总结
这个项目代码比较长.但是结构很清晰,用来转换音频数据,从输入文件中读取音频.然后初始化输入输出的编解码器,格式上下文,还使用了一个buffer缓冲区.然后解码音频数据,进行转换,转换完成后,把数据送入fifo缓冲区中,然后在从fifo中取出原始帧.进行编码.写出到文件中,这里形成了一个单一的生产者消费者的模式,中间通过fifo缓冲区.
流程图
初始化格式上下文,编解码上下文
分别初始化输入和输出的
graph TB
main{main}
-->oif[open_input_file]
-->afoi[avformat_open_input]
-->affsi[avformat_find_stream_info]
-->afd[avcodec_find_decoder]
-->acac[avcodec_alloc_context3]
-->acptc[avcodec_parameters_to_context]
-->aco[avcodec_open2]
main-->oof[open_output_file]
-->ao[avio_open]
-->afac[avformat_alloc_context]
-->agf[av_guess_format]
-->acfe[avcodec_find_encoder]
-->afns[avformat_new_stream]
-->acaco[avcodec_alloc_context3]
-->acoo[avcodec_open2]
-->acpfc[avcodec_parameters_from_context]
到这里输入输出的编解码器上下文,和格式上下文已经初始化完成,这个代码太长了.分段处理
初始化转换上下文及buf
graph TB
ir[init_resampler]
-->saso[swr_alloc_set_opts]
-->si[swr_init]
-->aafa[av_audio_fifo_alloc]
-->adcas[read_decode_convert_and_store]
-->leaw[load_encode_and_write]
-->iopf[init_output_frame]
-->aaffr[av_audio_fifo_read]
-->eaf[encode_audio_frame]
-->aip[av_init_packet]
-->acsf[avcodec_send_frame]
-->acrp[avcodec_receive_packet]
-->awf[av_write_frame]
-->woft[write_output_file_trailer]
-->release[release]
adca[read_decode_convert_and_store]
-->afa[av_frame_alloc]
-->ip[av_init_packet]
-->arf[av_read_frame]
-->acsp[avcodec_send_packet]
-->acrf[avcodec_receive_frame]
-->apu[av_packet_unref]
-->ics[init_converted_samples]
-->sw[swr_convert]
-->astf[add_samples_to_fifo]
-->aafr[av_audio_fifo_realloc]
-->aafw[av_audio_fifo_write]
-->free[release]
这里就是初始化了转换上下文和fifo的缓冲区,把输入数据解码后送入fifo缓冲区,然后在从fifo缓冲区中读入数据到frame中,然后送入编码器编码,便完成后写到输出格式上下文,最后在写出尾部数据. 最后在关闭各种结构和缓冲.
这里有几个while循环我给忽略了.有些细节需要慢慢理解.
代码
/**
* @file
* 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)
*/
#include <stdio.h>
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavcodec/avcodec.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
#include "libswresample/swresample.h"
/* The output bit rate in bit/s */
#define OUTPUT_BIT_RATE 96000
/* The number of output channels */
#define OUTPUT_CHANNELS 2
/**
* Open an input file and the required decoder.
* @param filename File to be opened
* @param[out] input_format_context Format context of opened file
* @param[out] input_codec_context Codec context of opened file
* @return Error code (0 if successful)
*/
static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context) {
AVCodecContext *avctx;
AVCodec *input_codec;
int error;
//初始化输入格式上下文
/* Open the input file to read from it. */
if ((error = avformat_open_input(input_format_context, filename, NULL,
NULL)) < 0) {
fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
filename, av_err2str(error));
*input_format_context = NULL;
return error;
}
//找到流信息
/* 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));
avformat_close_input(input_format_context);
return error;
}
/* Make sure that there is only one stream in the input file. */
if ((*input_format_context)->nb_streams != 1) {
fprintf(stderr, "Expected one audio input stream, but found %d\n",
(*input_format_context)->nb_streams);
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
//找到输入流的解码器,根据解码参数
/* 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;
}
//分配解码器上下文
/* Allocate a new decoding context. */
avctx = avcodec_alloc_context3(input_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate a decoding context\n");
avformat_close_input(input_format_context);
return AVERROR(ENOMEM);
}
//拷贝输入格式中的参数到解码器上下文中
/* Initialize the stream parameters with demuxer information. */
error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
if (error < 0) {
avformat_close_input(input_format_context);
avcodec_free_context(&avctx);
return error;
}
//打开解码器上下文
/* 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));
avcodec_free_context(&avctx);
avformat_close_input(input_format_context);
return error;
}
/* Save the decoder context for easier access later. */
*input_codec_context = avctx;
return 0;
}
/**
* 打开输出文件,初始化编码器上下文
* Open an output file and the required encoder.
