diff options
Diffstat (limited to 'doc')
| -rw-r--r-- | doc/examples/decoding_encoding.c | 136 | ||||
| -rw-r--r-- | doc/platform.texi | 4 | ||||
| -rw-r--r-- | doc/protocols.texi | 5 |
3 files changed, 126 insertions, 19 deletions
diff --git a/doc/examples/decoding_encoding.c b/doc/examples/decoding_encoding.c index 3576cc0..6295001 100644 --- a/doc/examples/decoding_encoding.c +++ b/doc/examples/decoding_encoding.c @@ -34,6 +34,7 @@ #include <libavutil/imgutils.h> #include <libavutil/opt.h> #include <libavcodec/avcodec.h> +#include <libavutil/audioconvert.h> #include <libavutil/mathematics.h> #include <libavutil/samplefmt.h> @@ -41,6 +42,59 @@ #define AUDIO_INBUF_SIZE 20480 #define AUDIO_REFILL_THRESH 4096 +/* check that a given sample format is supported by the encoder */ +static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt) +{ + const enum AVSampleFormat *p = codec->sample_fmts; + + while (*p != AV_SAMPLE_FMT_NONE) { + if (*p == sample_fmt) + return 1; + p++; + } + return 0; +} + +/* just pick the highest supported samplerate */ +static int select_sample_rate(AVCodec *codec) +{ + const int *p; + int best_samplerate = 0; + + if (!codec->supported_samplerates) + return 44100; + + p = codec->supported_samplerates; + while (*p) { + best_samplerate = FFMAX(*p, best_samplerate); + p++; + } + return best_samplerate; +} + +/* select layout with the highest channel count */ +static int select_channel_layout(AVCodec *codec) +{ + const uint64_t *p; + uint64_t best_ch_layout = 0; + int best_nb_channells = 0; + + if (!codec->channel_layouts) + return AV_CH_LAYOUT_STEREO; + + p = codec->channel_layouts; + while (*p) { + int nb_channels = av_get_channel_layout_nb_channels(*p); + + if (nb_channels > best_nb_channells) { + best_ch_layout = *p; + best_nb_channells = nb_channels; + } + p++; + } + return best_ch_layout; +} + /* * Audio encoding example */ @@ -48,11 +102,13 @@ static void audio_encode_example(const char *filename) { AVCodec *codec; AVCodecContext *c= NULL; - int frame_size, i, j, out_size, outbuf_size; + AVFrame *frame; + AVPacket pkt; + int i, j, k, ret, got_output; + int buffer_size; FILE *f; - short *samples; + uint16_t *samples; float t, tincr; - uint8_t *outbuf; printf("Encode audio file %s\n", filename); @@ -67,9 +123,19 @@ static void audio_encode_example(const char *filename) /* put sample parameters */ c->bit_rate = 64000; - c->sample_rate = 44100; - c->channels = 2; + + /* check that the encoder supports s16 pcm input */ c->sample_fmt = AV_SAMPLE_FMT_S16; + if (!check_sample_fmt(codec, c->sample_fmt)) { + fprintf(stderr, "encoder does not support %s", + av_get_sample_fmt_name(c->sample_fmt)); + exit(1); + } + + /* select other audio parameters supported by the encoder */ + c->sample_rate = select_sample_rate(codec); + c->channel_layout = select_channel_layout(codec); + c->channels = av_get_channel_layout_nb_channels(c->channel_layout); /* open it */ if (avcodec_open2(c, codec, NULL) < 0) { @@ -77,35 +143,71 @@ static void audio_encode_example(const char *filename) exit(1); } - /* the codec gives us the frame size, in samples */ - frame_size = c->frame_size; - samples = malloc(frame_size * 2 * c->channels); - outbuf_size = 10000; - outbuf = malloc(outbuf_size); - f = fopen(filename, "wb"); if (!f) { fprintf(stderr, "could not open %s\n", filename); exit(1); } + /* frame containing input raw audio */ + frame = avcodec_alloc_frame(); + if (!frame) { + fprintf(stderr, "could not allocate audio frame\n"); + exit(1); + } + + frame->nb_samples = c->frame_size; + frame->format = c->sample_fmt; + frame->channel_layout = c->channel_layout; + + /* the codec gives us the frame size, in samples, + * we calculate the size of the samples buffer in bytes */ + buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, + c->sample_fmt, 0); + samples = av_malloc(buffer_size); + if (!samples) { + fprintf(stderr, "could not allocate %d bytes for samples buffer\n", + buffer_size); + exit(1); + } + /* setup the data pointers in the AVFrame */ + ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, + (const uint8_t*)samples, buffer_size, 0); + if (ret < 0) { + fprintf(stderr, "could not setup audio frame\n"); + exit(1); + } + /* encode a single tone sound */ t = 0; tincr = 2 * M_PI * 440.0 / c->sample_rate; for(i=0;i<200;i++) { - for(j=0;j<frame_size;j++) { + av_init_packet(&pkt); + pkt.data = NULL; // packet data will be allocated by the encoder + pkt.size = 0; + + for (j = 0; j < c->frame_size; j++) { samples[2*j] = (int)(sin(t) * 10000); - samples[2*j+1] = samples[2*j]; + + for (k = 1; k < c->channels; k++) + samples[2*j + k] = samples[2*j]; t += tincr; } /* encode the samples */ - out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples); - fwrite(outbuf, 1, out_size, f); + ret = avcodec_encode_audio2(c, &pkt, frame, &got_output); + if (ret < 0) { + fprintf(stderr, "error encoding audio frame\n"); + exit(1); + } + if (got_output) { + fwrite(pkt.data, 1, pkt.size, f); + av_free_packet(&pkt); + } } fclose(f); - free(outbuf); - free(samples); + av_freep(&samples); + av_freep(&frame); avcodec_close(c); av_free(c); } diff --git a/doc/platform.texi b/doc/platform.texi index dcbb0f6..332b6b1 100644 --- a/doc/platform.texi +++ b/doc/platform.texi @@ -318,9 +318,9 @@ following "Devel" ones: binutils, gcc4-core, make, git, mingw-runtime, texi2html @end example -And the following "Utils" one: +In order to run FATE you will also need the following "Utils" packages: @example -diffutils +bc, diffutils @end example Then run diff --git a/doc/protocols.texi b/doc/protocols.texi index 4833cd3..3a82c7f 100644 --- a/doc/protocols.texi +++ b/doc/protocols.texi @@ -267,6 +267,11 @@ value will be sent. Stream identifier to play or to publish. This option overrides the parameter specified in the URI. +@item rtmp_subscribe +Name of live stream to subscribe to. By default no value will be sent. +It is only sent if the option is specified or if rtmp_live +is set to live. + @item rtmp_swfurl URL of the SWF player for the media. By default no value will be sent. |
