启发:https://zhuanlan.zhihu.com/p/439000616
前言:在后台开发过程中,需求是转码前后的文件,不保存,直接写入到缓存中,可以直接、方便的插入到后台开发过程,接受上个模块的输入,以及提供给下一个模块使用。
名词解释
- AVFormatContext: 存储音频数据的容器类;
- AVCodecContext: 存储编码使用参数,列举重要参数:codec_type, bit_rate, sample_rate, channels, 等等;
- AVCodec: 存储编解码器信息的结构体,列举重要参数:name, long_name, id(codecid),channel_layouts等等;
- AVIOContext: 管理输入输出数据的结构体,列举重要参数:buffer, buffer_size, buf_ptr, buf_end, opaque;
- AVPacket: 存储压缩编解码数据相关信息的结构体,列举重要参数:data, size, pts, dts, stream_index等等;
- AVStream: 存储每个音频/视频信息的结构体,列举重要参数: index, codec, duration, time_base等等;
- AVFrame: 码流参数较多的结构体,一般存储原始数据(非压缩数据)
- AVAudioFifo: 缓冲区,以音频采样为基本单位的先进先出队列
- codec: 编解码算法,主要分为有损和无损
- container: 容器格式,即音频流,特定格式的文件中的协议
- transcoding: 转码功能,将一种音频的编码方式转为另外的一种编码方式
流程如下:
输入的是音频流,例如wav、MP3等等,输出为指定格式AAC编码的音频流,例如m4a、MP4等,流式输入获取编码器 、上下文信息等,输出流指定封装格式,先将输入的音频重采样到指定AAC编码格式,接着使用FIFO形式处理解码数据,解码器根据packet根据指定的问codec_id解压成对应的帧,再根据指定的鹅codec_id将frame编码成packets,最后做封装打包为音频流。transcoding_aac.c源代码:
#include <stdio.h>
#include <iostream>
#include <mutex>
#include <sys/stat.h>
extern "C" {
#include "libavformat/avformat.h"
#include "libavcodec/avcodec.h"
#include "libswresample/swresample.h"
#include "libavdevice/avdevice.h"
#include <libavutil/file.h>
#include "libavformat/avio.h"
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/frame.h>
#include <libavutil/samplefmt.h>
#include <libavutil/audio_fifo.h>
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
}
using namespace std;
#ifdef av_err2str
#undef av_err2str
#include <string>
av_always_inline std::string av_err2string(int errnum) {
char str[AV_ERROR_MAX_STRING_SIZE];
return av_make_error_string(str, AV_ERROR_MAX_STRING_SIZE, errnum);
}
#define av_err2str(err) av_err2string(err).c_str()
#endif // av_err2str
#define OUTPUT_BIT_RATE 96000
struct buffer_data_output{
uint8_t *buf;
size_t size;
uint8_t *ptr;
size_t room;
};
struct buffer_data_input {
uint8_t *ptr;
uint8_t *buf;
std::size_t size;
std::size_t file_size;
};
static int iowrite_to_buffer(void *opaque, uint8_t *buf, int buf_size)
{
struct buffer_data_output *bd = (struct buffer_data_output *)opaque;
while (buf_size > bd->room)
{
int64_t offset = bd->ptr - bd->buf;
bd->buf = (uint8_t*)av_realloc_f(bd->buf, 2, bd->size);
if (!bd->buf)
return AVERROR(ENOMEM);
bd->size = bd->size*2;
bd->ptr = bd->buf + offset;
bd->room = bd->size - offset;
}
memcpy(bd->ptr, buf, buf_size);
bd->ptr += buf_size;
bd->room -= buf_size;
// printf("write packet pkt_size:%d used_buf_size:%zu buf_size:%zu buf_room:%zu\n", buf_size, bd->ptr - bd->buf, bd->size, bd->room);
return buf_size;
}
int64_t seek_buffer(void *ptr, int64_t pos, int whence)
{
struct buffer_data_output *bd = (struct buffer_data_output *)ptr;
int64_t ret = -1;
switch (whence)
{
case AVSEEK_SIZE:
return bd->size;
break;
case SEEK_SET:
bd->ptr = bd->buf + pos;
// bd->room = bd->size - pos;
break;
case SEEK_CUR:
if (bd->room > pos) {
bd->ptr += pos;
bd->room -= pos;
} else return EOF;
break;
case SEEK_END:
if (bd->size > pos) {
bd->ptr = (bd->buf + bd->size) - pos;
int curPos = bd->ptr - bd->buf;
bd->room = bd->size - curPos;
} else return EOF;
break;
default:
break;
}
ret = bd->ptr - bd->buf;
// printf("whence=%d , offset=%ld , buffer_size=%ld, buffer_room=%ld\n", whence, pos, bd->size, bd->room);
return ret;
}
static int read_packet_input(void *opaque, uint8_t *buf, int buf_size){
struct buffer_data_input *bd = (struct buffer_data_input *)opaque;
buf_size = FFMIN(buf_size, bd->size);
if (!buf_size)
return AVERROR_EOF;
memcpy(buf, bd->ptr, buf_size);
bd->ptr += buf_size;
bd->size -= buf_size;
return buf_size;
}
int64_t seek_buffer_input(void *ptr, int64_t pos, int whence){
struct buffer_data_input *bd = (struct buffer_data_input *)ptr;
int64_t ret = -1;
switch (whence)
{
