使用 libavcodec 将音频编码为 aac [英] Encode audio to aac with libavcodec
问题描述
我正在使用 libavcodec(截至 10 年 3 月 3 日的最新 git)将原始 pcm 编码为 aac(已启用 libfaac 支持).我通过调用 avcodec_encode_audio 来做到这一点每次重复使用 codec_context->frame_size 样本.前四个调用成功返回，但第五个调用永远不会返回.当我使用 gdb中断，堆栈已损坏.
I'm using libavcodec (latest git as of 3/3/10) to encode raw pcm to aac (libfaac support enabled). I do this by calling avcodec_encode_audio repeatedly with codec_context->frame_size samples each time. The first four calls return successfully, but the fifth call never returns. When I use gdb to break, the stack is corrupt.
如果我使用 audacity 将 pcm 数据导出到 .wav 文件，那么我可以使用命令行 ffmpeg 转换为 aac 没有任何问题，所以我确定它是我做错了什么.
If I use audacity to export the pcm data to a .wav file, then I can use command-line ffmpeg to convert to aac without any issues, so I'm sure it's something I'm doing wrong.
我编写了一个小型测试程序来重现我的问题.它读取来自文件的测试数据，可在此处获得:http://birdie.protoven.com/audio.pcm(约 2 秒的签名 16 位LE pcm)
I've written a small test program that duplicates my problem. It reads the test data from a file, which is available here: http://birdie.protoven.com/audio.pcm (~2 seconds of signed 16 bit LE pcm)
如果我直接使用 FAAC，我可以让它一切正常，但是如果我可以只使用 libavcodec，代码会更简洁一些，因为我也在编码视频，并将两者都写入 mp4.
I can make it all work if I use FAAC directly, but the code would be a little cleaner if I could just use libavcodec, as I'm also encoding video, and writing both to an mp4.
ffmpeg 版本信息:
ffmpeg version info:
FFmpeg version git-c280040, Copyright (c) 2000-2010 the FFmpeg developers
built on Mar 3 2010 15:40:46 with gcc 4.4.1
configuration: --enable-libfaac --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-pthreads --enable-debug=3 --enable-shared
libavutil 50.10. 0 / 50.10. 0
libavcodec 52.55. 0 / 52.55. 0
libavformat 52.54. 0 / 52.54. 0
libavdevice 52. 2. 0 / 52. 2. 0
libswscale 0.10. 0 / 0.10. 0
libpostproc 51. 2. 0 / 51. 2. 0
有什么我没有设置，或者在我的编解码器中设置不正确上下文，也许?非常感谢任何帮助！
Is there something I'm not setting, or setting incorrectly in my codec context, maybe? Any help is greatly appreciated!
这是我的测试代码:
#include <stdio.h>
#include <libavcodec/avcodec.h>
void EncodeTest(int sampleRate, int channels, int audioBitrate,
uint8_t *audioData, size_t audioSize)
{
AVCodecContext *audioCodec;
AVCodec *codec;
uint8_t *buf;
int bufSize, frameBytes;
avcodec_register_all();
//Set up audio encoder
codec = avcodec_find_encoder(CODEC_ID_AAC);
if (codec == NULL) return;
audioCodec = avcodec_alloc_context();
audioCodec->bit_rate = audioBitrate;
audioCodec->sample_fmt = SAMPLE_FMT_S16;
audioCodec->sample_rate = sampleRate;
audioCodec->channels = channels;
audioCodec->profile = FF_PROFILE_AAC_MAIN;
audioCodec->time_base = (AVRational){1, sampleRate};
audioCodec->codec_type = CODEC_TYPE_AUDIO;
if (avcodec_open(audioCodec, codec) < 0) return;
bufSize = FF_MIN_BUFFER_SIZE * 10;
buf = (uint8_t *)malloc(bufSize);
if (buf == NULL) return;
frameBytes = audioCodec->frame_size * audioCodec->channels * 2;
while (audioSize >= frameBytes)
{
int packetSize;
packetSize = avcodec_encode_audio(audioCodec, buf, bufSize, (short *)audioData);
printf("encoder returned %d bytes of data
", packetSize);
audioData += frameBytes;
audioSize -= frameBytes;
}
}
int main()
{
FILE *stream = fopen("audio.pcm", "rb");
size_t size;
uint8_t *buf;
if (stream == NULL)
{
printf("Unable to open file
");
return 1;
}
fseek(stream, 0, SEEK_END);
size = ftell(stream);
fseek(stream, 0, SEEK_SET);
buf = (uint8_t *)malloc(size);
fread(buf, sizeof(uint8_t), size, stream);
fclose(stream);
EncodeTest(32000, 2, 448000, buf, size);
}
推荐答案
如果比特率低于 386000，问题似乎就会消失.不知道为什么会这样，因为我可以以比直接使用 FAAC 更高的比特率进行编码.但是 128000 对我来说已经足够了，所以我可以继续前进.
The problem seems to go away if the bitrate is less than 386000. Not sure why this is, as I can encode at bitrates higher than that using FAAC directly. But 128000 is good enough for my purposes, so I'm able to move forward.
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