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| /* | |
| * Copyright (c) 2013-2018 Andreas Unterweger | |
| * | |
| * This file is part of Libav. | |
| * | |
| * Libav 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. | |
| * | |
| * Libav 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 Libav; 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 Libav. | |
| * Formats other than MP4 are supported based on the output file extension. | |
| * @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/avstring.h" | |
| #include "libavutil/frame.h" | |
| #include "libavutil/opt.h" | |
| #include "libavresample/avresample.h" | |
| /* The output bit rate in bit/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 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. | |
| * @param filename File to be opened | |
| * @param[out] input_format_context Format context of opened file | |
| * @param[out] input_codec_context Codec context of opened file | |
| * @return Error code (0 if successful) | |
| */ | |
| static int open_input_file(const char *filename, | |
| AVFormatContext **input_format_context, | |
| AVCodecContext **input_codec_context) | |
| { | |
| AVCodecContext *avctx; | |
| 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]->codecpar->codec_id))) { | |
| fprintf(stderr, "Could not find input codec\n"); | |
| avformat_close_input(input_format_context); | |
| return AVERROR_EXIT; | |
| } | |
| /* Allocate a new decoding context. */ | |
| avctx = avcodec_alloc_context3(input_codec); | |
| if (!avctx) { | |
| fprintf(stderr, "Could not allocate a decoding context\n"); | |
| avformat_close_input(input_format_context); | |
| return AVERROR(ENOMEM); | |
| } | |
| /* Initialize the stream parameters with demuxer information. */ | |
| error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar); | |
| if (error < 0) { | |
| avformat_close_input(input_format_context); | |
| avcodec_free_context(&avctx); | |
| return error; | |
| } | |
| /* Open the decoder for the audio stream to use it later. */ | |
| if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) { | |
| fprintf(stderr, "Could not open input codec (error '%s')\n", | |
| get_error_text(error)); | |
| avcodec_free_context(&avctx); | |
| avformat_close_input(input_format_context); | |
| return error; | |
| } | |
| /* Save the decoder context for easier access later. */ | |
| *input_codec_context = avctx; | |
| 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. | |
| * @param filename File to be opened | |
| * @param input_codec_context Codec context of input file | |
| * @param[out] output_format_context Format context of output file | |
| * @param[out] output_codec_context Codec context of output file | |
| * @return Error code (0 if successful) | |
| */ | |
| static int open_output_file(const char *filename, | |
| AVCodecContext *input_codec_context, | |
| AVFormatContext **output_format_context, | |
| AVCodecContext **output_codec_context) | |
| { | |
| AVCodecContext *avctx = NULL; | |
| 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, NULL))) { | |
| fprintf(stderr, "Could not create new stream\n"); | |
| error = AVERROR(ENOMEM); | |
| goto cleanup; | |
| } | |
| avctx = avcodec_alloc_context3(output_codec); | |
| if (!avctx) { | |
| fprintf(stderr, "Could not allocate an encoding context\n"); | |
| error = AVERROR(ENOMEM); | |
| goto cleanup; | |
| } | |
| /* Set the basic encoder parameters. | |
| * The input file's sample rate is used to avoid a sample rate conversion. */ | |
| avctx->channels = OUTPUT_CHANNELS; | |
| avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS); | |
| avctx->sample_rate = input_codec_context->sample_rate; | |
| avctx->sample_fmt = output_codec->sample_fmts[0]; | |
| avctx->bit_rate = OUTPUT_BIT_RATE; | |
| /* Allow the use of the experimental AAC encoder. */ | |
| avctx->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) | |
| avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; | |
| /* Open the encoder for the audio stream to use it later. */ | |
| if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) { | |
| fprintf(stderr, "Could not open output codec (error '%s')\n", | |
| get_error_text(error)); | |
| goto cleanup; | |
| } | |
| error = avcodec_parameters_from_context(stream->codecpar, avctx); | |
| if (error < 0) { | |
| fprintf(stderr, "Could not initialize stream parameters\n"); | |
| goto cleanup; | |
| } | |
| /* Save the encoder context for easier access later. */ | |
| *output_codec_context = avctx; | |
| return 0; | |
| cleanup: | |
| avcodec_free_context(&avctx); | |
| avio_close((*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. | |
| * @param packet Packet to be initialized | |
| */ | |
| 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. | |
| * @param[out] frame Frame to be initialized | |
| * @return Error code (0 if successful) | |
| */ | |
| 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 | |
| * libavresample takes care of this, but requires initialization. | |
| * @param input_codec_context Codec context of the input file | |
| * @param output_codec_context Codec context of the output file | |
| * @param[out] resample_context Resample context for the required conversion | |
| * @return Error code (0 if successful) | |
| */ | |
| static int init_resampler(AVCodecContext *input_codec_context, | |
| AVCodecContext *output_codec_context, | |
| AVAudioResampleContext **resample_context) | |
| { | |
| /* Only initialize the resampler if it is necessary, i.e., | |
