RepoMirrors/ffmpeg
1
0
Fork 0
mirror of https://git.ffmpeg.org/ffmpeg.git synced 2024-12-29 10:52:20 +00:00
Code Issues Projects Releases Wiki Activity

examples: Fixed and extended Doxygen documentation

Browse Source 
Added parameter descriptions for all functions
 and converted in-function comments into regular
 (non-Doxygen) comments.

Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This commit is contained in:
Andreas Unterweger 2017-04-10 13:06:18 +02:00 committed by Vittorio Giovara
parent efddf2c09a
commit b200a2c8da
1 changed files with 215 additions and 163 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

378
doc/examples/transcode_aac.c
View File

@ -1,4 +1,6 @@
/*
* Copyright (c) 2013-2017 Andreas Unterweger
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
@ -8,7 +10,7 @@
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* 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 Libav.
* Formats other than MP4 are supported based on the output file extension.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/
@ -39,9 +42,9 @@
#include "libavresample/avresample.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
/**
@ -56,7 +59,13 @@ static char *get_error_text(const int error)
return error_buffer;
}
/** 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)
@ -65,7 +74,7 @@ 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",
@ -74,7 +83,7 @@ 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",
get_error_text(error));
@ -82,7 +91,7 @@ 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);
@ -90,14 +99,14 @@ 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");
@ -105,7 +114,7 @@ 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);
@ -113,7 +122,7 @@ 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",
get_error_text(error));
@ -122,7 +131,7 @@ 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;
@ -132,6 +141,11 @@ 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,
@ -144,7 +158,7 @@ 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",
@ -152,16 +166,16 @@ 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");
@ -171,13 +185,13 @@ 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);
@ -191,31 +205,27 @@ 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",
get_error_text(error));
@ -228,7 +238,7 @@ 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;
@ -241,16 +251,23 @@ 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())) {
@ -264,27 +281,28 @@ 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
* libavresample takes care of this, but requires initialization.
* @param input_codec_context Codec context of the input file
* @param output_codec_context Codec context of the output file
* @param[out] resample_context Resample context for the required conversion
* @return Error code (0 if successful)
*/
static int init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
AVAudioResampleContext **resample_context)
{
/**
* Only initialize the resampler if it is necessary, i.e.,
* if and only if the sample formats differ.
*/
/* Only initialize the resampler if it is necessary, i.e.,
* if and only if the sample formats differ. */
if (input_codec_context->sample_fmt != output_codec_context->sample_fmt ||
input_codec_context->channels != output_codec_context->channels) {
int error;
/** Create a resampler context for the conversion. */
/* Create a resampler context for the conversion. */
if (!(*resample_context = avresample_alloc_context())) {
fprintf(stderr, "Could not allocate resample context\n");
return AVERROR(ENOMEM);
}
/**
* Set the conversion parameters.
/* 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).
@ -302,7 +320,7 @@ static int init_resampler(AVCodecContext *input_codec_context,
av_opt_set_int(*resample_context, "out_sample_fmt",
output_codec_context->sample_fmt, 0);
/** Open the resampler with the specified parameters. */
/* Open the resampler with the specified parameters. */
if ((error = avresample_open(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
avresample_free(resample_context);
@ -312,10 +330,15 @@ 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");
@ -324,7 +347,11 @@ 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;
@ -336,20 +363,32 @@ 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 the the end of the file, flush the decoder below. */
/* If we are the the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {
@ -359,12 +398,10 @@ 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",
@ -373,10 +410,8 @@ 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);
@ -387,6 +422,13 @@ 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,
@ -394,8 +436,7 @@ 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).
*/
@ -405,10 +446,8 @@ 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,
@ -425,8 +464,15 @@ 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(uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
@ -434,7 +480,7 @@ static int convert_samples(uint8_t **input_data,
{
int error;
/** Convert the samples using the resampler. */
/* Convert the samples using the resampler. */
if ((error = avresample_convert(resample_context, converted_data, 0,
frame_size, input_data, 0, frame_size)) < 0) {
fprintf(stderr, "Could not convert input samples (error '%s')\n",
@ -442,11 +488,9 @@ static int convert_samples(uint8_t **input_data,
return error;
}
/**
* Perform a sanity check so that the number of converted samples is
/* Perform a sanity check so that the number of converted samples is
* not greater than the number of samples to be converted.
* If the sample rates differ, this case has to be handled differently
*/
* If the sample rates differ, this case has to be handled differently. */
if (avresample_available(resample_context)) {
fprintf(stderr, "Converted samples left over\n");
return AVERROR_EXIT;
@ -455,23 +499,28 @@ static int convert_samples(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");
@ -481,55 +530,63 @@ 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 resample_context Resample context for the conversion
* @param[out] finished Indicates whether the end of file has
* been reached and all data has been
* decoded. If this flag is false,
* there is more data to be decoded,
* i.e., this function has to be called
* again.
* @return Error code (0 if successful)
*/
static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
AVAudioResampleContext *resampler_context,
AVAudioResampleContext *resample_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(input_frame->extended_data, converted_input_samples,
input_frame->nb_samples, resampler_context))
input_frame->nb_samples, resample_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;
@ -550,6 +607,10 @@ 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,
@ -557,28 +618,24 @@ 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",
get_error_text(error));
@ -589,30 +646,36 @@ 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",
@ -621,7 +684,7 @@ 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",
@ -639,37 +702,37 @@ 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);
@ -679,7 +742,11 @@ 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;
@ -691,7 +758,6 @@ 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;
@ -700,89 +766,75 @@ 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))
@ -792,7 +844,7 @@ 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;