ffmpeg version N-82997-g557c0df Copyright (c) 2000-2017 the FFmpeg developers

  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609

  configuration: --enable-libx264 --enable-gpl --enable-decoder=h264 --enable-encoder=libx264 --enable-shared --enable-static --disable-yasm --enable-nonfree --enable-libfdk-aac --enable-shared --enable-ffplay

  libavutil      55. 43.100 / 55. 43.100

  libavcodec     57. 70.101 / 57. 70.101

  libavformat    57. 61.100 / 57. 61.100

  libavdevice    57.  2.100 / 57.  2.100

  libavfilter     6. 68.100 /  6. 68.100

  libswscale      4.  3.101 /  4.  3.101

  libswresample   2.  4.100 /  2.  4.100

  libpostproc    54.  2.100 / 54.  2.100

/*

 * 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/buffersink.h>

#include <libavfilter/buffersrc.h>

#include <libavutil/opt.h>

#define ENABLE_FILTERS 1

static const char *filter_descr = "[in0][in1]amix=inputs=2[out]";//"aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";

static const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";

static AVFormatContext *fmt_ctx1;

static AVFormatContext *fmt_ctx2;

static AVCodecContext *dec_ctx1;

static AVCodecContext *dec_ctx2;

AVFilterContext *buffersink_ctx;

AVFilterContext *buffersrc_ctx1;

AVFilterContext *buffersrc_ctx2;

AVFilterGraph *filter_graph;

static int audio_stream_index_1 = -1;

static int audio_stream_index_2 = -1;

static int open_input_file_1(const char *filename)

{

    int ret;

    AVCodec *dec;

    if ((ret = avformat_open_input(&fmt_ctx1, filename, NULL, NULL)) < 0) {

        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");

        return ret;

    }

    if ((ret = avformat_find_stream_info(fmt_ctx1, 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_ctx1, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);

    if (ret < 0) {

        av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");

        return ret;

    }

    audio_stream_index_1 = ret;

    dec_ctx1 = fmt_ctx1->streams[audio_stream_index_1]->codec;

    av_opt_set_int(dec_ctx1, "refcounted_frames", 1, 0);

    /* init the audio decoder */

    if ((ret = avcodec_open2(dec_ctx1, dec, NULL)) < 0) {

        av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");

        return ret;

    }

    return 0;

}

static int open_input_file_2(const char *filename)

{

    int ret;

    AVCodec *dec;

    if ((ret = avformat_open_input(&fmt_ctx2, filename, NULL, NULL)) < 0) {

        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");

        return ret;

    }

    if ((ret = avformat_find_stream_info(fmt_ctx2, 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_ctx2, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);

    if (ret < 0) {

        av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");

        return ret;

    }

    audio_stream_index_2 = ret;

    dec_ctx2 = fmt_ctx2->streams[audio_stream_index_2]->codec;

    av_opt_set_int(dec_ctx2, "refcounted_frames", 1, 0);

    /* init the audio decoder */

    if ((ret = avcodec_open2(dec_ctx2, 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 args1[512];

    char args2[512];

    int ret = 0;

    AVFilter *abuffersrc1  = avfilter_get_by_name("abuffer");

    AVFilter *abuffersrc2  = avfilter_get_by_name("abuffer");

    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");

    AVFilterInOut *outputs1 = avfilter_inout_alloc();

    AVFilterInOut *outputs2 = 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_1 = fmt_ctx1->streams[audio_stream_index_1]->time_base;

    AVRational time_base_2 = fmt_ctx2->streams[audio_stream_index_2]->time_base;

    filter_graph = avfilter_graph_alloc();

    if (!outputs1 || !inputs || !filter_graph) {

        ret = AVERROR(ENOMEM);

        goto end;

    }

    /* buffer audio source: the decoded frames from the decoder will be inserted here. */

    if (!dec_ctx1->channel_layout)

        dec_ctx1->channel_layout = av_get_default_channel_layout(dec_ctx1->channels);

    snprintf(args1, sizeof(args1),

            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,

             time_base_1.num, time_base_1.den, dec_ctx1->sample_rate,

             av_get_sample_fmt_name(dec_ctx1->sample_fmt), dec_ctx1->channel_layout);

    ret = avfilter_graph_create_filter(&buffersrc_ctx1, abuffersrc1, "in1",

                                       args1, NULL, filter_graph);

    if (ret < 0) {

        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");

        goto end;

    }

#if (ENABLE_FILTERS)

    /* buffer audio source: the decoded frames from the decoder will be inserted here. */

    if (!dec_ctx2->channel_layout)

        dec_ctx2->channel_layout = av_get_default_channel_layout(dec_ctx2->channels);

    snprintf(args2, sizeof(args2),

            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,

             time_base_2.num, time_base_2.den, dec_ctx2->sample_rate,

             av_get_sample_fmt_name(dec_ctx2->sample_fmt), dec_ctx2->channel_layout);

    ret = avfilter_graph_create_filter(&buffersrc_ctx2, abuffersrc1, "in2",

                                       args2, NULL, filter_graph);

    if (ret < 0) {

        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");

        goto end;

    }

#endif

    /* 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;

    }

    /*

     * Set the endpoints for the filter graph. The filter_graph will

     * be linked to the graph described by filters_descr.