* Also set some basic encoder parameters.
* Some of these parameters are based on the input file's parameters.
* @param filename File to be opened
* @param input_codec_context Codec context of input file
* @param[out] output_format_context Format context of output file
* @param[out] output_codec_context Codec context of output file
* @return Error code (0 if successful)
*/
static int open_output_file(const char *filename,
AVCodecContext *input_codec_context,
AVFormatContext **output_format_context,
AVCodecContext **output_codec_context) {
AVCodecContext *avctx = NULL;
AVIOContext *output_io_context = NULL;
AVStream *stream = NULL;
AVCodec *output_codec = NULL;
int error;
//打开输出文件.创建io上下文
/* Open the output file to write to it. */
if ((error = avio_open(&output_io_context, filename,
AVIO_FLAG_WRITE)) < 0) {
fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
filename, av_err2str(error));
return error;
}
//输出格式上下文,分配控件
/* 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);
}
//保存输出io上下文
/* Associate the output file (pointer) with the container format context. */
(*output_format_context)->pb = output_io_context;
//根据文件名猜猜输出格式
/* Guess the desired container format based on the file extension. */
if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
NULL))) {
fprintf(stderr, "Could not find output file format\n");
goto cleanup;
}
//保存文件名
if (!((*output_format_context)->url = av_strdup(filename))) {
fprintf(stderr, "Could not allocate url.\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
//找到指定编码器
/* 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;
}
//从输出格式中创建 输出流
/* Create a new audio stream in the output file container. */
if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
//用编码器初始化输出编码器上下文
avctx = avcodec_alloc_context3(output_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate an encoding context\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
//填充输出上下文的参数,这里看到,输出的编码器上下文参数是自己指定的,而输入的解码器的参数则来自文件的流中的参数
/* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion. */
avctx->channels = OUTPUT_CHANNELS;
avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
avctx->sample_rate = input_codec_context->sample_rate;
avctx->sample_fmt = output_codec->sample_fmts[0];
avctx->bit_rate = OUTPUT_BIT_RATE;
/* Allow the use of the experimental AAC encoder. */
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/* Set the sample rate for the container. */
stream->time_base.den = input_codec_context->sample_rate;
stream->time_base.num = 1;
/* Some container formats (like MP4) require global headers to be present.
* Mark the encoder so that it behaves accordingly. */
if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/* Open the encoder for the audio stream to use it later. */
//在用输出的编码器打开输出编码器上下文的参数
if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
av_err2str(error));
goto cleanup;
}
// 用编码器上下文的参数填充输出流参数
error = avcodec_parameters_from_context(stream->codecpar, avctx);
if (error < 0) {
fprintf(stderr, "Could not initialize stream parameters\n");
goto cleanup;
}
/* Save the encoder context for easier access later. */
*output_codec_context = avctx;
return 0;
cleanup:
avcodec_free_context(&avctx);
avio_closep(&(*output_format_context)->pb);
avformat_free_context(*output_format_context);
*output_format_context = NULL;
return error < 0 ? error : AVERROR_EXIT;
}
/**
* Initialize one data packet for reading or writing.
