Merge commit 'b200a2c8da403b5a5c8b50f8cb4a75fd4f0131b1'

* commit 'b200a2c8da403b5a5c8b50f8cb4a75fd4f0131b1':
  examples: Fixed and extended Doxygen documentation

Merged-by: James Almer <jamrial@gmail.com>

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;