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#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;
}
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