[Libav-user] New libav API usage axamples

Paolo Prete p4olo_prete at yahoo.it
Mon Mar 27 03:05:16 EEST 2017

during my last job's project I had to use very often the AV library for many purposes. Then, I created many snippets of code which are aligned to the ffmpeg's 3.2 version: they don't use deprecated functions (no warnings from compiler) and can be useful as API usage examples, considering that the current state of the doc/examples directory seems not good and a bit messy. All the snippets that I wrote are short, and they cover many audio+video tasks, from grabbing from audio/video devices to network streaming. If the FFMPEG developers think that they can be pushed in the doc/examples directory, I can spend time in re-organizing all the material and send it progressively to the FFMPEG project. For now, I send an example which converts a raw audio file to float-planar and encodes it to adts-aac. Please, give me some feedback and I'll go on in contributing to the project by sending other examples.

/* * Copyright (c) 2017 Paolo Prete (p4olo_prete at yahoo.it) * * 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 adts-aac encoding raw audio files.  * This example reads a raw audio input file, converts it to float-planar format, performs aac encoding and puts the encoded frames into an ADTS container. The encoded stream is written to  * a file named "out.aac" * The raw input audio file can be created with: ffmpeg -i some_audio_file -f f32le -acodec pcm_f32le -ac 2 -ar 16000 raw_audio_file.raw *  * @example encode_raw_audio_file_to_aac.c */
#include <libavcodec/avcodec.h>#include <libavformat/avformat.h>#include <libavutil/timestamp.h>#include <libswresample/swresample.h>


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 write_adts_muxed_data (void *opaque, uint8_t *adts_data, int size){    FILE *encoded_audio_file = (FILE *)opaque;    fwrite(adts_data, 1, size, encoded_audio_file); //(f)    return size;}

