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Diffstat (limited to 'doc/examples/filtering_audio.c')
| -rw-r--r-- | doc/examples/filtering_audio.c | 280 |
1 files changed, 280 insertions, 0 deletions
diff --git a/doc/examples/filtering_audio.c b/doc/examples/filtering_audio.c new file mode 100644 index 0000000000..46595fb3b8 --- /dev/null +++ b/doc/examples/filtering_audio.c @@ -0,0 +1,280 @@ +/* + * Copyright (c) 2010 Nicolas George + * Copyright (c) 2011 Stefano Sabatini + * Copyright (c) 2012 Clément Bœsch + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * API example for audio decoding and filtering + * @example filtering_audio.c + */ + +#include <unistd.h> + +#include <libavcodec/avcodec.h> +#include <libavformat/avformat.h> +#include <libavfilter/avfiltergraph.h> +#include <libavfilter/avcodec.h> +#include <libavfilter/buffersink.h> +#include <libavfilter/buffersrc.h> +#include <libavutil/opt.h> + +static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono"; +static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -"; + +static AVFormatContext *fmt_ctx; +static AVCodecContext *dec_ctx; +AVFilterContext *buffersink_ctx; +AVFilterContext *buffersrc_ctx; +AVFilterGraph *filter_graph; +static int audio_stream_index = -1; + +static int open_input_file(const char *filename) +{ + int ret; + AVCodec *dec; + + if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n"); + return ret; + } + + if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n"); + return ret; + } + + /* select the audio stream */ + ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n"); + return ret; + } + audio_stream_index = ret; + dec_ctx = fmt_ctx->streams[audio_stream_index]->codec; + av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0); + + /* init the audio decoder */ + if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n"); + return ret; + } + + return 0; +} + +static int init_filters(const char *filters_descr) +{ + char args[512]; + int ret = 0; + AVFilter *abuffersrc = avfilter_get_by_name("abuffer"); + AVFilter *abuffersink = avfilter_get_by_name("abuffersink"); + AVFilterInOut *outputs = avfilter_inout_alloc(); + AVFilterInOut *inputs = avfilter_inout_alloc(); + static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 }; + static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 }; + static const int out_sample_rates[] = { 8000, -1 }; + const AVFilterLink *outlink; + AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base; + + filter_graph = avfilter_graph_alloc(); + if (!outputs || !inputs || !filter_graph) { + ret = AVERROR(ENOMEM); + goto end; + } + + /* buffer audio source: the decoded frames from the decoder will be inserted here. */ + if (!dec_ctx->channel_layout) + dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels); + snprintf(args, sizeof(args), + "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64, + time_base.num, time_base.den, dec_ctx->sample_rate, + av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout); + ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in", + args, NULL, filter_graph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n"); + goto end; + } + + /* buffer audio sink: to terminate the filter chain. */ + ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out", + NULL, NULL, filter_graph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n"); + goto end; + } + + ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1, + AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n"); + goto end; + } + + ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1, + AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n"); + goto end; + } + + ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1, + AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n"); + goto end; + } + + /* Endpoints for the filter graph. */ + outputs->name = av_strdup("in"); + outputs->filter_ctx = buffersrc_ctx; + outputs->pad_idx = 0; + outputs->next = NULL; + + inputs->name = av_strdup("out"); + inputs->filter_ctx = buffersink_ctx; + inputs->pad_idx = 0; + inputs->next = NULL; + + if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr, + &inputs, &outputs, NULL)) < 0) + goto end; + + if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0) + goto end; + + /* Print summary of the sink buffer + * Note: args buffer is reused to store channel layout string */ + outlink = buffersink_ctx->inputs[0]; + av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout); + av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n", + (int)outlink->sample_rate, + (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"), + args); + +end: + avfilter_inout_free(&inputs); + avfilter_inout_free(&outputs); + + return ret; +} + +static void print_frame(const AVFrame *frame) +{ + const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame)); + const uint16_t *p = (uint16_t*)frame->data[0]; + const uint16_t *p_end = p + n; + + while (p < p_end) { + fputc(*p & 0xff, stdout); + fputc(*p>>8 & 0xff, stdout); + p++; + } + fflush(stdout); +} + +int main(int argc, char **argv) +{ + int ret; + AVPacket packet0, packet; + AVFrame *frame = av_frame_alloc(); + AVFrame *filt_frame = av_frame_alloc(); + int got_frame; + + if (!frame || !filt_frame) { + perror("Could not allocate frame"); + exit(1); + } + if (argc != 2) { + fprintf(stderr, "Usage: %s file | %s\n", argv[0], player); + exit(1); + } + + av_register_all(); + avfilter_register_all(); + + if ((ret = open_input_file(argv[1])) < 0) + goto end; + if ((ret = init_filters(filter_descr)) < 0) + goto end; + + /* read all packets */ + packet0.data = NULL; + packet.data = NULL; + while (1) { + if (!packet0.data) { + if ((ret = av_read_frame(fmt_ctx, &packet)) < 0) + break; + packet0 = packet; + } + + if (packet.stream_index == audio_stream_index) { + got_frame = 0; + ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n"); + continue; + } + packet.size -= ret; + packet.data += ret; + + if (got_frame) { + /* push the audio data from decoded frame into the filtergraph */ + if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) { + av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n"); + break; + } + + /* pull filtered audio from the filtergraph */ + while (1) { + ret = av_buffersink_get_frame(buffersink_ctx, filt_frame); + if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) + break; + if (ret < 0) + goto end; + print_frame(filt_frame); + av_frame_unref(filt_frame); + } + } + + if (packet.size <= 0) + av_free_packet(&packet0); + } else { + /* discard non-wanted packets */ + av_free_packet(&packet0); + } + } +end: + avfilter_graph_free(&filter_graph); + avcodec_close(dec_ctx); + avformat_close_input(&fmt_ctx); + av_frame_free(&frame); + av_frame_free(&filt_frame); + + if (ret < 0 && ret != AVERROR_EOF) { + fprintf(stderr, "Error occurred: %s\n", av_err2str(ret)); + exit(1); + } + + exit(0); +} |