case AVSEEK_SIZE:
return bd->file_size;
break;
case SEEK_SET:
if (bd->file_size > pos) {
bd->ptr = bd->buf + pos;
bd->size = bd->file_size - pos;
} else return EOF;
break;
case SEEK_CUR:
if (bd->size > pos) {
bd->ptr += pos;
bd->size -= pos;
} else return EOF;
break;
case SEEK_END:
if (bd->file_size > pos) {
bd->ptr = (bd->buf + bd->file_size) - pos;
int curPos = bd->ptr - bd->buf;
bd->size = bd->file_size - curPos;
} else return EOF;
break;
default:
break;
}
return bd->ptr - bd->buf;
}
struct buffer_data_input bd = {0};
struct buffer_data_output bd_zm = {0};
/**
* Open an input file and the required decoder.
* @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(uint8_t* buffer, size_t buffer_size, AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
{
AVCodecContext *avctx;
const AVCodec *input_codec;
AVIOContext *avio_ctx;
uint8_t *in_buffer = NULL;
size_t avio_ctx_buffer_size = 4096;
int error;
in_buffer = (uint8_t*)av_malloc(avio_ctx_buffer_size);
if (!in_buffer) {
std::cout << "av malloc error!" << std::endl;
return AVERROR_EXIT;
}
bd.buf = bd.ptr = buffer;
bd.size = bd.file_size = buffer_size;
avio_ctx = avio_alloc_context(in_buffer, avio_ctx_buffer_size, 0, &bd, &read_packet_input, NULL, &seek_buffer_input);
*input_format_context = avformat_alloc_context();
if(!(*input_format_context)) {
std::cout << "avformat_alloc_context error!" << std::endl;
return -1;
}
(*input_format_context)->pb = avio_ctx;
(*input_format_context)->flags |= AVFMT_FLAG_CUSTOM_IO;
if ((error = avformat_open_input(input_format_context, NULL, NULL,
NULL)) < 0) {
fprintf(stderr, "Could not open input file (error '%s')\n", av_err2str(error));
*input_format_context = NULL;
return error;
}
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;
}
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;
}
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;
}
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);
}
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;
}
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;
}
*input_codec_context = avctx;
return 0;
}
/**
* 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 channels Output file channels
* @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(int channels, AVCodecContext *input_codec_context,
AVFormatContext **output_format_context,
AVCodecContext **output_codec_context)
{
AVCodecContext *avctx = NULL;
AVStream *stream = NULL;
const AVCodec *output_codec = NULL;
int error;
AVIOContext *avio_ctx_zm = NULL;
uint8_t *avio_ctx_buffer_zm = NULL;
size_t avio_ctx_buffer_size_zm = 4096;
const size_t bd_out_buf_size = 1024;
const AVOutputFormat *output_format = av_guess_format("mp4", NULL , NULL);
avformat_alloc_output_context2(output_format_context, output_format, NULL, NULL);
if (!(*output_format_context)) {
av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
return AVERROR_UNKNOWN;
}
bd_zm.ptr = bd_zm.buf = (uint8_t*)av_malloc(bd_out_buf_size);
if (!bd_zm.buf) {
return AVERROR(ENOMEM);
}
bd_zm.size = bd_zm.room = bd_out_buf_size;
avio_ctx_buffer_zm = (uint8_t*)av_malloc(avio_ctx_buffer_size_zm);
if (!avio_ctx_buffer_zm) {
av_log(NULL, AV_LOG_ERROR, "allocate buffer error\n");
return AVERROR(ENOMEM);
}
avio_ctx_zm = avio_alloc_context(avio_ctx_buffer_zm, avio_ctx_buffer_size_zm,
1, &bd_zm, NULL, iowrite_to_buffer, seek_buffer);
if (!avio_ctx_zm) {
av_log(NULL, AV_LOG_ERROR, "allocate avio context error\n");
return AVERROR(ENOMEM);
}
(*output_format_context)->pb = avio_ctx_zm;
if (!(*output_format_context)->pb) {
av_log(NULL, AV_LOG_ERROR, "output format context pb is empty\n");
return AVERROR(ENOMEM);
}
// (*output_format_context)->flags |= AVFMT_FLAG_CUSTOM_IO;
(*output_format_context)->oformat = output_format;