int main(int argc, char **argv){            if (argc != 2) {        av_log(NULL, AV_LOG_ERROR, "Usage: %s <raw audio input file (CHANNELS, INPUT_SAMPLE_FMT, SAMPLE_RATE)>\n", argv[0]);        return 1;    }                int ret_val = 0;    int cleanup_step = 1;                    FILE *input_audio_file = fopen(argv[1], "rb");    if(!input_audio_file){        av_log(NULL, AV_LOG_ERROR, "Could not open input audio file\n");        return AVERROR_EXIT;    }        FILE *encoded_audio_file = fopen("out.aac", "wb");      if(!encoded_audio_file){        av_log(NULL, AV_LOG_ERROR, "Could not open output audio file\n");        ret_val = AVERROR_EXIT;        goto cleanup;    }         ++cleanup_step;    
            //    // Allocate the encoder's context and open the encoder    //    AVCodec *audio_codec = avcodec_find_encoder(AV_CODEC_ID_AAC);    if(!audio_codec){        av_log(NULL, AV_LOG_ERROR, "Could not find aac codec\n");        ret_val = AVERROR_EXIT;        goto cleanup;    }    AVCodecContext *audio_encoder_ctx = avcodec_alloc_context3(audio_codec);    if(!audio_codec){        av_log(NULL, AV_LOG_ERROR, "Could not allocate the encoding context\n");        ret_val = AVERROR_EXIT;        goto cleanup;    }        ++cleanup_step;    audio_encoder_ctx->sample_fmt = AV_SAMPLE_FMT_FLTP;    audio_encoder_ctx->bit_rate = ENCODER_BITRATE;    audio_encoder_ctx->sample_rate = SAMPLE_RATE; // You can use any other sample rate provided by the input file on condition that it is supported by the codec (use AVCodec::supported_samplerates for listing supported sample rates)    audio_encoder_ctx->channels = CHANNELS;    audio_encoder_ctx->channel_layout = av_get_default_channel_layout(CHANNELS);    audio_encoder_ctx->time_base = (AVRational){1, SAMPLE_RATE};    audio_encoder_ctx->codec_type = AVMEDIA_TYPE_AUDIO ;    if ((ret_val = avcodec_open2(audio_encoder_ctx, audio_codec, NULL)) < 0) {        av_log(NULL, AV_LOG_ERROR, "Could not open input codec (error '%s')\n", get_error_text(ret_val));        goto cleanup;    }    ++cleanup_step;            //    // Allocate an AVFrame which will be filled with the input file's data.     //    AVFrame *input_audio_frame;    if (!(input_audio_frame = av_frame_alloc())) {        av_log(NULL, AV_LOG_ERROR, "Could not allocate input frame\n");        ret_val = AVERROR(ENOMEM);        goto cleanup;    }        input_audio_frame->nb_samples     = audio_encoder_ctx->frame_size;    input_audio_frame->format         = INPUT_SAMPLE_FMT;    input_audio_frame->channels       = CHANNELS;    input_audio_frame->sample_rate    = SAMPLE_RATE;    input_audio_frame->channel_layout = av_get_default_channel_layout(CHANNELS);    // Allocate the frame's data buffer     if ((ret_val = av_frame_get_buffer(input_audio_frame, 0)) < 0) {        av_log(NULL, AV_LOG_ERROR, "Could not allocate container for input frame samples (error '%s')\n", get_error_text(ret_val));        ret_val = AVERROR(ENOMEM);        goto cleanup;    }                    //    // Input data must be converted to float-planar format, which is the format required by the AAC encoder. We allocate a SwrContext and an AVFrame (which will contain the converted samples)    // for this task. The AVFrame will feed the encoding function (avcodec_send_frame())    //    SwrContext *audio_convert_context = swr_alloc_set_opts(NULL, av_get_default_channel_layout(CHANNELS), AV_SAMPLE_FMT_FLTP, SAMPLE_RATE, av_get_default_channel_layout(CHANNELS), INPUT_SAMPLE_FMT, SAMPLE_RATE, 0, NULL);    if (!audio_convert_context) {        av_log(NULL, AV_LOG_ERROR, "Could not allocate resample context\n");                         ret_val = AVERROR(ENOMEM);        goto cleanup;    }        ++cleanup_step;    AVFrame *converted_audio_frame;    if (!(converted_audio_frame = av_frame_alloc())) {        av_log(NULL, AV_LOG_ERROR, "Could not allocate resampled frame\n");        ret_val = AVERROR(ENOMEM);        goto cleanup;    }         ++cleanup_step;    converted_audio_frame->nb_samples     = audio_encoder_ctx->frame_size;    converted_audio_frame->format         = audio_encoder_ctx->sample_fmt;    converted_audio_frame->channels       = audio_encoder_ctx->channels;    converted_audio_frame->channel_layout = audio_encoder_ctx->channel_layout;    converted_audio_frame->sample_rate    = SAMPLE_RATE;         if ((ret_val = av_frame_get_buffer(converted_audio_frame, 0)) < 0) {        av_log(NULL, AV_LOG_ERROR, "Could not allocate a buffer for resampled frame samples (error '%s')\n", get_error_text(ret_val));        goto cleanup;    }                    //    // Create the ADTS container for the encoded frames    //    AVOutputFormat *adts_container = av_guess_format("adts", NULL, NULL);    if (!adts_container) {        av_log(NULL, AV_LOG_ERROR, "Could not find adts output format\n");               ret_val = AVERROR_EXIT;        goto cleanup;    }         AVFormatContext *adts_container_ctx;    if ((ret_val = avformat_alloc_output_context2(&adts_container_ctx, adts_container, "", NULL)) < 0){        av_log(NULL, AV_LOG_ERROR, "Could not create output context (error '%s')\n", get_error_text(ret_val));        goto cleanup;    }    ++cleanup_step;    size_t adts_container_buffer_size = 4096;    uint8_t *adts_container_buffer;    if(!