Edit File: transcode_aac.c
/* * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * simple audio converter * * @example transcode_aac.c * Convert an input audio file to AAC in an MP4 container using FFmpeg. * @author Andreas Unterweger (dustsigns@gmail.com) */ #include <stdio.h> #include "libavformat/avformat.h" #include "libavformat/avio.h" #include "libavcodec/avcodec.h" #include "libavutil/audio_fifo.h" #include "libavutil/avassert.h" #include "libavutil/avstring.h" #include "libavutil/frame.h" #include "libavutil/opt.h" #include "libswresample/swresample.h" /** The output bit rate in kbit/s */ #define OUTPUT_BIT_RATE 96000 /** The number of output channels */ #define OUTPUT_CHANNELS 2 /** * Convert an error code into a text message. * @param error Error code to be converted * @return Corresponding error text (not thread-safe) */ static const char *get_error_text(const int error) { static char error_buffer[255]; av_strerror(error, error_buffer, sizeof(error_buffer)); return error_buffer; } /** 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) { fprintf(stderr, "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) { fprintf(stderr, "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) { fprintf(stderr, "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))) { fprintf(stderr, "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) { fprintf(stderr, "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) { fprintf(stderr, "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())) { fprintf(stderr, "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))) { fprintf(stderr, "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_AAC))) { fprintf(stderr, "Could not find an AAC encoder.\n"); goto cleanup; } /** Create a new audio stream in the output file container. */ if (!(stream = avformat_new_stream(*output_format_context, output_codec))) { fprintf(stderr, "Could not create new stream\n"); error = AVERROR(ENOMEM); goto cleanup; } /** Save the encoder context for easier access later. */ *output_codec_context = stream->codec; /** * Set the basic encoder parameters. * The input file's sample rate is used to avoid a sample rate conversion. */ (*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 = output_codec->sample_fmts[0]; (*output_codec_context)->bit_rate = OUTPUT_BIT_RATE; /** Allow the use of the experimental AAC encoder */ (*output_codec_context)->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL; /** Set the sample rate for the container. */ stream->time_base.den = input_codec_context->sample_rate; stream->time_base.num = 1; /** * 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 |= AV_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) { fprintf(stderr, "Could not open output codec (error '%s')\n", get_error_text(error)); goto cleanup; } return 0; cleanup: avio_closep(&(*output_format_context)->pb); avformat_free_context(*output_format_context); *output_format_context = NULL; return error < 0 ? error : AVERROR_EXIT; } /** 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; } /** Initialize one audio frame for reading from the input file */ static int init_input_frame(AVFrame **frame) { if (!(*frame = av_frame_alloc())) { fprintf(stderr, "Could not allocate input frame\n"); return AVERROR(ENOMEM); } return 0; } /** * Initialize the audio resampler based on the input and output codec settings. * If the input and output sample formats differ, a conversion is required * libswresample takes care of this, but requires initialization. */ static int init_resampler(AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, SwrContext **resample_context) { int error; /** * Create a resampler context for the conversion. * Set the conversion parameters. * Default channel layouts based on the number of channels * are assumed for simplicity (they are sometimes not detected * properly by the demuxer and/or decoder). */ *resample_context = swr_alloc_set_opts(NULL, av_get_default_channel_layout(output_codec_context->channels), output_codec_context->sample_fmt, output_codec_context->sample_rate, av_get_default_channel_layout(input_codec_context->channels), input_codec_context->sample_fmt, input_codec_context->sample_rate, 0, NULL); if (!*resample_context) { fprintf(stderr, "Could not allocate resample context\n"); return AVERROR(ENOMEM); } /** * Perform a sanity check so that the number of converted samples is * not greater than the number of samples to be converted. * If the sample rates differ, this case has to be handled differently */ av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate); /** Open the resampler with the specified parameters. */ if ((error = swr_init(*resample_context)) < 0) { fprintf(stderr, "Could not open resample context\n"); swr_free(resample_context); return error; } return 0; } /** Initialize a FIFO buffer for the audio samples to be encoded. */ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context) { /** Create the FIFO buffer based on the specified output sample format. */ if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt, output_codec_context->channels, 1))) { fprintf(stderr, "Could not allocate FIFO\n"); return AVERROR(ENOMEM); } return 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) { fprintf(stderr, "Could not write output file header (error '%s')\n", get_error_text(error)); return error; } return 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 at the end of the file, flush the decoder below. */ if (error == AVERROR_EOF) *finished = 1; else { fprintf(stderr, "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) { fprintf(stderr, "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; } /** * Initialize a temporary storage for the specified number of audio samples. * The conversion requires temporary storage due to the different format. * The number of audio samples to be allocated is specified in frame_size. */ static int init_converted_samples(uint8_t ***converted_input_samples, AVCodecContext *output_codec_context, int frame_size) { int error; /** * Allocate as many pointers as there are audio channels. * Each pointer will later point to the audio samples of the corresponding * channels (although it may be NULL for interleaved formats). */ if (!