[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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