if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
fprintf(stderr, "Could not find an AAC encoder.\n");
goto cleanup;
}
if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
avctx = avcodec_alloc_context3(output_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate an encoding context\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
avctx->channels = channels;
avctx->channel_layout = av_get_default_channel_layout(avctx->channels);
avctx->sample_rate = input_codec_context->sample_rate;
avctx->sample_fmt = output_codec->sample_fmts[0];
avctx->bit_rate = OUTPUT_BIT_RATE;
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
stream->time_base.den = input_codec_context->sample_rate;
stream->time_base.num = 1;
if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
av_err2str(error));
goto cleanup;
}
error = avcodec_parameters_from_context(stream->codecpar, avctx);
if (error < 0) {
fprintf(stderr, "Could not initialize stream parameters\n");
goto cleanup;
}
*output_codec_context = avctx;
if ((avformat_write_header(*output_format_context, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
return -1;
}
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[out] packet Packet to be initialized
* @return Error code (0 if successful)
*/
static int init_packet(AVPacket **packet)
{
if (!(*packet = av_packet_alloc())) {
fprintf(stderr, "Could not allocate packet\n");
return AVERROR(ENOMEM);
}
return 0;
}
/**
* 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;
}
/**
* 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;
*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);
}
av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
if ((error = swr_init(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
swr_free(resample_context);
return error;
}
return 0;
}
/**
* 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)
{
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;
}
/**
* 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)
{
AVPacket *input_packet;
int error;
error = init_packet(&input_packet);
if (error < 0)
return error;
if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
if (error == AVERROR_EOF)
*finished = 1;
else {
fprintf(stderr, "Could not read frame (error '%s')\n",
av_err2str(error));
goto cleanup;
}
}
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));
goto cleanup;
}
error = avcodec_receive_frame(input_codec_context, frame);
if (error == AVERROR(EAGAIN)) {
error = 0;
goto cleanup;
} 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;
} else {
*data_present = 1;
goto cleanup;
}
cleanup:
av_packet_free(&input_packet);
return error;
}
/**
* 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;
if (!(*converted_input_samples = static_cast<uint8_t **>(calloc(output_codec_context->channels,
sizeof(**converted_input_samples))))) {
fprintf(stderr, "Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
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;
}
/**
* 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;
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;
}
/**
* 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;
if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
fprintf(stderr, "Could not reallocate FIFO\n");
return error;
}
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)
*/
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)
{
AVFrame *input_frame = NULL;
uint8_t **converted_input_samples = NULL;
int data_present = 0;
int ret = AVERROR_EXIT;
if (init_input_frame(&input_frame))
goto cleanup;
if (decode_audio_frame(input_frame, input_format_context,
input_codec_context, &data_present, finished))
goto cleanup;
if (*finished) {
ret = 0;
goto cleanup;
}
if (data_present) {
