我想使用 ffmpeg 的 libav*/libswresample 转码并向下/重新采样音频以进行输出 - 我使用 ffmpeg 的 (4.x) transcode_aac.c 和 resample_audio.c 作为参考 - 但代码会生成音频显然不是 ffmpeg 本身会产生的故障（即 ffmpeg -i foo.wav -ar 22050 foo.m4a）
基于 ffmpeg 示例，要重新采样音频，我似乎需要将输出 AVAudioContext 和 SwrContext Sample_rate 设置为我想要的值，并确保为 swr_convert() 提供正确数量的 输出样本一旦我有解码的输入音频，基于 av_rescale_rnd( swr_delay(), ...) 。我已注意确保合并代码中考虑了输出样本的所有相关计算（如下）：
然而，生成的音频文件存在音频故障。社区是否知道有关如何进行转码和重新采样的任何参考资料或者此示例中缺少的内容？
/* compile and run:
gcc -I/usr/include/ffmpeg transcode-swr-aac.c -lavformat -lavutil -lavcodec -lswresample -lm
./a.out foo.wav foo.m4a
*/
/*
* Copyright (c) 2013-2018 Andreas Unterweger
*
* This file is part of FFmpeg.
... ...
*
* @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 ([email protected])
*/
#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/channel_layout.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
#include "libswresample/swresample.h"
#define OUTPUT_BIT_RATE 128000
#define OUTPUT_CHANNELS 2
static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
{
AVCodecContext *avctx;
const AVCodec *input_codec;
const AVStream *stream;
int error;
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;
}
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;
}
stream = (*input_format_context)->streams[0];
if (!(input_codec = avcodec_find_decoder(stream->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);
}
/* Initialize the stream parameters with demuxer information. */
error = avcodec_parameters_to_context(avctx, stream->codecpar);
if (error < 0) {
avformat_close_input(input_format_context);
avcodec_free_context(&avctx);
return error;
}
/* 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;
}
/* Set the packet timebase for the decoder. */
avctx->pkt_timebase = stream->time_base;
/* Save the decoder context for easier access later. */
*input_codec_context = avctx;
return 0;
}
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;
const AVCodec *output_codec = NULL;
int error;
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;
}
if (!(*output_format_context = avformat_alloc_context())) {
fprintf(stderr, "Could not allocate output format context\n");
return AVERROR(ENOMEM);
}
(*output_format_context)->pb = output_io_context;
if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
NULL))) {
fprintf(stderr, "Could not find output file format\n");
goto cleanup;
}
if (!((*output_format_context)->url = av_strdup(filename))) {
fprintf(stderr, "Could not allocate url.\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
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. */
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;
}
/* Set the basic encoder parameters.
* SET OUR DESIRED output sample_rate here
*/
avctx->channels = OUTPUT_CHANNELS;
avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
// avctx->sample_rate = input_codec_context->sample_rate;
avctx->sample_rate = 22050;
avctx->sample_fmt = output_codec->sample_fmts[0];
avctx->bit_rate = OUTPUT_BIT_RATE;
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/* Set the sample rate for the container. */
stream->time_base.den = avctx->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;
}
/* 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.