(*converted_input_samples = calloc(output_codec_context->channels, sizeof(**converted_input_samples)))) { fprintf(stderr, "Could not allocate converted input sample pointers\n"); return AVERROR(ENOMEM); } /** * Allocate memory for the samples of all channels in one consecutive * block for convenience. */ if ((error = av_samples_alloc(*converted_input_samples, NULL, output_codec_context->channels, frame_size, output_codec_context->sample_fmt, 0)) < 0) { fprintf(stderr, "Could not allocate converted input samples (error '%s')\n", get_error_text(error)); av_freep(&(*converted_input_samples)[0]); free(*converted_input_samples); return error; } return 0; } /** * Convert the input audio samples into the output sample format. * The conversion happens on a per-frame basis, the size of which is specified * by frame_size. */ static int convert_samples(const uint8_t **input_data, uint8_t **converted_data, const int frame_size, SwrContext *resample_context) { int error; /** Convert the samples using the resampler. */ if ((error = swr_convert(resample_context, converted_data, frame_size, input_data , frame_size)) < 0) { fprintf(stderr, "Could not convert input samples (error '%s')\n", get_error_text(error)); return error; } return 0; } /** Add converted input audio samples to the FIFO buffer for later processing. */ static int add_samples_to_fifo(AVAudioFifo *fifo, uint8_t **converted_input_samples, const int frame_size) { int error; /** * Make the FIFO as large as it needs to be to hold both, * the old and the new samples. */ if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) { fprintf(stderr, "Could not reallocate FIFO\n"); return error; } /** Store the new samples in the FIFO buffer. */ if (av_audio_fifo_write(fifo, (void **)converted_input_samples, frame_size) < frame_size) { fprintf(stderr, "Could not write data to FIFO\n"); return AVERROR_EXIT; } return 0; } /** * Read one audio frame from the input file, decodes, converts and stores * it in the FIFO buffer. */ static int read_decode_convert_and_store(AVAudioFifo *fifo, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, SwrContext *resampler_context, int *finished) { /** Temporary storage of the input samples of the frame read from the file. */ AVFrame *input_frame = NULL; /** Temporary storage for the converted input samples. */ uint8_t **converted_input_samples = NULL; int data_present; int ret = AVERROR_EXIT; /** Initialize temporary storage for one input frame. */ if (init_input_frame(&input_frame)) goto cleanup; /** Decode one frame worth of audio samples. */ if (decode_audio_frame(input_frame, input_format_context, input_codec_context, &data_present, finished)) goto cleanup; /** * 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) { ret = 0; goto cleanup; } /** If there is decoded data, convert and store it */ if (data_present) { /** Initialize the temporary storage for the converted input samples. */ if (init_converted_samples(&converted_input_samples, output_codec_context, input_frame->nb_samples)) goto cleanup; /** * Convert the input samples to the desired output sample format. * This requires a temporary storage provided by converted_input_samples. */ if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples, input_frame->nb_samples, resampler_context)) goto cleanup; /** Add the converted input samples to the FIFO buffer for later processing. */ if (add_samples_to_fifo(fifo, converted_input_samples, input_frame->nb_samples)) goto cleanup; ret = 0; } ret = 0; cleanup: if (converted_input_samples) { av_freep(&converted_input_samples[0]); free(converted_input_samples); } av_frame_free(&input_frame); return ret; } /** * Initialize one input frame for writing to the output file. * The frame will be exactly frame_size samples large. */ static int init_output_frame(AVFrame **frame, AVCodecContext *output_codec_context, int frame_size) { int error; /** Create a new frame to store the audio samples. */ if (!(*frame = av_frame_alloc())) { fprintf(stderr, "Could not allocate output frame\n"); return AVERROR_EXIT; } /** * Set the frame's parameters, especially its size and format. * av_frame_get_buffer needs this to allocate memory for the * audio samples of the frame. * Default channel layouts based on the number of channels * are assumed for simplicity. */ (*frame)->nb_samples = frame_size; (*frame)->channel_layout = output_codec_context->channel_layout; (*frame)->format = output_codec_context->sample_fmt; (*frame)->sample_rate = output_codec_context->sample_rate; /** * Allocate the samples of the created frame. This call will make * sure that the audio frame can hold as many samples as specified. */ if ((error = av_frame_get_buffer(*frame, 0)) < 0) { fprintf(stderr, "Could allocate output frame samples (error '%s')\n", get_error_text(error)); av_frame_free(frame); return error; } return 0; } /** Global