if (init_converted_samples(&converted_input_samples, output_codec_context,
input_frame->nb_samples))
goto cleanup;
if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
input_frame->nb_samples, resampler_context))
goto cleanup;
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;
}
/**
* 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;
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate output frame\n");
return AVERROR_EXIT;
}
(*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;
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;
/**
* 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)
{
AVPacket *output_packet;
int error;
error = init_packet(&output_packet);
if (error < 0)
return error;
if (frame) {
frame->pts = pts;
pts += frame->nb_samples;
}
error = avcodec_send_frame(output_codec_context, frame);
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));
goto cleanup;
}
/* Receive one encoded frame from the encoder. */
error = avcodec_receive_packet(output_codec_context, output_packet);
if (error == AVERROR(EAGAIN)) {
error = 0;
goto cleanup;
} 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;
} else {
*data_present = 1;
}
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_free(&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)
*/
static int load_encode_and_write(AVAudioFifo *fifo,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context)
{
AVFrame *output_frame;
const int frame_size = FFMIN(av_audio_fifo_size(fifo),
output_codec_context->frame_size);
int data_written;
if (init_output_frame(&output_frame, output_codec_context, frame_size))
return AVERROR_EXIT;
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;
}
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;
}
int tom4atype(uint8_t* buffer, size_t buffer_size, int channels, std::string &output)
{
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;
//打开输入文件buffer去读
if (open_input_file(buffer, buffer_size, &input_format_context,
&input_codec_context))
goto cleanup;
//打开输出文件buffer去写
if (open_output_file(channels, input_codec_context, &output_format_context,
&output_codec_context))
goto cleanup;
//初始化转换上下文
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
if (init_fifo(&fifo, output_codec_context))
goto cleanup;
while (1) {
//输出时每个通道的采样数
const int output_frame_size = output_codec_context->frame_size;
int finished = 0;
while (av_audio_fifo_size(fifo) < output_frame_size) {
//解码输入数据,进行转换,然后放到输出数据中,类似于生产者-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 (finished)
break;
}
//样本数足够,写buffer
while (av_audio_fifo_size(fifo) >= output_frame_size ||
(finished && av_audio_fifo_size(fifo) > 0))
if (load_encode_and_write(fifo, output_format_context,
output_codec_context))
goto cleanup;
if (finished) {
int data_written;
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;
}
}
//写文件尾到buffer
av_write_trailer(output_format_context);
//写结果
output.assign((char*)bd_zm.buf, bd_zm.size - bd_zm.room);
std::cout << "output size: " << output.size() << std::endl;
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) {
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;
}
int main(int argc, char** argv) {
if (argc != 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(1);
}
uint8_t* buffer;
size_t buffer_size;
int ret = av_file_map(argv[1], &buffer, &buffer_size, 0, NULL);
if (ret < 0) {
std::cout << "av_file_map error" << std::endl;
return -1;
}
std::string output;
tom4atype(buffer, buffer_size, 2, output);
FILE *pFile;
pFile = fopen(argv[2], "wb");
if (pFile) {
// fwrite(bd_zm.buf, bd_zm.size - bd_zm.room, 1, pFile); //跟ffmpeg命令行比, 最后多个8bit的空值
fwrite(output.data(), bd_zm.size - bd_zm.room, 1, pFile);
puts("Wrote to file!");
}
else {
puts("Something wrong writing to File.");
}
fclose(pFile);
}