[英]How can I Convert raw file to an audio file (.wav) by FFmpeg in c++
I want to covert mp4 format to wav format with different sample rate in my c++ application.我想在我的 c++ 应用程序中将 mp4 格式转换为具有不同采样率的 wav 格式。 First of all I have extracted audio from mp4 file by ffmpeg in c++, then i have converted that to a raw file, but I down not know how can I convert raw file to a wav file with different sample rate.
首先,我通过 c++ 中的 ffmpeg 从 mp4 文件中提取了音频,然后我将其转换为原始文件,但我不知道如何将原始文件转换为具有不同采样率的 wav 文件。 How can I solve this?
我该如何解决这个问题?
#include "ffmpeg.h"
int decode_packet(int *got_frame, int cached)
{
int ret = 0;
int decoded = pkt.size;
*got_frame = 0;
if (pkt.stream_index == video_stream_idx) {
/* decode video frame */
ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
// fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
if (*got_frame) {
if (frame->width != width || frame->height != height ||
frame->format != pix_fmt) {
/* To handle this change, one could call av_image_alloc again and
* decode the following frames into another rawvideo file. */
// fprintf(stderr, "Error: Width, height and pixel format have to be "
// "constant in a rawvideo file, but the width, height or "
// "pixel format of the input video changed:\n"
// "old: width = %d, height = %d, format = %s\n"
// "new: width = %d, height = %d, format = %s\n",
// width, height, av_get_pix_fmt_name(pix_fmt),
// frame->width, frame->height,
// av_get_pix_fmt_name(frame->format));
return -1;
}
printf("video_frame%s n:%d coded_n:%d\n",
cached ? "(cached)" : "",
video_frame_count++, frame->coded_picture_number);
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy(video_dst_data, video_dst_linesize,
(const uint8_t **)(frame->data), frame->linesize,
pix_fmt, width, height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
}
} else if (pkt.stream_index == audio_stream_idx) {
/* decode audio frame */
ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
// fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
return ret;
}
/* Some audio decoders decode only part of the packet, and have to be
* called again with the remainder of the packet data.
* Sample: fate-suite/lossless-audio/luckynight-partial.shn
* Also, some decoders might over-read the packet. */
decoded = FFMIN(ret, pkt.size);
if (*got_frame) {
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample((AVSampleFormat)frame->format);
// printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
// cached ? "(cached)" : "",
// audio_frame_count++, frame->nb_samples,
// av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
// fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
//encode function
encode(cOut, frame, &pktout, audio_dst_file);
// av_init_packet(&pktout);
// pktout.data = NULL; // packet data will be allocated by the encoder
// pktout.size = 0;
// /* encode the samples */
// ret = avcodec_encode_audio2(cOut, &pktout, frame, &got_outputOut);
// if (ret < 0) {
// fprintf(stderr, "Error encoding audio frame\n");
// exit(1);
// }
// if (got_outputOut) {
// fwrite(pktout.data, 1, pktout.size, audio_dst_file);
// av_free_packet(&pktout);
// }
}
}
/* If we use frame reference counting, we own the data and need
* to de-reference it when we don't use it anymore */
if (*got_frame && refcount)
av_frame_unref(frame);
return decoded;
}
First you should use Libswresample to resample audio data.首先,您应该使用Libswresample重新采样音频数据。
Then you can save audio raw data with wav format.然后你可以用 wav 格式保存音频原始数据。
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