timestamp for the audio frames */ static int64_t pts = 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); /** Set a timestamp based on the sample rate for the container. */ if (frame) { frame->pts = pts; pts += frame->nb_samples; } /** * 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) { fprintf(stderr, "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) { fprintf(stderr, "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; } /** * Load one audio frame from the FIFO buffer, encode and write it to the * output file. */ static int load_encode_and_write(AVAudioFifo *fifo, AVFormatContext *output_format_context, AVCodecContext *output_codec_context) { /** Temporary storage of the output samples of the frame written to the file. */ AVFrame *output_frame; /** * Use the maximum number of possible samples per frame. * If there is less than the maximum possible frame size in the FIFO * buffer use this number. Otherwise, use the maximum possible frame size */ const int frame_size = FFMIN(av_audio_fifo_size(fifo), output_codec_context->frame_size); int data_written; /** Initialize temporary storage for one output frame. */ if (init_output_frame(&output_frame, output_codec_context, frame_size)) return AVERROR_EXIT; /** * Read as many samples from the FIFO buffer as required to fill the frame. * The samples are stored in the frame temporarily. */ if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) { fprintf(stderr, "Could not read data from FIFO\n"); av_frame_free(&output_frame); return AVERROR_EXIT; } /** Encode one frame worth of audio samples. */ if (encode_audio_frame(output_frame, output_format_context, output_codec_context, &data_written)) { av_frame_free(&output_frame); return AVERROR_EXIT; } av_frame_free(&output_frame); 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) { fprintf(stderr, "Could not write output file trailer (error '%s')\n", get_error_text(error)); return error; } return 0; } /** Convert an audio file to an AAC file in an MP4 container. */ int main(int argc, char **argv) { AVFormatContext *input_format_context = NULL, *output_format_context = NULL; AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL; SwrContext *resample_context = NULL; AVAudioFifo *fifo = NULL; int ret = AVERROR_EXIT; if (argc < 3) { fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]); exit(1); } /** Register all codecs and formats so that they can be used. */ av_register_all(); /** Open the input file for reading. */ if (open_input_file(argv[1], &input_format_context, &input_codec_context)) goto cleanup; /** Open the output file for writing. */ if (open_output_file(argv[2], input_codec_context, &output_format_context, &output_codec_context)) goto cleanup; /** Initialize the resampler to be able to convert audio sample formats. */ if (init_resampler(input_codec_context, output_codec_context, &resample_context)) goto cleanup; /** Initialize the FIFO buffer to store audio samples to be encoded. */ if (init_fifo(&fifo, output_codec_context)) goto cleanup; /** Write the header of the output file container. */ if (write_output_file_header(output_format_context)) goto cleanup; /** * Loop as long as we have input samples to read or output samples * to write; abort as soon as we have neither. */ while (1) { /** Use the encoder's desired frame size for processing. */ const int output_frame_size = output_codec_context->frame_size; int finished = 0; /** * Make sure that there is one frame worth of samples in the FIFO * buffer so that the encoder can do its work. * Since the decoder's and the encoder's frame size may differ, we * need to FIFO buffer to store as many frames worth of input samples * that they make up at least one frame worth of output samples. */ while (av_audio_fifo_size(fifo) < output_frame_size) { /** * Decode one frame worth of audio samples, convert it to the * output sample format and put it into the FIFO buffer. */ if (read_decode_convert_and_store(fifo, input_format_context, input_codec_context, output_codec_context, resample_context, &finished)) goto cleanup; /** * If we are at the end of the input file, we continue * encoding the remaining audio samples to the output file. */ if (finished) break; } /** * If we have enough samples for the encoder, we encode them. * At the end of the file, we pass the remaining samples to * the encoder. */ while (av_audio_fifo_size(fifo) >= output_frame_size || (finished && av_audio_fifo_size(fifo) > 0)) /** * Take one frame worth of audio samples from the FIFO buffer, * encode it and write it to the output file. */ if (load_encode_and_write(fifo, output_format_context, output_codec_context)) goto cleanup; /** * If we are at the end of the input file and have encoded * all remaining samples, we can exit this loop and finish. */ if (finished) { int data_written; /** Flush the encoder as it may have delayed frames. */ do { if (encode_audio_frame(NULL, output_format_context, output_codec_context, &data_written)) goto cleanup; } while (data_written); break; } } /** Write the trailer of the output file container. */ if (write_output_file_trailer(output_format_context)) goto cleanup; ret = 0; cleanup: if (fifo) av_audio_fifo_free(fifo); swr_free(&resample_context); if (output_codec_context) avcodec_close(output_codec_context); if (output_format_context) { avio_closep(&output_format_context->pb); avformat_free_context(output_format_context); } if (input_codec_context) avcodec_close(input_codec_context); if (input_format_context) avformat_close_input(&input_format_context); return ret; }