| * if and only if the sample formats differ. */ | |
| if (input_codec_context->sample_fmt != output_codec_context->sample_fmt || | |
| input_codec_context->channels != output_codec_context->channels) { | |
| int error; | |
| /* Create a resampler context for the conversion. */ | |
| if (!(*resample_context = avresample_alloc_context())) { | |
| fprintf(stderr, "Could not allocate resample context\n"); | |
| return AVERROR(ENOMEM); | |
| } | |
| /* 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). | |
| */ | |
| av_opt_set_int(*resample_context, "in_channel_layout", | |
| av_get_default_channel_layout(input_codec_context->channels), 0); | |
| av_opt_set_int(*resample_context, "out_channel_layout", | |
| av_get_default_channel_layout(output_codec_context->channels), 0); | |
| av_opt_set_int(*resample_context, "in_sample_rate", | |
| input_codec_context->sample_rate, 0); | |
| av_opt_set_int(*resample_context, "out_sample_rate", | |
| output_codec_context->sample_rate, 0); | |
| av_opt_set_int(*resample_context, "in_sample_fmt", | |
| input_codec_context->sample_fmt, 0); | |
| av_opt_set_int(*resample_context, "out_sample_fmt", | |
| output_codec_context->sample_fmt, 0); | |
| /* Open the resampler with the specified parameters. */ | |
| if ((error = avresample_open(*resample_context)) < 0) { | |
| fprintf(stderr, "Could not open resample context\n"); | |
| avresample_free(resample_context); | |
| return error; | |
| } | |
| } | |
| return 0; | |
| } | |
| /** | |
| * Initialize a FIFO buffer for the audio samples to be encoded. | |
| * @param[out] fifo Sample buffer | |
| * @param output_codec_context Codec context of the output file | |
| * @return Error code (0 if successful) | |
| */ | |
| 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. | |
| * @param output_format_context Format context of the output file | |
| * @return Error code (0 if successful) | |
| */ | |
| 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. | |
| * @param frame Audio frame to be decoded | |
| * @param input_format_context Format context of the input file | |
| * @param input_codec_context Codec context of the input file | |
| * @param[out] data_present Indicates whether data has been decoded | |
| * @param[out] finished Indicates whether the end of file has | |
| * been reached and all data has been | |
| * decoded. If this flag is false, there | |
| * is more data to be decoded, i.e., this | |
| * function has to be called again. | |
| * @return Error code (0 if successful) | |
| */ | |
| 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 the 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; | |
| } | |
| } | |
| /* Send the audio frame stored in the temporary packet to the decoder. | |
| * The input audio stream decoder is used to do this. */ | |
| if ((error = avcodec_send_packet(input_codec_context, &input_packet)) < 0) { | |
| fprintf(stderr, "Could not send packet for decoding (error '%s')\n", | |
| get_error_text(error)); | |
| return error; | |
| } | |
| /* Receive one frame from the decoder. */ | |
| error = avcodec_receive_frame(input_codec_context, frame); | |
| /* If the decoder asks for more data to be able to decode a frame, | |
| * return indicating that no data is present. */ | |
| if (error == AVERROR(EAGAIN)) { | |
| error = 0; | |
| goto cleanup; | |
| /* If the end of the input file is reached, stop decoding. */ | |
| } else if (error == AVERROR_EOF) { | |
| *finished = 1; | |
| error = 0; | |
| goto cleanup; | |
| } else if (error < 0) { | |
| fprintf(stderr, "Could not decode frame (error '%s')\n", | |
| get_error_text(error)); | |
| goto cleanup; | |
| /* Default case: Return decoded data. */ | |
| } else { | |
| *data_present = 1; | |
| goto cleanup; | |
| } | |
| cleanup: | |
| av_packet_unref(&input_packet); | |
| return error; | |
| } | |
| /** | |
| * 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. | |
| * @param[out] converted_input_samples Array of converted samples. The | |
| * dimensions are reference, channel | |
| * (for multi-channel audio), sample. | |
| * @param output_codec_context Codec context of the output file | |
| * @param frame_size Number of samples to be converted in | |
| * each round | |
| * @return Error code (0 if successful) | |
| */ | |
| 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. | |
| * @param input_data Samples to be decoded. The dimensions are | |
| * channel (for multi-channel audio), sample. | |
| * @param[out] converted_data Converted samples. The dimensions are channel | |
| * (for multi-channel audio), sample. | |
| * @param frame_size Number of samples to be converted | |
| * @param resample_context Resample context for the conversion | |
| * @return Error code (0 if successful) | |
| */ | |
| static int convert_samples(uint8_t **input_data, | |
| uint8_t **converted_data, const int frame_size, | |
| AVAudioResampleContext *resample_context) | |
| { | |
| int error; | |
| /* Convert the samples using the resampler. */ | |
| if ((error = avresample_convert(resample_context, converted_data, 0, | |
| frame_size, input_data, 0, frame_size)) < 0) { | |
| fprintf(stderr, "Could not convert input samples (error '%s')\n", | |
| get_error_text(error)); | |
| return error; | |
| } | |
| /* 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. */ | |
| if (avresample_available(resample_context)) { | |
| fprintf(stderr, "Converted samples left over\n"); | |
| return AVERROR_EXIT; | |
| } | |
| return 0; | |
| } | |
| /** | |
