diff options
author | James Almer <jamrial@gmail.com> | 2017-10-26 22:58:39 +0300 |
---|---|---|
committer | James Almer <jamrial@gmail.com> | 2017-10-26 23:03:09 +0300 |
commit | 8a3d3b624075c56b844cbd0f3d58c6e291c8e18c (patch) | |
tree | e02685a321c1cce5613ede25958bcab45c18b923 /doc/examples/transcode_aac.c | |
parent | 7066a92f87a418842b9c3ce9995e9c91370f4b62 (diff) | |
parent | b200a2c8da403b5a5c8b50f8cb4a75fd4f0131b1 (diff) |
Merge commit 'b200a2c8da403b5a5c8b50f8cb4a75fd4f0131b1'
* commit 'b200a2c8da403b5a5c8b50f8cb4a75fd4f0131b1':
examples: Fixed and extended Doxygen documentation
Merged-by: James Almer <jamrial@gmail.com>
Diffstat (limited to 'doc/examples/transcode_aac.c')
-rw-r--r-- | doc/examples/transcode_aac.c | 361 |
1 files changed, 209 insertions, 152 deletions
diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c index 6c2f4fb475..9fd5c00d60 100644 --- a/doc/examples/transcode_aac.c +++ b/doc/examples/transcode_aac.c @@ -1,4 +1,6 @@ /* + * Copyright (c) 2013-2017 Andreas Unterweger + * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or @@ -8,7 +10,7 @@ * * 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 + * 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 @@ -18,10 +20,11 @@ /** * @file - * simple audio converter + * Simple audio converter * * @example transcode_aac.c * Convert an input audio file to AAC in an MP4 container using FFmpeg. + * Formats other than MP4 are supported based on the output file extension. * @author Andreas Unterweger (dustsigns@gmail.com) */ @@ -40,12 +43,18 @@ #include "libswresample/swresample.h" -/** The output bit rate in kbit/s */ +/* The output bit rate in bit/s */ #define OUTPUT_BIT_RATE 96000 -/** The number of output channels */ +/* The number of output channels */ #define OUTPUT_CHANNELS 2 -/** Open an input file and the required decoder. */ +/** + * 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) @@ -54,7 +63,7 @@ static int open_input_file(const char *filename, AVCodec *input_codec; int error; - /** Open the input file to read from it. */ + /* 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", @@ -63,7 +72,7 @@ static int open_input_file(const char *filename, return error; } - /** Get information on the input file (number of streams etc.). */ + /* 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", av_err2str(error)); @@ -71,7 +80,7 @@ static int open_input_file(const char *filename, return error; } - /** Make sure that there is only one stream in the input file. */ + /* 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); @@ -79,14 +88,14 @@ static int open_input_file(const char *filename, return AVERROR_EXIT; } - /** Find a decoder for the audio stream. */ + /* 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 */ + /* Allocate a new decoding context. */ avctx = avcodec_alloc_context3(input_codec); if (!avctx) { fprintf(stderr, "Could not allocate a decoding context\n"); @@ -94,7 +103,7 @@ static int open_input_file(const char *filename, return AVERROR(ENOMEM); } - /** initialize the stream parameters with demuxer information */ + /* 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); @@ -102,7 +111,7 @@ static int open_input_file(const char *filename, return error; } - /** Open the decoder for the audio stream to use it later. */ + /* 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", av_err2str(error)); @@ -111,7 +120,7 @@ static int open_input_file(const char *filename, return error; } - /** Save the decoder context for easier access later. */ + /* Save the decoder context for easier access later. */ *input_codec_context = avctx; return 0; @@ -121,6 +130,11 @@ static int open_input_file(const char *filename, * 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, @@ -133,7 +147,7 @@ static int open_output_file(const char *filename, AVCodec *output_codec = NULL; int error; - /** Open the output file to write to it. */ + /* 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", @@ -141,16 +155,16 @@ static int open_output_file(const char *filename, return error; } - /** Create a new format context for the output container format. */ + /* 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. */ + /* 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. */ + /* 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"); @@ -160,13 +174,13 @@ static int open_output_file(const char *filename, av_strlcpy((*output_format_context)->filename, filename, sizeof((*output_format_context)->filename)); - /** Find the encoder to be used by its name. */ + /* 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. */ + /* 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); @@ -180,31 +194,27 @@ static int open_output_file(const char *filename, goto cleanup; } - /** - * Set the basic encoder parameters. - * The input file's sample rate is used to avoid a sample rate conversion. - */ + /* 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 */ + /* Allow the use of the experimental AAC encoder. */ avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL; - /** Set the sample rate for the container. */ + /* 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. - */ + /* 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. */ + /* 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", av_err2str(error)); @@ -217,7 +227,7 @@ static int open_output_file(const char *filename, goto cleanup; } - /** Save the encoder context for easier access later. */ + /* Save the encoder context for easier access later. */ *output_codec_context = avctx; return 0; @@ -230,16 +240,23 @@ cleanup: return error < 0 ? error : AVERROR_EXIT; } -/** Initialize one data packet for reading or writing. */ +/** + * 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. */ + /* 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 */ +/** + * 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())) { @@ -253,6 +270,10 @@ static int init_input_frame(AVFrame **frame) * 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. + * @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, @@ -260,7 +281,7 @@ static int init_resampler(AVCodecContext *input_codec_context, { int error; - /** + /* * Create a resampler context for the conversion. * Set the conversion parameters. * Default channel layouts based on the number of channels @@ -279,14 +300,14 @@ static int init_resampler(AVCodecContext *input_codec_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. */ + /* 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); @@ -295,10 +316,15 @@ static int init_resampler(AVCodecContext *input_codec_context, return 0; } -/** Initialize a FIFO buffer for the audio samples to be encoded. */ +/** + * 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. */ + /* 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"); @@ -307,7 +333,11 @@ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context) return 0; } -/** Write the header of the output file container. */ +/** + * 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; @@ -319,20 +349,32 @@ static int write_output_file_header(AVFormatContext *output_format_context) return 0; } -/** Decode one audio frame from the input file. */ +/** + * 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. */ + /* 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. */ + /* 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 we are at the end of