[Libav-user] New libav API usage axamples

Renaud BOBIN renaud.bobin at subsea-tech.com
Mon Mar 27 10:00:22 EEST 2017


Hello,

 

That’s a good idea !

I’m using ffmpeg to decode stream for a Network Camera, but the last time i use it, it was with the old library.

Example with the current library will be really helpful for all new users !

Thx for your work

 

Cordialement,

Renaud BOBIN
Project Engineer

 

SUBSEA TECH

Marine and Underwater Technologies

167 Plage de l'Estaque 13016 Marseille - France

Tel: +33 (0) 4 91 51 76 71

Web:  <http://www.subsea-tech.com/> www.subsea-tech.com

 

De : Libav-user [mailto:libav-user-bounces at ffmpeg.org] De la part de Paolo Prete
Envoyé : lundi 27 mars 2017 02:05
À : libav-user at ffmpeg.org
Objet : [Libav-user] New libav API usage axamples

 

Hello, 

 

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 <mailto: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>

 

 

#define ENCODER_BITRATE 64000

#define SAMPLE_RATE 16000

#define INPUT_SAMPLE_FMT AV_SAMPLE_FMT_FLT

#define CHANNELS 2

 

 

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;    

 

    

    

    av_register_all();

 

    

    

    //

    // 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;

        }        

    }

 

    

    av_write_trailer(adts_container_ctx);  

 

    

    

    

cleanup:    

 

 

    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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