/**
- @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;
//初始化輸入格式上下文
/ 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;
}
//找到輸入流的解碼器,根據解碼參數
/ 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;
}
//分配解碼器上下文
/* 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);
}
//拷貝輸入格式中的參數到解碼器上下文中
/ 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;
}
//打開解碼器上下文
/* 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;
}
/**
打開輸出文件,初始化編碼器上下文
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;
//打開輸出文件.創(chuàng)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;
}
//輸出格式上下文,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);
}
//保存輸出io上下文
/ Associate the output file (pointer) with the container format context. /
(output_format_context)->pb = output_io_context;
//根據文件名猜猜輸出格式
/ 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;
}
//找到指定編碼器
/ 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;
}
//從輸出格式中創(chuàng)建 輸出流
/ 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;
}
//用編碼器初始化輸出編碼器上下文
avctx = avcodec_alloc_context3(output_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate an encoding context\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
//填充輸出上下文的參數,這里看到,輸出的編碼器上下文參數是自己指定的,而輸入的解碼器的參數則來自文件的流中的參數
/ 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. */
//在用輸出的編碼器打開輸出編碼器上下文的參數
if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
av_err2str(error));
goto cleanup;
}
// 用編碼器上下文的參數填充輸出流參數
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;
} - The input file's sample rate is used to avoid a sample rate conversion. */
/**
- 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;
}
/**
- 初始化一個幀
- 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;
}
/**
用輸入輸出的編解碼上下文來生產轉換上下文
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).
/
//根據輸出輸入的編解碼器來分配 轉換上下文的參數
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. /
//參數設置完成了. 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;
}
/**
- 初始化一個先進先出的緩沖區(qū),針對輸出格式的大小
- 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. /
//根據輸出參數返回創(chuàng)建一個先進先出的緩存
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;
}
/**
- 寫出輸出文件的頭信息
- 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 beendecoded. If this flag is false, thereis more data to be decoded, i.e., thisfunction has to be called again.@return Error code (0 if successful)
-
從輸入文件中解碼音頻真
*/
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);
//讀取數據到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數據送入解碼器
/ 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;
}
//獲取解碼后的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;
} - The input audio stream decoder is used to do this. /
/**
分配個臨時用的存儲區(qū)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. /
//給數組分配空間緩沖
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;
}
/**
轉換輸入數據到輸出數據,根據frame_size轉換
數據從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;
}
/**
把轉換后的數據寫入到fifo 的緩沖區(qū)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. */
//擴充fifo.保證數據能放下.
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. */
//把輸入存入 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;
} - the old and the new samples. */
/**
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 beendecoded. If this flag is false,there is more data to be decoded,i.e., this function has to be calledagain.-
@return Error code (0 if successful)
*/
//這里做了什么. 分配臨時存儲轉換數據的空間,然后幾碼輸入數據.獲取幀,在把幀進行轉換,轉換完寫入fifo緩沖區(qū)
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))//用來存放解碼后的輸入數據
goto cleanup;
/ Decode one frame worth of audio samples. /
//解碼輸入數據,獲得輸入幀.
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 samplesin 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) {//這里表示已經有了解碼后的幀數據,下邊進行專門
/ Initialize the temporary storage for the converted input samples. */
//根據輸出參數獲取用于接收轉碼數據的數組 ,為數組分配空間
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. */
//轉碼音頻數據inputFrame的數據.輸出到 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. */
//把轉換后的數據寫入到fifo 緩沖區(qū)中
if (add_samples_to_fifo(fifo, converted_input_samples,
input_frame->nb_samples))
goto cleanup;
ret = 0;
}
ret = 0; - This requires a temporary storage provided by converted_input_samples. */
cleanup:
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
free(converted_input_samples);
}
av_frame_free(&input_frame);return ret;
}
/**
用輸出文件的編解碼器上下文初始化輸出幀,分配內存
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;
/**
編碼音頻真,使用輸出編碼器,編碼完后的數據,寫出到輸出格式上下文中
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 送入編碼器.
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;
}
//取出編碼后的數據
/ 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;
}
//把編碼后的數據寫出到輸出文件中
/ 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;
} - The output audio stream encoder is used to do this. /
/**
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)
*/
//編碼數據然后寫出到文件中
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;
//初始化臨時存儲視頻幀的結構
/* 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. */
//把數據從fifo 中讀入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. */
//數據編碼器編碼,然后把編碼后的packet數據寫出到文件中
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;
}
/**
- 寫出尾部數據
- 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;
}
/**
轉換音頻數據
@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. /
//打開輸入文件格式 并初始化輸入格式上下文
if (open_input_file(argv[1], &input_format_context,
&input_codec_context))
goto cleanup;
/ Open the output file for writing. /
//打開輸出文件并初始輸出編碼器上下文和輸出格式上下文
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. /
//初始化轉換上下文
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
/ Initialize the FIFO buffer to store audio samples to be encoded. /
//創(chuàng)建輸出buffer
if (init_fifo(&fifo, output_codec_context))
goto cleanup;
/ Write the header of the output file container. */
//寫出頭信息
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. /
//只要有輸入的采樣或者輸出的采樣.就循環(huán),
while (1) {
/ Use the encoder's desired frame size for processing. */
const int output_frame_size = output_codec_context->frame_size;//每個通道的采樣數
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. /
//用輸出的最大幀數做控制,把數據從輸入中取出放入輸出中
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. */
//解碼輸入數據.進行轉換,然后放入輸出數據中, 類似生產者消費者
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;
}
- output sample format and put it into the FIFO buffer. */
/* 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. /
//如果樣本數已經足夠了.就寫到文件中
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. */
//編碼數據.寫出到文件中
if (load_encode_and_write(fifo, output_format_context,
output_codec_context))
goto cleanup;
- encode it and write it to the output file. */
/* 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;
//刷新編碼器中的數據
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;
} while (data_written);
break;
}
}
//寫出數據尾部
/* 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;
} -
