ffmpeg-amix

#include <stdio.h>

#import "libavcodec/avcodec.h"

#include "libavutil/channel_layout.h"
#include "libavutil/md5.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"

#include "libavfilter/avfilter.h"
#include "libavfilter/buffersink.h"
#include "libavfilter/buffersrc.h"

#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>

#include "libavformat/avio.h"

#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"

#define INPUT_SAMPLERATE     44100
#define INPUT_FORMAT         AV_SAMPLE_FMT_S16
#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_STEREO

/** The output bit rate in kbit/s */
#define OUTPUT_BIT_RATE 44100
/** The number of output channels */
#define OUTPUT_CHANNELS 2
/** The audio sample output format */
#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16

#define VOLUME_VAL 0.90

AVFormatContext *output_format_context = NULL;
AVCodecContext *output_codec_context = NULL;

AVFormatContext *input_format_context_0 = NULL;
AVCodecContext *input_codec_context_0 = NULL;
AVFormatContext *input_format_context_1 = NULL;
AVCodecContext *input_codec_context_1 = NULL;

AVFilterGraph *graph;
AVFilterContext *src0,*src1, *sink;

static char *const get_error_text(const int error)
{
    static char error_buffer[255];
    av_strerror(error, error_buffer, sizeof(error_buffer));
    return error_buffer;
}

static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src0, AVFilterContext **src1,
                             AVFilterContext **sink)
{
    AVFilterGraph *filter_graph;
    AVFilterContext *abuffer1_ctx;
    AVFilter        *abuffer1;
	AVFilterContext *abuffer0_ctx;
    AVFilter        *abuffer0;
    AVFilterContext *mix_ctx;
    AVFilter        *mix_filter;
    AVFilterContext *abuffersink_ctx;
    AVFilter        *abuffersink;
	
	char args[512];
	
    int err;
	
    /* Create a new filtergraph, which will contain all the filters. */
    filter_graph = avfilter_graph_alloc();
    if (!filter_graph) {
        av_log(NULL, AV_LOG_ERROR, "Unable to create filter graph.\n");
        return AVERROR(ENOMEM);
    }
	
	/****** abuffer 0 ********/
    
    /* Create the abuffer filter;
     * it will be used for feeding the data into the graph. */
    abuffer0 = avfilter_get_by_name("abuffer");
    if (!abuffer0) {
        av_log(NULL, AV_LOG_ERROR, "Could not find the abuffer filter.\n");
        return AVERROR_FILTER_NOT_FOUND;
    }
	
	/* buffer audio source: the decoded frames from the decoder will be inserted here. */
    if (!input_codec_context_0->channel_layout)
        input_codec_context_0->channel_layout = av_get_default_channel_layout(input_codec_context_0->channels);
    snprintf(args, sizeof(args),
			 "sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
             input_codec_context_0->sample_rate,
             av_get_sample_fmt_name(input_codec_context_0->sample_fmt), input_codec_context_0->channel_layout);
	
	
	err = avfilter_graph_create_filter(&abuffer0_ctx, abuffer0, "src0",
                                       args, NULL, filter_graph);
    if (err < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
        return err;
    }
    
	/****** abuffer 1 ******* */
	
	/* Create the abuffer filter;
     * it will be used for feeding the data into the graph. */
    abuffer1 = avfilter_get_by_name("abuffer");
    if (!abuffer1) {
        av_log(NULL, AV_LOG_ERROR, "Could not find the abuffer filter.\n");
        return AVERROR_FILTER_NOT_FOUND;
    }
	
	/* buffer audio source: the decoded frames from the decoder will be inserted here. */
    if (!input_codec_context_1->channel_layout)
        input_codec_context_1->channel_layout = av_get_default_channel_layout(input_codec_context_1->channels);
    snprintf(args, sizeof(args),
			 "sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
             input_codec_context_1->sample_rate,
             av_get_sample_fmt_name(input_codec_context_1->sample_fmt), input_codec_context_1->channel_layout);
	
	
	err = avfilter_graph_create_filter(&abuffer1_ctx, abuffer1, "src1",
                                       args, NULL, filter_graph);
    if (err < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
        return err;
    }
	