*/
static int init_packet(AVPacket **packet)
{
if (!(*packet = av_packet_alloc())) {
fprintf(stderr, "Could not allocate packet\n");
return AVERROR(ENOMEM);
}
return 0;
}
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;
}
static int init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext **resample_context)
{
int error;
/**
* create the resample, including ref to the desired output sample rate
*/
*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 < 0) {
fprintf(stderr, "Could not allocate resample context\n");
return AVERROR(ENOMEM);
}
if ((error = swr_init(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
swr_free(resample_context);
return error;
}
return 0;
}
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;
}
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;
}
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;
*data_present = 0;
*finished = 0;
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;
}
static int init_converted_samples(uint8_t ***converted_input_samples,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
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);
}
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;
}
static int convert_samples(const uint8_t **input_data, const int input_nb_samples,
uint8_t **converted_data, const int output_nb_samples,
SwrContext *resample_context)
{
int error;
if ((error = swr_convert(resample_context,
converted_data, output_nb_samples,
input_data , input_nb_samples)) < 0) {
fprintf(stderr, "Could not convert input samples (error '%s')\n",
av_err2str(error));
return error;
}
return 0;
}
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;
}
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;
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) {
/* 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;
/* figure out how many samples are required for target sample_rate incl
* any items left in the swr buffer
*/
int output_nb_samples = av_rescale_rnd(
swr_get_delay(resampler_context, input_codec_context->sample_rate) + input_frame->nb_samples,
output_codec_context->sample_rate,
input_codec_context->sample_rate,
AV_ROUND_UP);
/* ignore, just to ensure we've got enough buffer alloc'd for conversion buffer */
av_assert1(input_frame->nb_samples > output_nb_samples);
/* Convert the input samples to the desired output sample format, via swr_convert().
*/
if (convert_samples((const uint8_t**)input_frame->extended_data, input_frame->nb_samples,
converted_input_samples, output_nb_samples,
resampler_context))
goto cleanup;
/* Add the converted input samples to the FIFO buffer for later processing. */
if (add_samples_to_fifo(fifo, converted_input_samples,
output_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;
}
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;
}
/* 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;
/**
* Encode one frame worth of audio to the output file.
*/
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;
/* Set a timestamp based on the sample rate for the container. */
if (frame) {
frame->pts = pts;
pts += frame->nb_samples;
}
*data_present = 0;
error = avcodec_send_frame(output_codec_context, frame);
if (error < 0 && error != AVERROR_EOF) {
fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
av_err2str(error));
goto cleanup;
}
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;
}
/* 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_free(&output_packet);
return error;
}
/**
* Load one audio frame from the FIFO buffer, encode and write it to the
* output file.
*/
static int load_encode_and_write(AVAudioFifo *fifo,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context)
{
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;
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. */
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. */
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;
}
/**
* Write the trailer of the output file container.
*/
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;
}
int 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);
}
if (open_input_file(argv[1], &input_format_context,
&input_codec_context))
goto cleanup;
if (open_output_file(argv[2], input_codec_context,
&output_format_context, &output_codec_context))
goto cleanup;
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
if (init_fifo(&fifo, output_codec_context))
goto cleanup;
if (write_output_file_header(output_format_context))
goto cleanup;
while (1) {
/* Use the encoder's desired frame size for processing. */
const int output_frame_size = output_codec_context->frame_size;
int finished = 0;
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. */
if (read_decode_convert_and_store(fifo, input_format_context,
input_codec_context,
output_codec_context,
resample_context, &finished))
goto cleanup;
if (finished)
break;
}
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 {
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;
} while (data_written);
break;
}
}
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;
}
浏览 ffmpeg/libav 邮件列表后，特别是https://ffmpeg.org/pipermail/libav-user/2017-July/010496.html，我能够修改 ffmpeg transcode_aac.c 示例来执行采样率转换。
在原始代码中，主循环在一个函数中读取/解码/隐藏/存储，然后将样本传递给编码器使用的
AVAudioFifo某些编码器需要特定数量的样本 - 如果您提供的样本数量较少，则编码器垫似乎会达到预期，这会导致我第一次尝试中提到的故障。
根据 ffmpeg 邮件列表，关键是缓冲/连接解码的输入样本，直到我们有足够的样本用于编码器的至少一帧。为此，我们将读取/解码与转换/存储分开，并将读取/解码数据存储在新的中介
AVAudioFifofifostatic int read_decode_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
const int audio_stream_idx,
int *finished)
{
AVFrame *input_frame = 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, audio_stream_idx, &data_present, finished))
goto cleanup;
if (*finished) {
ret = 0;
goto cleanup;
}
if (data_present) {
/* Add the converted input samples to the FIFO buffer for later processing. */
if (add_samples_to_fifo(fifo, (uint8_t**)input_frame->extended_data, input_frame->nb_samples))
goto cleanup;
}
ret = 0;
cleanup:
av_frame_free(&input_frame);
return ret;
}
static int load_convert_and_store(AVAudioFifo* output_samples_fifo, const AVFormatContext* output_context, AVCodecContext* output_codec_context, int output_frame_size,
AVAudioFifo* input_samples_fifo, const AVFormatContext* input_context, AVCodecContext* input_codec_context,
SwrContext* resample_context)
{
uint8_t **converted_input_samples = NULL;
int ret = AVERROR_EXIT;
AVFrame *input_frame;
const int frame_size = FFMIN(av_audio_fifo_size(input_samples_fifo),
output_frame_size);
// yes this is init_output_frame
if (init_output_frame(&input_frame, input_codec_context, frame_size))
return AVERROR_EXIT;
if (av_audio_fifo_read(input_samples_fifo, (void **)input_frame->data, frame_size) < frame_size) {
fprintf(stderr, "Could not read data from input samples FIFO");
av_frame_free(&input_frame);
return AVERROR_EXIT;
}
int nb_samples = (output_codec_context->sample_rate == input_codec_context->sample_rate) ?
