packed_sample_fmt(sfmt);
+    }
+
+    n_channels = c->channels;
+    if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
+        goto end;
+
+    printf("Play the output audio file with the command:\n"
+           "ffplay -f %s -ac %d -ar %d %s\n",
+           fmt, n_channels, c->sample_rate,
+           outfilename);
+end:
     fclose(outfile);
     fclose(f);