    /****** amix ******* */
    /* Create mix filter. */
    mix_filter = avfilter_get_by_name("amix");
    if (!mix_filter) {
        av_log(NULL, AV_LOG_ERROR, "Could not find the mix filter.\n");
        return AVERROR_FILTER_NOT_FOUND;
    }
    
    snprintf(args, sizeof(args), "inputs=2");
	
	err = avfilter_graph_create_filter(&mix_ctx, mix_filter, "amix",
                                       args, NULL, filter_graph);

    if (err < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio amix filter\n");
        return err;
    }
	
    /* Finally create the abuffersink filter;
     * it will be used to get the filtered data out of the graph. */
    abuffersink = avfilter_get_by_name("abuffersink");
    if (!abuffersink) {
        av_log(NULL, AV_LOG_ERROR, "Could not find the abuffersink filter.\n");
        return AVERROR_FILTER_NOT_FOUND;
    }
	
    abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
    if (!abuffersink_ctx) {
        av_log(NULL, AV_LOG_ERROR, "Could not allocate the abuffersink instance.\n");
        return AVERROR(ENOMEM);
    }
	
    /* Same sample fmts as the output file. */
    err = av_opt_set_int_list(abuffersink_ctx, "sample_fmts",
                              ((int[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }),
                              AV_SAMPLE_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
    
    uint8_t ch_layout[64];
    av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, OUTPUT_CHANNELS);
    av_opt_set    (abuffersink_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
    
    if (err < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could set options to the abuffersink instance.\n");
        return err;
    }
    
    err = avfilter_init_str(abuffersink_ctx, NULL);
    if (err < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could not initialize the abuffersink instance.\n");
        return err;
    }
    
    
    /* Connect the filters; */
    
	err = avfilter_link(abuffer0_ctx, 0, mix_ctx, 0);
	if (err >= 0)
        err = avfilter_link(abuffer1_ctx, 0, mix_ctx, 1);
	if (err >= 0)
        err = avfilter_link(mix_ctx, 0, abuffersink_ctx, 0);
    if (err < 0) {
        av_log(NULL, AV_LOG_ERROR, "Error connecting filters\n");
        return err;
    }
	
    /* Configure the graph. */
    err = avfilter_graph_config(filter_graph, NULL);
    if (err < 0) {
        av_log(NULL, AV_LOG_ERROR, "Error while configuring graph : %s\n", get_error_text(err));
        return err;
    }
    
    char* dump =avfilter_graph_dump(filter_graph, NULL);
    av_log(NULL, AV_LOG_ERROR, "Graph :\n%s\n", dump);
	
    *graph = filter_graph;
    *src0   = abuffer0_ctx;
	*src1   = abuffer1_ctx;
    *sink  = abuffersink_ctx;
    
    return 0;
}


/** Open an input file and the required decoder. */
static int open_input_file(const char *filename,
                           AVFormatContext **input_format_context,
                           AVCodecContext **input_codec_context)
{
    AVCodec *input_codec;
    int error;
	
    /** Open the input file to read from it. */
    if ((error = avformat_open_input(input_format_context, filename, NULL,
                                     NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could not open input file '%s' (error '%s')\n",
               filename, get_error_text(error));
        *input_format_context = NULL;
        return error;
    }
	
    /** Get information on the input file (number of streams etc.). */
    if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could not open find stream info (error '%s')\n",
               get_error_text(error));
        avformat_close_input(input_format_context);
        return error;
    }
	
    /** Make sure that there is only one stream in the input file. */
    if ((*input_format_context)->nb_streams != 1) {
        av_log(NULL, AV_LOG_ERROR, "Expected one audio input stream, but found %d\n",
               (*input_format_context)->nb_streams);
        avformat_close_input(input_format_context);
        return AVERROR_EXIT;
    }
	
    /** Find a decoder for the audio stream. */
    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
        av_log(NULL, AV_LOG_ERROR, "Could not find input codec\n");
        avformat_close_input(input_format_context);
        return AVERROR_EXIT;
    }
	
    /** Open the decoder for the audio stream to use it later. */
    if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
                               input_codec, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could not open input codec (error '%s')\n",
               get_error_text(error));
        avformat_close_input(input_format_context);
        return error;
    }
	