input_frame->nb_samples :
av_rescale_rnd(swr_get_delay(resample_context, input_codec_context->sample_rate) + input_frame->nb_samples, output_codec_context->sample_rate, input_codec_context->sample_rate, AV_ROUND_UP);
if (init_converted_samples(&converted_input_samples, output_codec_context,
nb_samples))
goto cleanup;
/* **** Modify convert_samples() to return the value from swr_convert() **** */
if ( (nb_samples = convert_samples((const uint8_t**)input_frame->extended_data, input_frame->nb_samples,
converted_input_samples, output_codec_context->frame_size,
resample_context)) < 0)
goto cleanup;
if (add_samples_to_fifo(output_samples_fifo, converted_input_samples, nb_samples))
goto cleanup;
ret = 0;
cleanup:
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
free(converted_input_samples);
}
av_frame_free(&input_frame);
return ret;
}
int main()
{
...
while (1)
{
const int output_frame_size = output_codec_context->frame_size;
int finished = 0;
/* Re: Resample frame to specified number of samples
* https://ffmpeg.org/pipermail/libav-user/2017-July/010496.html
* Yes, you need to buffer sufficient audio frames to feed to the encoder.
*
* Calculate the number of in samples:
in_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) +
out_nb_samples,
in_sample_rate, c->sample_rate, AV_ROUND_DOWN);
then allocate buffers to concatenate the in samples until you have enough
to pass to swr_ctx.
*/
while (av_audio_fifo_size(input_samples_fifo) < output_frame_size) {
if (read_decode_and_store(input_samples_fifo,
input_format_context, input_codec_context,
audio_stream_idx,
&finished))
goto cleanup;
if (finished)
break;
}
while (av_audio_fifo_size(input_samples_fifo) >= output_frame_size ||
(finished && av_audio_fifo_size(input_samples_fifo) > 0)) {
/* take all input samples and convert them before handing off to encoder
*/
if (load_convert_and_store(fifo,
output_format_context, output_codec_context, output_frame_size,
input_samples_fifo, input_format_context, input_codec_context,
resample_context))
goto cleanup;
}
}
/* 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. */
.... // existing code
}
我的经验要少得多，所以非常感谢您的耐心。我已将您共享的第二个代码与原始代码结合起来，但仍然遇到错误（特别是：无法重新分配 FIFO）。我想知道你是否也遇到过这种情况？将读取/解码/转换/存储分为两个单独的功能的概念非常棒！按照建议，我分配了第二个 FIFO 来初步存储样本，以最好地满足缓冲区/连接逻辑。然而，我对你的decode_audio_frame()函数有点迷失。看来您添加了一个名为“audio_stream_idx”的新变量。你能提供完整的代码或新的decode()吗？看来将解码数据存储到“input_samples_fifo”中是问题所在。我们不需要先转换它吗？我假设你的新decode()以某种方式处理了这个问题。也尝试解决我的 AAC 重新采样问题。请求帮助！