* @param packet Packet to be initialized
*/
static void init_packet(AVPacket *packet) {
av_init_packet(packet);
/* Set the packet data and size so that it is recognized as being empty. */
packet->data = NULL;
packet->size = 0;
}
/**
* 初始化一个帧
* Initialize one audio frame for reading from the input file.
* @param[out] frame Frame to be initialized
* @return Error code (0 if successful)
*/
static int init_input_frame(AVFrame **frame) {
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate input frame\n");
return AVERROR(ENOMEM);
}
return 0;
}
/**
* 用输入输出的编解码上下文来生产转换上下文
* 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,
SwrContext **resample_context) {
int error;
/*
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
* are assumed for simplicity (they are sometimes not detected
* properly by the demuxer and/or decoder).
*/
//根据输出输入的编解码器来分配 转换上下文的参数
*resample_context = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(
output_codec_context->channels),
output_codec_context->sample_fmt,
output_codec_context->sample_rate,
av_get_default_channel_layout(
input_codec_context->channels),
input_codec_context->sample_fmt,
input_codec_context->sample_rate,
0, NULL);
if (!*resample_context) {
fprintf(stderr, "Could not allocate resample context\n");
return AVERROR(ENOMEM);
}
/*
* Perform a sanity check so that the number of converted samples is
* not greater than the number of samples to be converted.
* If the sample rates differ, this case has to be handled differently
*/
av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
/* Open the resampler with the specified parameters. */
//参数设置完成了. 然后初始化上下文
if ((error = swr_init(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
swr_free(resample_context);
return error;
}
return 0;
}
/**
* 初始化一个先进先出的缓冲区,针对输出格式的大小
* 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. */
//根据输出参数返回创建一个先进先出的缓存
if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
output_codec_context->channels, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
return AVERROR(ENOMEM);
}
return 0;
}
/**
* 写出输出文件的头信息
* Write the header of the output file container.
* @param output_format_context Format context of the output file
* @return Error code (0 if successful)
*/
static int write_output_file_header(AVFormatContext *output_format_context) {
int error;
if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
fprintf(stderr, "Could not write output file header (error '%s')\n",
av_err2str(error));
return error;
}
return 0;
}
/**
* Decode one audio frame from the input file.
* @param frame Audio frame to be decoded
* @param input_format_context Format context of the input file
* @param input_codec_context Codec context of the input file
* @param[out] data_present Indicates whether data has been decoded
* @param[out] finished Indicates whether the end of file has
* been reached and all data has been
* decoded. If this flag is false, there
* is more data to be decoded, i.e., this
* function has to be called again.
* @return Error code (0 if successful)
* 从输入文件中解码音频真
*/
static int decode_audio_frame(AVFrame *frame,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
int *data_present, int *finished) {
/* Packet used for temporary storage. */
AVPacket input_packet;
int error;
init_packet(&input_packet);
//读取数据到packet中
/* Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
/* If we are at the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {
fprintf(stderr, "Could not read frame (error '%s')\n",
av_err2str(error));
return error;
}
}
//packet数据送入解码器
/* Send the audio frame stored in the temporary packet to the decoder.
* The input audio stream decoder is used to do this. */
if ((error = avcodec_send_packet(input_codec_context, &input_packet)) < 0) {
fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
av_err2str(error));
return error;
}
//获取解码后的frame
/* Receive one frame from the decoder. */
error = avcodec_receive_frame(input_codec_context, frame);
/* If the decoder asks for more data to be able to decode a frame,
* return indicating that no data is present. */
if (error == AVERROR(EAGAIN)) {
error = 0;
goto cleanup;
/* If the end of the input file is reached, stop decoding. */
} else if (error == AVERROR_EOF) {
*finished = 1;
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not decode frame (error '%s')\n",
av_err2str(error));
goto cleanup;
/* Default case: Return decoded data. */
} else {
*data_present = 1;
goto cleanup;
}
cleanup:
av_packet_unref(&input_packet);
return error;
}
/**
* 分配个临时用的存储区域,
* Initialize a temporary storage for the specified number of audio samples.