    /** Save the decoder context for easier access later. */
    *input_codec_context = (*input_format_context)->streams[0]->codec;
	
    return 0;
}

/**
 * Open an output file and the required encoder.
 * Also set some basic encoder parameters.
 * Some of these parameters are based on the input file's parameters.
 */
static int open_output_file(const char *filename,
                            AVCodecContext *input_codec_context,
                            AVFormatContext **output_format_context,
                            AVCodecContext **output_codec_context)
{
    AVIOContext *output_io_context = NULL;
    AVStream *stream               = NULL;
    AVCodec *output_codec          = NULL;
    int error;
	
    /** Open the output file to write to it. */
    if ((error = avio_open(&output_io_context, filename,
                           AVIO_FLAG_WRITE)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s' (error '%s')\n",
               filename, get_error_text(error));
        return error;
    }
	
    /** Create a new format context for the output container format. */
    if (!(*output_format_context = avformat_alloc_context())) {
        av_log(NULL, AV_LOG_ERROR, "Could not allocate output format context\n");
        return AVERROR(ENOMEM);
    }
	
    /** Associate the output file (pointer) with the container format context. */
    (*output_format_context)->pb = output_io_context;
	
    /** Guess the desired container format based on the file extension. */
    if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
                                                              NULL))) {
        av_log(NULL, AV_LOG_ERROR, "Could not find output file format\n");
        goto cleanup;
    }
	
    av_strlcpy((*output_format_context)->filename, filename,
               sizeof((*output_format_context)->filename));
	
    /** Find the encoder to be used by its name. */
    if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_PCM_S16LE))) {
        av_log(NULL, AV_LOG_ERROR, "Could not find an PCM encoder.\n");
        goto cleanup;
    }
	
    /** Create a new audio stream in the output file container. */
    if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
        av_log(NULL, AV_LOG_ERROR, "Could not create new stream\n");
        error = AVERROR(ENOMEM);
        goto cleanup;
    }
	
    /** Save the encoder context for easiert access later. */
    *output_codec_context = stream->codec;
	
    /**
     * Set the basic encoder parameters.
     */
    (*output_codec_context)->channels       = OUTPUT_CHANNELS;
    (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
    (*output_codec_context)->sample_rate    = input_codec_context->sample_rate;
    (*output_codec_context)->sample_fmt     = AV_SAMPLE_FMT_S16;
    //(*output_codec_context)->bit_rate       = input_codec_context->bit_rate;
    
    av_log(NULL, AV_LOG_INFO, "output bitrate %d\n", (*output_codec_context)->bit_rate);
	
    /**
     * Some container formats (like MP4) require global headers to be present
     * Mark the encoder so that it behaves accordingly.
     */
    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
        (*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
	
    /** Open the encoder for the audio stream to use it later. */
    if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could not open output codec (error '%s')\n",
               get_error_text(error));
        goto cleanup;
    }
	
    return 0;
	
cleanup:
    avio_close((*output_format_context)->pb);
    avformat_free_context(*output_format_context);
    *output_format_context = NULL;
    return error < 0 ? error : AVERROR_EXIT;
}

/** Initialize one audio frame for reading from the input file */
static int init_input_frame(AVFrame **frame)
{
    if (!(*frame = av_frame_alloc())) {
        av_log(NULL, AV_LOG_ERROR, "Could not allocate input frame\n");
        return AVERROR(ENOMEM);
    }
    return 0;
}

/** Initialize one data packet for reading or writing. */
static void init_packet(AVPacket *packet)
{
    av_init_packet(packet);
    /** Set the packet data and size so that it is recognized as being empty. */
    packet->data = NULL;
    packet->size = 0;
}


/** Decode one audio frame from the input file. */
static int decode_audio_frame(AVFrame *frame,
                              AVFormatContext *input_format_context,
                              AVCodecContext *input_codec_context,
                              int *data_present, int *finished)
{
    /** Packet used for temporary storage. */
    AVPacket input_packet;
    int error;
    init_packet(&input_packet);
	
    /** Read one audio frame from the input file into a temporary packet. */
    if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
        /** If we are the the end of the file, flush the decoder below. */
        if (error == AVERROR_EOF)
            *finished = 1;
        else {
            av_log(NULL, AV_LOG_ERROR, "Could not read frame (error '%s')\n",
                   get_error_text(error));
            return error;
        }
    }
	