the file, flush the decoder below. */ if (error == AVERROR_EOF) *finished = 1; else { @@ -342,12 +384,10 @@ static int decode_audio_frame(AVFrame *frame, } } - /** - * Decode the audio frame stored in the temporary packet. + /* 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. - */ + * 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", @@ -356,10 +396,8 @@ static int decode_audio_frame(AVFrame *frame, return error; } - /** - * If the decoder has not been flushed completely, we are not finished, - * so that this function has to be called again. - */ + /* 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); @@ -370,6 +408,13 @@ static int decode_audio_frame(AVFrame *frame, * 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, @@ -377,8 +422,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples, { int error; - /** - * Allocate as many pointers as there are audio channels. + /* 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). */ @@ -388,10 +432,8 @@ static int init_converted_samples(uint8_t ***converted_input_samples, return AVERROR(ENOMEM); } - /** - * Allocate memory for the samples of all channels in one consecutive - * block for convenience. - */ + /* 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, @@ -408,8 +450,15 @@ static int init_converted_samples(uint8_t ***converted_input_samples, /** * 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. + * 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(const uint8_t **input_data, uint8_t **converted_data, const int frame_size, @@ -417,7 +466,7 @@ static int convert_samples(const uint8_t **input_data, { int error; - /** Convert the samples using the resampler. */ + /* Convert the samples using the resampler. */ if ((error = swr_convert(resample_context, converted_data, frame_size, input_data , frame_size)) < 0) { @@ -429,23 +478,28 @@ static int convert_samples(const uint8_t **input_data, return 0; } -/** Add converted input audio samples to the FIFO buffer for later processing. */ +/** + * 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. - */ + /* 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. */ + /* 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"); @@ -455,8 +509,20 @@ static int add_samples_to_fifo(AVAudioFifo *fifo, } /** - * Read one audio frame from the input file, decodes, converts and stores + * 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 resampler_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, @@ -465,45 +531,41 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo, SwrContext *resampler_context, int *finished) { - /** Temporary storage of the input samples of the frame read from the file. */ + /* Temporary storage of the input samples of the frame read from the file. */ AVFrame *input_frame = NULL; - /** Temporary storage for the converted input samples. */ + /* 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. */ + /* Initialize temporary storage for one input frame. */ if (init_input_frame(&input_frame)) goto cleanup; - /** Decode one frame worth of audio samples. */ + /* 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 + /* 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. - */ + * 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 there is decoded data, convert and store it. */ if (data_present) { - /** Initialize the temporary storage for the converted input samples. */ + /* 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. - */ + /* 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. */ + /* 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; @@ -524,6 +586,10 @@ cleanup: /** * 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, @@ -531,28 +597,24 @@ static int init_output_frame(AVFrame **frame, { int error; - /** Create a new frame to store the audio samples. */ + /* 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. + /* 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. - */ + * 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. - */ + /* 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", av_err2str(error)); @@ -563,30 +625,36 @@ static int init_output_frame(AVFrame **frame, return 0; } -/** Global timestamp for the audio frames */ +/* Global timestamp for the audio frames. */ static int64_t pts = 0; -/** Encode one frame worth of audio to the output file. */ +/** + * 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 + * decoded + * @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. */ + /* 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. */ + /* 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. - */ + /* 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", @@ -595,7 +663,7 @@ static int encode_audio_frame(AVFrame *frame, return error; } - /** Write one audio frame from the temporary packet to the output file. */ + /* 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", @@ -613,37 +681,37 @@ static int encode_audio_frame(AVFrame *frame, /** * 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. */ + /* 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. + /* 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 - */ + * 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. */ + /* 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. - */ + /* 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. */ + /* 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); @@ -653,7 +721,11 @@ static int load_encode_and_write(AVAudioFifo *fifo, return 0; } -/** Write the trailer of the output file container. */ +/** + * 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; @@ -665,7 +737,6 @@ static int write_output_file_trailer(AVFormatContext *output_format_context) 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; @@ -674,89 +745,75 @@ int main(int argc, char **argv) AVAudioFifo *fifo = NULL; int ret = AVERROR_EXIT; - if (argc < 3) { + 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. */ + /* Register all codecs and formats so that they can be used. */ av_register_all(); - /** Open the input file for reading. */ + /* 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. */ + /* 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. */ + /* 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. */ + /* 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. */ + /* 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. - */ + /* 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. */ + /* 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 + /* 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. - */ + * 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. - */ + /* 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 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. + /* 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. - */ + * 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. - */ + /* 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 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. */ + /* Flush the encoder as it may have delayed frames. */ do { if (encode_audio_frame(NULL, output_format_context, output_codec_context, &data_written)) @@ -766,7 +823,7 @@ int main(int argc, char **argv) } } - /** Write the trailer of the output file container. */ + /* Write the trailer of the output file container. */ if (write_output_file_trailer(output_format_context)) goto cleanup; ret = 0; |