     */

    /*

     * The buffer source output must be connected to the input pad of

     * the first filter described by filters_descr; since the first

     * filter input label is not specified, it is set to "in" by

     * default.

     */

    outputs1->name       = av_strdup("in0");

    outputs1->filter_ctx = buffersrc_ctx1;

    outputs1->pad_idx    = 0;

#if (ENABLE_FILTERS)

    outputs1->next       = outputs2;

    outputs2->name       = av_strdup("in1");

    outputs2->filter_ctx = buffersrc_ctx2;

    outputs2->pad_idx    = 0;

    outputs2->next       = NULL;

#else

    outputs1->next       = NULL;

#endif

    /*

     * The buffer sink input must be connected to the output pad of

     * the last filter described by filters_descr; since the last

     * filter output label is not specified, it is set to "out" by

     * default.

     */

    inputs->name       = av_strdup("out");

    inputs->filter_ctx = buffersink_ctx;

    inputs->pad_idx    = 0;

    inputs->next       = NULL;

    AVFilterInOut* filter_outputs[2];

    filter_outputs[0] = outputs1;

#if (ENABLE_FILTERS)

    filter_outputs[1] = outputs2;

#endif

    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,

                                        &inputs, &outputs1, NULL)) < 0)//filter_outputs

    {

        av_log(NULL, AV_LOG_ERROR, "parse ptr fail, ret: %d\n", ret);

        goto end;

    }

    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)

    {

        av_log(NULL, AV_LOG_ERROR, "config graph fail, ret: %d\n", ret);

        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(args1, sizeof(args1), -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), "?"),

           args1);

end:

    avfilter_inout_free(&inputs);

    avfilter_inout_free(&outputs1);

    return ret;

}

static void print_frame(const AVFrame *frame)

#if 0

{

    FILE *file = NULL;

    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;

    file = fopen("tmp.pcm", "ab+");

    if (NULL == file){

      perror("fopen tmp.mp3 error\n");

      return;

    } else {

      perror("fopen tmp.aac successful\n");

    }

    fwrite(frame->data[0], n * 2, 1, file);

    fclose(file);

    file = NULL;

}

#else

{

    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);

}

#endif

int main(int argc, char **argv)

{

    int ret;

    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_1(argv[1])) < 0)

    {

        av_log(NULL, AV_LOG_ERROR, "open input file fail, ret: %d\n", ret);

        goto end;

    }

    if ((ret = open_input_file_2(argv[2])) < 0)

    {

        av_log(NULL, AV_LOG_ERROR, "open input file fail, ret: %d\n", ret);

        goto end;

    }

    if ((ret = init_filters(filter_descr)) < 0)

    {

        av_log(NULL, AV_LOG_ERROR, "init filters fail, ret: %d\n", ret);

        goto end;

    }

    AVPacket packet0, packet;

    AVPacket _packet0, _packet;

    /* read all packets */

    packet0.data = NULL;

    packet.data = NULL;

    _packet0.data = NULL;

    _packet.data = NULL;

    while (1) {

        if (!packet0.data) {

            if ((ret = av_read_frame(fmt_ctx1, &packet)) < 0)

                break;

            packet0 = packet;

        }

        if (packet.stream_index == audio_stream_index_1) {

            got_frame = 0;

            ret = avcodec_decode_audio4(dec_ctx1, 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) {

                av_log(NULL, AV_LOG_ERROR, "push frame\n");

                /* push the audio data from decoded frame into the filtergraph */

                if (av_buffersrc_add_frame_flags(buffersrc_ctx1, frame, 0) < 0) {

                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");

                    break;

                }

                av_log(NULL, AV_LOG_ERROR, "pull frame\n");

            }

            if (packet.size <= 0)

                av_packet_unref(&packet0);

        } else {

            /* discard non-wanted packets */

            av_packet_unref(&packet0);

        }

        if (!_packet0.data) {

            if ((ret = av_read_frame(fmt_ctx2, &_packet)) < 0)

                break;

            _packet0 = _packet;

        }

        if (_packet.stream_index == audio_stream_index_2) {

            got_frame = 0;

            ret = avcodec_decode_audio4(dec_ctx2, 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) {

                av_log(NULL, AV_LOG_ERROR, "push frame\n");

                /* push the audio data from decoded frame into the filtergraph */

                if (av_buffersrc_add_frame_flags(buffersrc_ctx2, frame, 0) < 0) {

                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");

                    break;

                }

                av_log(NULL, AV_LOG_ERROR, "pull frame\n");

            }

            if (_packet.size <= 0)

                av_packet_unref(&_packet0);

        } else {

            /* discard non-wanted packets */

            av_packet_unref(&_packet0);

        }

        /* pull filtered audio from the filtergraph */

        if (got_frame)

        {

            while (1) {

                ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);

                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)

                    break;

                if (ret < 0)

                {

                    av_log(NULL, AV_LOG_ERROR, "buffersink get frame fail, ret: %d\n", ret);

                    goto end;

                }

                print_frame(filt_frame);

                av_frame_unref(filt_frame);

            }

        }

    }

end:

    avfilter_graph_free(&filter_graph);

    avcodec_close(dec_ctx1);

    avformat_close_input(&fmt_ctx1);

    avcodec_close(dec_ctx2);

    avformat_close_input(&fmt_ctx2);

    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);

}