Message ID | 20190821113428.33142-1-lq@chinaffmpeg.org |
---|---|
State | New |
Headers | show |
On Wed, Aug 21, 2019 at 07:34:28PM +0800, Steven Liu wrote: > Signed-off-by: Steven Liu <lq@chinaffmpeg.org> > --- > doc/examples/decode_audio.c | 51 +++++++++++++++++++++++++++++++++++++ > 1 file changed, 51 insertions(+) > > diff --git a/doc/examples/decode_audio.c b/doc/examples/decode_audio.c > index 19dcafd2c8..23d0c9bf50 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 = c->channels; > + 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; duplicate code with line#53? > + 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); > > -- > 2.17.2 (Apple Git-113) > > > > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
diff --git a/doc/examples/decode_audio.c b/doc/examples/decode_audio.c index 19dcafd2c8..23d0c9bf50 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 = c->channels; + 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);
Signed-off-by: Steven Liu <lq@chinaffmpeg.org> --- doc/examples/decode_audio.c | 51 +++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+)