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 ([email protected]) * * 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; }
/* * Copyright (c) 2017 Paolo Prete ([email protected]) * * 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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