    /**
     * Decode the audio frame stored in the temporary packet.
     * The input audio stream decoder is used to do this.
     * If we are at the end of the file, pass an empty packet to the decoder
     * to flush it.
     */
    if ((error = avcodec_decode_audio4(input_codec_context, frame,
                                       data_present, &input_packet)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could not decode frame (error '%s')\n",
               get_error_text(error));
        av_free_packet(&input_packet);
        return error;
    }
	
    /**
     * If the decoder has not been flushed completely, we are not finished,
     * so that this function has to be called again.
     */
    if (*finished && *data_present)
        *finished = 0;
    av_free_packet(&input_packet);
    return 0;
}

/** Encode one frame worth of audio to the output file. */
static int encode_audio_frame(AVFrame *frame,
                              AVFormatContext *output_format_context,
                              AVCodecContext *output_codec_context,
                              int *data_present)
{
    /** Packet used for temporary storage. */
    AVPacket output_packet;
    int error;
    init_packet(&output_packet);
	
    /**
     * Encode the audio frame and store it in the temporary packet.
     * The output audio stream encoder is used to do this.
     */
    if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
                                       frame, data_present)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could not encode frame (error '%s')\n",
               get_error_text(error));
        av_free_packet(&output_packet);
        return error;
    }
	
    /** Write one audio frame from the temporary packet to the output file. */
    if (*data_present) {
        if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
            av_log(NULL, AV_LOG_ERROR, "Could not write frame (error '%s')\n",
                   get_error_text(error));
            av_free_packet(&output_packet);
            return error;
        }
		
        av_free_packet(&output_packet);
    }
	
    return 0;
}

static int process_all(){
	int ret = 0;
	
	int data_present = 0;
	int finished = 0;
    
    int nb_inputs = 2;
    
    AVFormatContext* input_format_contexts[2];
    AVCodecContext* input_codec_contexts[2];
    input_format_contexts[0] = input_format_context_0;
    input_format_contexts[1] = input_format_context_1;
    input_codec_contexts[0] = input_codec_context_0;
    input_codec_contexts[1] = input_codec_context_1;
    
    AVFilterContext* buffer_contexts[2];
    buffer_contexts[0] = src0;
    buffer_contexts[1] = src1;
	
    int input_finished[2];
    input_finished[0] = 0;
    input_finished[1] = 0;
    
    int input_to_read[2];
    input_to_read[0] = 1;
    input_to_read[1] = 1;
    
    int total_samples[2];
    total_samples[0] = 0;
    total_samples[1] = 0;
    
    int total_out_samples = 0;
    
    int nb_finished = 0;
    
    
	
    while (nb_finished < nb_inputs) {
        int data_present_in_graph = 0;
		
        for(int i = 0 ; i < nb_inputs ; i++){
            if(input_finished[i] || input_to_read[i] == 0){
                continue;
            }
            
            input_to_read[i] = 0;
            
            AVFrame *frame = NULL;
            
            if(init_input_frame(&frame) > 0){
                goto end;
            }
            
            /** Decode one frame worth of audio samples. */
            if ( (ret = decode_audio_frame(frame, input_format_contexts[i], input_codec_contexts[i], &data_present, &finished))){
                goto end;
            }
            /**
             * If we are at the end of the file and there are no more samples
             * in the decoder which are delayed, we are actually finished.
             * This must not be treated as an error.
             */
            if (finished && !data_present) {
                input_finished[i] = 1;
                nb_finished++;
                ret = 0;
                av_log(NULL, AV_LOG_INFO, "Input n°%d finished. Write NULL frame \n", i);
                
                ret = av_buffersrc_write_frame(buffer_contexts[i], NULL);
                if (ret < 0) {
                    av_log(NULL, AV_LOG_ERROR, "Error writing EOF null frame for input %d\n", i);
                    goto end;
                }
            }
            else if (data_present) { /** If there is decoded data, convert and store it */
                /* push the audio data from decoded frame into the filtergraph */
                ret = av_buffersrc_write_frame(buffer_contexts[i], frame);
                if (ret < 0) {
                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
                    goto end;
                }
                
                av_log(NULL, AV_LOG_INFO, "add %d samples on input %d (%d Hz, time=%f, ttime=%f)\n",
                       frame->nb_samples, i, input_codec_contexts[i]->sample_rate,
                       (double)frame->nb_samples / input_codec_contexts[i]->sample_rate,
                       (double)(total_samples[i] += frame->nb_samples) / input_codec_contexts[i]->sample_rate);
                
