diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2011-12-06 04:37:27 +0400 |
---|---|---|
committer | Michael Niedermayer <michaelni@gmx.at> | 2011-12-06 04:37:27 +0400 |
commit | b404ab9e74d3bca12d5989c366f5cfd746279067 (patch) | |
tree | fdbba6fdf7a4694fe7b7ecda6401ea6a2e01f95e /doc | |
parent | a448a5d1c4620aa58ec138fbffd46d18d42d53e0 (diff) | |
parent | 52401b82bd2ed30d4c4353cb084bf4ee679d0c22 (diff) |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
mov: Don't av_malloc(0).
avconv: only allocate 1 AVFrame per input stream
avconv: fix memleaks due to not freeing the AVFrame for audio
h264-fate: remove -strict 1 except where necessary (mr4/5-tandberg).
misc Doxygen markup improvements
doxygen: eliminate Qt-style doxygen syntax
g722: Add a regression test for muxing/demuxing in wav
g722: Change bits per sample to 4
g722dec: Signal skipping the lower bits via AVOptions instead of bits_per_coded_sample
api-example: update to use avcodec_decode_audio4()
avplay: use avcodec_decode_audio4()
avplay: use a separate buffer for playing silence
avformat: use avcodec_decode_audio4() in avformat_find_stream_info()
avconv: use avcodec_decode_audio4() instead of avcodec_decode_audio3()
mov: Allow empty stts atom.
doc: document preferred Doxygen syntax and make patcheck detect it
Conflicts:
avconv.c
ffplay.c
libavcodec/mlpdec.c
libavcodec/version.h
libavformat/mov.c
tests/codec-regression.sh
tests/fate/h264.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'doc')
-rw-r--r-- | doc/developer.texi | 5 | ||||
-rw-r--r-- | doc/examples/decoding_encoding.c | 29 |
2 files changed, 25 insertions, 9 deletions
diff --git a/doc/developer.texi b/doc/developer.texi index 2d3d418b97..75895d9147 100644 --- a/doc/developer.texi +++ b/doc/developer.texi @@ -77,6 +77,11 @@ Use the JavaDoc/Doxygen format (see examples below) so that code documentation can be generated automatically. All nontrivial functions should have a comment above them explaining what the function does, even if it is just one sentence. All structures and their member variables should be documented, too. + +Avoid Qt-style and similar Doxygen syntax with @code{!} in it, i.e. replace +@code{//!} with @code{///} and similar. Also @@ syntax should be employed +for markup commands, i.e. use @code{@@param} and not @code{\param}. + @example /** * @@file diff --git a/doc/examples/decoding_encoding.c b/doc/examples/decoding_encoding.c index ee0cb585f5..f87a8c9c41 100644 --- a/doc/examples/decoding_encoding.c +++ b/doc/examples/decoding_encoding.c @@ -33,6 +33,7 @@ #include "libavutil/opt.h" #include "libavcodec/avcodec.h" #include "libavutil/mathematics.h" +#include "libavutil/samplefmt.h" #define INBUF_SIZE 4096 #define AUDIO_INBUF_SIZE 20480 @@ -114,11 +115,11 @@ static void audio_decode_example(const char *outfilename, const char *filename) { AVCodec *codec; AVCodecContext *c= NULL; - int out_size, len; + int len; FILE *f, *outfile; - uint8_t *outbuf; uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE]; AVPacket avpkt; + AVFrame *decoded_frame = NULL; av_init_packet(&avpkt); @@ -139,8 +140,6 @@ static void audio_decode_example(const char *outfilename, const char *filename) exit(1); } - outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE); - f = fopen(filename, "rb"); if (!f) { fprintf(stderr, "could not open %s\n", filename); @@ -157,15 +156,27 @@ static void audio_decode_example(const char *outfilename, const char *filename) avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f); while (avpkt.size > 0) { - out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; - len = avcodec_decode_audio3(c, (short *)outbuf, &out_size, &avpkt); + int got_frame = 0; + + if (!decoded_frame) { + if (!(decoded_frame = avcodec_alloc_frame())) { + fprintf(stderr, "out of memory\n"); + exit(1); + } + } else + avcodec_get_frame_defaults(decoded_frame); + + len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt); if (len < 0) { fprintf(stderr, "Error while decoding\n"); exit(1); } - if (out_size > 0) { + if (got_frame) { /* if a frame has been decoded, output it */ - fwrite(outbuf, 1, out_size, outfile); + int data_size = av_samples_get_buffer_size(NULL, c->channels, + decoded_frame->nb_samples, + c->sample_fmt, 1); + fwrite(decoded_frame->data[0], 1, data_size, outfile); } avpkt.size -= len; avpkt.data += len; @@ -185,10 +196,10 @@ static void audio_decode_example(const char *outfilename, const char *filename) fclose(outfile); fclose(f); - free(outbuf); avcodec_close(c); av_free(c); + av_free(decoded_frame); } /* |