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author | Steven Liu <lq@chinaffmpeg.org> | 2019-08-21 19:34:28 +0800 |
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committer | Steven Liu <lq@chinaffmpeg.org> | 2019-08-26 11:08:52 +0800 |
commit | 7c2ff06b6e6aee9ddb6f4e4bddb1fa5f2ccc1885 (patch) | |
tree | 5584c9547d20fe6fc6f635fc4227a0ee448d71f0 | |
parent | 17576fda6546afc818fe88efb62bb7a097c19e96 (diff) | |
download | ffmpeg-7c2ff06b6e6aee9ddb6f4e4bddb1fa5f2ccc1885.tar.gz |
doc/examples/decode_audio: print message about how to play the output file
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
-rw-r--r-- | doc/examples/decode_audio.c | 51 |
1 files changed, 51 insertions, 0 deletions
diff --git a/doc/examples/decode_audio.c b/doc/examples/decode_audio.c index 19dcafd2c8..6f589b9878 100644 --- 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) { @@ -172,6 +201,28 @@ 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 */ + enum AVSampleFormat sfmt = c->sample_fmt; + int n_channels = 0; + const char *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); |