            }
            
            av_frame_free(&frame);
            
            data_present_in_graph = data_present | data_present_in_graph;
        }
        
        if(data_present_in_graph){
            AVFrame *filt_frame = av_frame_alloc();
            
            /* pull filtered audio from the filtergraph */
            while (1) {
                ret = av_buffersink_get_frame(sink, filt_frame);
                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF){
                    for(int i = 0 ; i < nb_inputs ; i++){
                        if(av_buffersrc_get_nb_failed_requests(buffer_contexts[i]) > 0){
                            input_to_read[i] = 1;
                            av_log(NULL, AV_LOG_INFO, "Need to read input %d\n", i);
                        }
                    }
                    
                    break;
                }
                if (ret < 0)
                    goto end;
                
                av_log(NULL, AV_LOG_INFO, "remove %d samples from sink (%d Hz, time=%f, ttime=%f)\n",
                       filt_frame->nb_samples, output_codec_context->sample_rate,
                       (double)filt_frame->nb_samples / output_codec_context->sample_rate,
                       (double)(total_out_samples += filt_frame->nb_samples) / output_codec_context->sample_rate);
                
                //av_log(NULL, AV_LOG_INFO, "Data read from graph\n");
                ret = encode_audio_frame(filt_frame, output_format_context, output_codec_context, &data_present);
                if (ret < 0)
                    goto end;
                
                av_frame_unref(filt_frame);
            }
            
            av_frame_free(&filt_frame);
        } else {
             av_log(NULL, AV_LOG_INFO, "No data in graph\n");
            for(int i = 0 ; i < nb_inputs ; i++){
                input_to_read[i] = 1;
            }
        }

    }
	
	return 0;
	
end:
    //    avcodec_close(input_codec_context);
    //    avformat_close_input(&input_format_context);
    //    av_frame_free(&frame);
    //    av_frame_free(&filt_frame);
	
    if (ret < 0 && ret != AVERROR_EOF) {
        av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
        exit(1);
    }
	
    exit(0);
}



/** Write the header of the output file container. */
static int write_output_file_header(AVFormatContext *output_format_context)
{
    int error;
    if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could not write output file header (error '%s')\n",
               get_error_text(error));
        return error;
    }
    return 0;
}

/** Write the trailer of the output file container. */
static int write_output_file_trailer(AVFormatContext *output_format_context)
{
    int error;
    if ((error = av_write_trailer(output_format_context)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Could not write output file trailer (error '%s')\n",
               get_error_text(error));
        return error;
    }
    return 0;
}


int main(int argc, const char * argv[])
{
    av_log_set_level(AV_LOG_VERBOSE);
    

    int err;
	
	av_register_all();
	avfilter_register_all();
	   
    char* audio1Path = "audio10.wav";
    
	if (open_input_file(audio1Path, &input_format_context_0, &input_codec_context_0) < 0){
        av_log(NULL, AV_LOG_ERROR, "Error while opening file 1\n");
		exit(1);
	}
	av_dump_format(input_format_context_0, 0, audio1Path, 0);
	
    char* audio2Path = "audio20.wav";
	
	if (open_input_file(audio2Path, &input_format_context_1, &input_codec_context_1) < 0){
		av_log(NULL, AV_LOG_ERROR, "Error while opening file 2\n");
		exit(1);
	}
	av_dump_format(input_format_context_1, 0, audio2Path, 0);
    
	
    /* Set up the filtergraph. */
    err = init_filter_graph(&graph, &src0, &src1, &sink);
	printf("Init err = %d\n", err);

	
	char* outputFile = "output.wav";
    remove(outputFile);
    
	av_log(NULL, AV_LOG_INFO, "Output file : %s\n", outputFile);
	
	err = open_output_file(outputFile, input_codec_context_0, &output_format_context, &output_codec_context);
	printf("open output file err : %d\n", err);
	av_dump_format(output_format_context, 0, outputFile, 1);
	
	if(write_output_file_header(output_format_context) < 0){
		av_log(NULL, AV_LOG_ERROR, "Error while writing header outputfile\n");
		exit(1);
	}

	process_all();
    
	if(write_output_file_trailer(output_format_context) < 0){
		av_log(NULL, AV_LOG_ERROR, "Error while writing header outputfile\n");
		exit(1);
	}
    	
	printf("FINISHED\n");
    
    return 0;
}