| * Add converted input audio samples to the FIFO buffer for later processing. | |
| * @param fifo Buffer to add the samples to | |
| * @param converted_input_samples Samples to be added. The dimensions are channel | |
| * (for multi-channel audio), sample. | |
| * @param frame_size Number of samples to be converted | |
| * @return Error code (0 if successful) | |
| */ | |
| 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, decode, convert and store | |
| * it in the FIFO buffer. | |
| * @param fifo Buffer used for temporary storage | |
| * @param input_format_context Format context of the input file | |
| * @param input_codec_context Codec context of the input file | |
| * @param output_codec_context Codec context of the output file | |
| * @param resample_context Resample context for the conversion | |
| * @param[out] finished Indicates whether the end of file has | |
| * been reached and all data has been | |
| * decoded. If this flag is false, | |
| * there is more data to be decoded, | |
| * i.e., this function has to be called | |
| * again. | |
| * @return Error code (0 if successful) | |
| */ | |
| static int read_decode_convert_and_store(AVAudioFifo *fifo, | |
| AVFormatContext *input_format_context, | |
| AVCodecContext *input_codec_context, | |
| AVCodecContext *output_codec_context, | |
| AVAudioResampleContext *resample_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 = 0; | |
| 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) { | |
| 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(input_frame->extended_data, converted_input_samples, | |
| input_frame->nb_samples, resample_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. | |
| * @param[out] frame Frame to be initialized | |
| * @param output_codec_context Codec context of the output file | |
| * @param frame_size Size of the frame | |
| * @return Error code (0 if successful) | |
| */ | |
| 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 not 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. | |
| * @param frame Samples to be encoded | |
| * @param output_format_context Format context of the output file | |
| * @param output_codec_context Codec context of the output file | |
| * @param[out] data_present Indicates whether data has been | |
| * encoded | |
| * @return Error code (0 if successful) | |
| */ | |
| 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; | |
| } | |
| /* Send the audio frame stored in the temporary packet to the encoder. | |
| * The output audio stream encoder is used to do this. */ | |
| error = avcodec_send_frame(output_codec_context, frame); | |
| /* The encoder signals that it has nothing more to encode. */ | |
| if (error == AVERROR_EOF) { | |
| error = 0; | |
| goto cleanup; | |
| } else if (error < 0) { | |
| fprintf(stderr, "Could not send packet for encoding (error '%s')\n", | |
| get_error_text(error)); | |
| return error; | |
| } | |
| /* Receive one encoded frame from the encoder. */ | |
| error = avcodec_receive_packet(output_codec_context, &output_packet); | |
| /* If the encoder asks for more data to be able to provide an | |
| * encoded frame, return indicating that no data is present. */ | |
| if (error == AVERROR(EAGAIN)) { | |
| error = 0; | |
| goto cleanup; | |
| /* If the last frame has been encoded, stop encoding. */ | |
| } else if (error == AVERROR_EOF) { | |
| error = 0; | |
| goto cleanup; | |
| } else if (error < 0) { | |
| fprintf(stderr, "Could not encode frame (error '%s')\n", | |
| get_error_text(error)); | |
| goto cleanup; | |
| /* Default case: Return encoded data. */ | |
| } else { | |
| *data_present = 1; | |
| } | |
| /* Write one audio frame from the temporary packet to the output file. */ | |
| if (*data_present && | |
| (error = av_write_frame(output_format_context, &output_packet)) < 0) { | |
| fprintf(stderr, "Could not write frame (error '%s')\n", | |
| get_error_text(error)); | |
| goto cleanup; | |
| } | |
| cleanup: | |
| av_packet_unref(&output_packet); | |
| return error; | |
| } | |
| /** | |
| * Load one audio frame from the FIFO buffer, encode and write it to the | |
| * output file. | |
| * @param fifo Buffer used for temporary storage | |
| * @param output_format_context Format context of the output file | |
| * @param output_codec_context Codec context of the output file | |
| * @return Error code (0 if successful) | |
| */ | |
| 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. | |
| * @param output_format_context Format context of the output file | |
| * @return Error code (0 if successful) | |
| */ | |
| 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; | |
| } | |
| int main(int argc, char **argv) | |
| { | |
| AVFormatContext *input_format_context = NULL, *output_format_context = NULL; | |
| AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL; | |
| AVAudioResampleContext *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 { | |
| data_written = 0; | |
| 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); | |
| if (resample_context) { | |
| avresample_close(resample_context); | |
| avresample_free(&resample_context); | |
| } | |
| if (output_codec_context) | |
| avcodec_free_context(&output_codec_context); | |
| if (output_format_context) { | |
| avio_close(output_format_context->pb); | |
| avformat_free_context(output_format_context); | |
| } | |
| if (input_codec_context) | |
| avcodec_free_context(&input_codec_context); | |
| if (input_format_context) | |
| avformat_close_input(&input_format_context); | |
| return ret; | |
| } |