* The conversion requires temporary storage due to the different format.
* The number of audio samples to be allocated is specified in frame_size.
* @param[out] converted_input_samples Array of converted samples. The
* dimensions are reference, channel
* (for multi-channel audio), sample.
* @param output_codec_context Codec context of the output file
* @param frame_size Number of samples to be converted in
* each round
* @return Error code (0 if successful)
*/
static int init_converted_samples(uint8_t ***converted_input_samples,
AVCodecContext *output_codec_context,
int frame_size) {
int error;
/* Allocate as many pointers as there are audio channels.
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
*/
if (!(*converted_input_samples = calloc(output_codec_context->channels,
sizeof(**converted_input_samples)))) {
fprintf(stderr, "Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
/* Allocate memory for the samples of all channels in one consecutive
* block for convenience. */
//给数组分配空间缓冲
if ((error = av_samples_alloc(*converted_input_samples, NULL,
output_codec_context->channels,
frame_size,
output_codec_context->sample_fmt, 0)) < 0) {
fprintf(stderr,
"Could not allocate converted input samples (error '%s')\n",
av_err2str(error));
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return error;
}
return 0;
}
/**
* 转换输入数据到输出数据,根据frame_size转换
* 数据从input_data 到 convert_data
* Convert the input audio samples into the output sample format.
* The conversion happens on a per-frame basis, the size of which is
* specified by frame_size.
* @param input_data Samples to be decoded. The dimensions are
* channel (for multi-channel audio), sample.
* @param[out] converted_data Converted samples. The dimensions are channel
* (for multi-channel audio), sample.
* @param frame_size Number of samples to be converted
* @param resample_context Resample context for the conversion
* @return Error code (0 if successful)
*/
static int convert_samples(const uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
SwrContext *resample_context) {
int error;
/* Convert the samples using the resampler. */
if ((error = swr_convert(resample_context,
converted_data, frame_size,
input_data, frame_size)) < 0) {
fprintf(stderr, "Could not convert input samples (error '%s')\n",
av_err2str(error));
return error;
}
return 0;
}
/**
* 把转换后的数据写入到fifo 的缓冲区中
* 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. */
//扩充fifo.保证数据能放下.
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. */
//把输入存入 fifo中
if (av_audio_fifo_write(fifo, (void **) converted_input_samples,
frame_size) < frame_size) {
fprintf(stderr, "Could not write data to FIFO\n");
return AVERROR_EXIT;
}
return 0;
}
/**
* Read one audio frame from the input file, decode, convert and store
* it in the FIFO buffer.
* @param fifo Buffer used for temporary storage
* @param input_format_context Format context of the input file
* @param input_codec_context Codec context of the input file
* @param output_codec_context Codec context of the output file
* @param 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)
*/
//这里做了什么. 分配临时存储转换数据的空间,然后几码输入数据.获取帧,在把帧进行转换,转换完写入fifo缓冲区
static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext *resampler_context,
int *finished) {
/* Temporary storage of the input samples of the frame read from the file. */
AVFrame *input_frame = NULL;
/* Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = NULL;
int data_present = 0;
int ret = AVERROR_EXIT;
/* Initialize temporary storage for one input frame. */
if (init_input_frame(&input_frame))//用来存放解码后的输入数据
goto cleanup;
/* Decode one frame worth of audio samples. */
//解码输入数据,获得输入帧.
if (decode_audio_frame(input_frame, input_format_context,
input_codec_context, &data_present, finished))
goto cleanup;
/* If we are at the end of the file and there are no more samples
* in the decoder which are delayed, we are actually finished.