(adts_container_buffer = (uint8_t* )av_malloc(adts_container_buffer_size))){        av_log(NULL, AV_LOG_ERROR, "Could not allocate a buffer for the I/O output context\n");               ret_val = AVERROR(ENOMEM);        goto cleanup;     }    ++cleanup_step;    // Create an I/O context for the adts container with a write callback (write_adts_muxed_data()), so that muxed data will be accessed through this function.    AVIOContext *adts_avio_ctx;    if (!(adts_avio_ctx = avio_alloc_context(adts_container_buffer, adts_container_buffer_size, 1, encoded_audio_file, NULL , &write_adts_muxed_data, NULL))) {        av_log(NULL, AV_LOG_ERROR, "Could not create I/O output context\n");        ret_val = AVERROR_EXIT;        goto cleanup;    }    ++cleanup_step;    // Link the container's context to the previous I/O context    adts_container_ctx->pb = adts_avio_ctx;    AVStream *adts_stream;    if (!(adts_stream = avformat_new_stream(adts_container_ctx, NULL))) {        av_log(NULL, AV_LOG_ERROR, "Could not create new stream\n");               ret_val = AVERROR(ENOMEM);        goto cleanup;            }        adts_stream->id = adts_container_ctx->nb_streams-1;    // Copy the encoder's parameters     avcodec_parameters_from_context(adts_stream->codecpar, audio_encoder_ctx);        // Allocate the stream private data and write the stream header    if(avformat_write_header(adts_container_ctx, NULL) < 0){        av_log(NULL, AV_LOG_ERROR, "avformat_write_header() error\n");        ret_val = AVERROR_EXIT;        goto cleanup;    }            ++cleanup_step;                //    // Fill the input frame's data buffer with input file data (a),     // Convert the input frame to float-planar format (b),     // Send the converted frame to the encoder (c),     // Get the encoded packet (d),    // Send the encoded packet to the adts muxer (e).     // Muxed data is caught in write_adts_muxed_data() callback and it is written to the output audio file ( (f) : see above)    //    AVPacket encoded_audio_packet;    av_init_packet(&encoded_audio_packet);    int encoded_pkt_counter = 1;    while(1) {        int audio_bytes_to_encode = fread(input_audio_frame->data[0], 1, input_audio_frame->linesize[0], input_audio_file); //(a)        swr_convert_frame(audio_convert_context, converted_audio_frame, (const AVFrame *)input_audio_frame); //(b)        if(audio_bytes_to_encode != input_audio_frame->linesize[0]){                        break;        }        else {            // Do encode            ret_val = avcodec_send_frame(audio_encoder_ctx, converted_audio_frame);  //(c)            if(ret_val == 0)                 ret_val = avcodec_receive_packet(audio_encoder_ctx, &encoded_audio_packet); //(d)            else{                av_log(NULL, AV_LOG_ERROR, "Error encoding frame (error '%s')\n", get_error_text(ret_val));                goto cleanup;            }                        if(ret_val == 0){                                int64_t pts = converted_audio_frame->nb_samples*(encoded_pkt_counter-1);                encoded_audio_packet.pts = encoded_audio_packet.dts = pts;                           if((ret_val == av_write_frame(adts_container_ctx, &encoded_audio_packet)) < 0){ //(e)                    av_log(NULL, AV_LOG_ERROR, "Error calling av_write_frame() (error '%s')\n", get_error_text(ret_val));                    goto cleanup;                }                else{                    av_log(NULL, AV_LOG_INFO, "Encoded AAC packet %d, size=%d, pts_time=%s\n", encoded_pkt_counter, encoded_audio_packet.size, av_ts2timestr(encoded_audio_packet.pts, &audio_encoder_ctx->time_base));                    ++encoded_pkt_counter;                }            }        }                }    // Flush delayed packets    int still_pkts_to_flush = 1;    int delayed_pkt_counter = 1;        while(still_pkts_to_flush){        int ret = avcodec_send_frame(audio_encoder_ctx, NULL);        if(ret != 0)            still_pkts_to_flush = 0;        ret = avcodec_receive_packet(audio_encoder_ctx, &encoded_audio_packet);        if(ret == 0){            int64_t pts = converted_audio_frame->nb_samples*(encoded_pkt_counter-1);            encoded_audio_packet.pts = encoded_audio_packet.dts = pts;             av_write_frame(adts_container_ctx, &encoded_audio_packet);            av_log(NULL, AV_LOG_INFO, "Flushed encoded AAC delayed packet %d, size=%d, pts_time=%s\n", delayed_pkt_counter, encoded_audio_packet.size, av_ts2timestr(encoded_audio_packet.pts, &audio_encoder_ctx->time_base));            ++delayed_pkt_counter;            ++encoded_pkt_counter;        }            }

    if(cleanup_step > 0)        fclose(input_audio_file);    if(cleanup_step > 1)        fclose(encoded_audio_file);     if(cleanup_step > 2)            avcodec_free_context(&audio_encoder_ctx);    if(cleanup_step > 3)             av_frame_free(&input_audio_frame);    if(cleanup_step > 4)             swr_free(&audio_convert_context);       if(cleanup_step > 5)             av_frame_free(&converted_audio_frame);    if(cleanup_step > 6)            avformat_free_context(adts_container_ctx);    if(cleanup_step > 7)            av_free(adts_container_buffer);    if(cleanup_step > 8)            av_free(adts_avio_ctx);      if(cleanup_step > 9)            av_packet_unref(&encoded_audio_packet);                return ret_val;    }

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