| /* |
| * 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_packet_unref(&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_packet_unref(&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_packet_unref(&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_packet_unref(&output_packet); |
| return error; |
| } |
| |
| av_packet_unref(&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; |
| } |