* This must not be treated as an error. */
if (*finished) {
ret = 0;
goto cleanup;
}
/* If there is decoded data, convert and store it. */
if (data_present) {//这里表示已经有了解码后的帧数据,下边进行专门
/* Initialize the temporary storage for the converted input samples. */
//根据输出参数获取用于接收转码数据的数组 ,为数组分配空间
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. */
//转码音频数据inputFrame的数据.输出到 converted_input_samples中
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. */
//把转换后的数据写入到fifo 缓冲区中
if (add_samples_to_fifo(fifo, converted_input_samples,
input_frame->nb_samples))
goto cleanup;
ret = 0;
}
ret = 0;
cleanup:
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
free(converted_input_samples);
}
av_frame_free(&input_frame);
return ret;
}
/**
* 用输出文件的编解码器上下文初始化输出帧,分配内存
* Initialize one input frame for writing to the output file.
* The frame will be exactly frame_size samples large.
* @param[out] frame Frame to be initialized
* @param output_codec_context Codec context of the output file
* @param frame_size Size of the frame
* @return Error code (0 if successful)
*/
static int init_output_frame(AVFrame **frame,
AVCodecContext *output_codec_context,
int frame_size) {
int error;
/* Create a new frame to store the audio samples. */
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate output frame\n");
return AVERROR_EXIT;
}
/* Set the frame's parameters, especially its size and format.
* av_frame_get_buffer needs this to allocate memory for the
* audio samples of the frame.
* Default channel layouts based on the number of channels
* are assumed for simplicity. */
(*frame)->nb_samples = frame_size;
(*frame)->channel_layout = output_codec_context->channel_layout;
(*frame)->format = output_codec_context->sample_fmt;
(*frame)->sample_rate = output_codec_context->sample_rate;
/* Allocate the samples of the created frame. This call will make
* sure that the audio frame can hold as many samples as specified. */
if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
av_err2str(error));
av_frame_free(frame);
return error;
}
return 0;
}
/* Global timestamp for the audio frames. */
static int64_t pts = 0;
/**
* 编码音频真,使用输出编码器,编码完后的数据,写出到输出格式上下文中
* Encode one frame worth of audio to the output file.
* @param frame Samples to be encoded
* @param output_format_context Format context of the output file
* @param output_codec_context Codec context of the output file
* @param[out] data_present Indicates whether data has been
* encoded
* @return Error code (0 if successful)
*/
static int encode_audio_frame(AVFrame *frame,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context,
int *data_present) {
/* Packet used for temporary storage. */
AVPacket output_packet;
int error;
//初始packet
init_packet(&output_packet);
/* Set a timestamp based on the sample rate for the container. */
if (frame) {
frame->pts = pts;
pts += frame->nb_samples;
}
/* Send the audio frame stored in the temporary packet to the encoder.
* The output audio stream encoder is used to do this. */
//frame 送入编码器.
error = avcodec_send_frame(output_codec_context, frame);
/* The encoder signals that it has nothing more to encode. */
if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
av_err2str(error));
return error;
}
//取出编码后的数据
/* Receive one encoded frame from the encoder. */
error = avcodec_receive_packet(output_codec_context, &output_packet);
/* If the encoder asks for more data to be able to provide an
* encoded frame, return indicating that no data is present. */
if (error == AVERROR(EAGAIN)) {
error = 0;
goto cleanup;
/* If the last frame has been encoded, stop encoding. */
} else if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not encode frame (error '%s')\n",
av_err2str(error));
goto cleanup;
/* Default case: Return encoded data. */
} else {
*data_present = 1;
}
//把编码后的数据写出到输出文件中
/* Write one audio frame from the temporary packet to the output file. */
if (*data_present &&
(error = av_write_frame(output_format_context, &output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
av_err2str(error));
goto cleanup;
}
cleanup:
av_packet_unref(&output_packet);
return error;
}
/**
* Load one audio frame from the FIFO buffer, encode and write it to the
* output file.
* @param fifo Buffer used for temporary storage
* @param output_format_context Format context of the output file
* @param output_codec_context Codec context of the output file
* @return Error code (0 if successful)
*/
//编码数据然后写出到文件中
static int load_encode_and_write(AVAudioFifo *fifo,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context) {
/* Temporary storage of the output samples of the frame written to the file. */
AVFrame *output_frame;
/* Use the maximum number of possible samples per frame.
* If there is less than the maximum possible frame size in the FIFO
* buffer use this number. Otherwise, use the maximum possible frame size. */
const int frame_size = FFMIN(av_audio_fifo_size(fifo),
output_codec_context->frame_size);
int data_written;
//初始化临时存储视频帧的结构
/* 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. */
//把数据从fifo 中读入output_frame中
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. */
//数据编码器编码,然后把编码后的packet数据写出到文件中
if (encode_audio_frame(output_frame, output_format_context,
output_codec_context, &data_written)) {
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
av_frame_free(&output_frame);
return 0;
}
/**
* 写出尾部数据
* Write the trailer of the output file container.
* @param output_format_context Format context of the output file
* @return Error code (0 if successful)
*/
static int write_output_file_trailer(AVFormatContext *output_format_context) {
int error;
if ((error = av_write_trailer(output_format_context)) < 0) {
fprintf(stderr, "Could not write output file trailer (error '%s')\n",
av_err2str(error));
return error;
}
return 0;
}
/**
* 转换音频数据
* @param argc
* @param argv
* @return
*/
int transcode_aac_main(int argc, char **argv) {
AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
SwrContext *resample_context = NULL;
AVAudioFifo *fifo = NULL;
int ret = AVERROR_EXIT;
if (argc != 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(1);
}
/* Open the input file for reading. */
//打开输入文件格式 并初始化输入格式上下文
if (open_input_file(argv[1], &input_format_context,
&input_codec_context))
goto cleanup;
/* Open the output file for writing. */
//打开输出文件并初始输出编码器上下文和输出格式上下文
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. */
//初始化转换上下文
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
/* Initialize the FIFO buffer to store audio samples to be encoded. */
//创建输出buffer
if (init_fifo(&fifo, output_codec_context))
goto cleanup;
/* Write the header of the output file container. */
//写出头信息
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. */
//只要有输入的采样或者输出的采样.就循环,
while (1) {
/* Use the encoder's desired frame size for processing. */
const int output_frame_size = output_codec_context->frame_size;//每个通道的采样数
int finished = 0;
/* Make sure that there is one frame worth of samples in the FIFO
* buffer so that the encoder can do its work.
* Since the decoder's and the encoder's frame size may differ, we
* need to FIFO buffer to store as many frames worth of input samples
* that they make up at least one frame worth of output samples. */
//用输出的最大帧数做控制,把数据从输入中取出放入输出中
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. */
//解码输入数据.进行转换,然后放入输出数据中, 类似生产者消费者
if (read_decode_convert_and_store(fifo, input_format_context,
input_codec_context,
output_codec_context,
resample_context, &finished))
goto cleanup;
/* If we are at the end of the input file, we continue
* encoding the remaining audio samples to the output file. */
if (finished)
break;
}
/* If we have enough samples for the encoder, we encode them.
* At the end of the file, we pass the remaining samples to
* the encoder. */
//如果样本数已经足够了.就写到文件中
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. */
//编码数据.写出到文件中
if (load_encode_and_write(fifo, output_format_context,
output_codec_context))
goto cleanup;
/* If we are at the end of the input file and have encoded
* all remaining samples, we can exit this loop and finish. */
if (finished) {
int data_written;
/* Flush the encoder as it may have delayed frames. */
do {
data_written = 0;
//刷新编码器中的数据
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;
} while (data_written);
break;
}
}
//写出数据尾部
/* Write the trailer of the output file container. */
if (write_output_file_trailer(output_format_context))
goto cleanup;
ret = 0;
cleanup:
if (fifo)
av_audio_fifo_free(fifo);
swr_free(&resample_context);
if (output_codec_context)
avcodec_free_context(&output_codec_context);
if (output_format_context) {
avio_closep(&output_format_context->pb);
avformat_free_context(output_format_context);
}
if (input_codec_context)
avcodec_free_context(&input_codec_context);
if (input_format_context)
avformat_close_input(&input_format_context);
return ret;
}
