用libavcodec编码音频到aac [英] Encode audio to aac with libavcodec
问题描述
我正在使用libavcodec（最新的git为3/3/10）将raw pcm编码到aac
（启用了libfaac支持）。我每次通过使用codec_context-> frame_size样本重复调用avcodec_encode_audio
。前四个
调用成功返回，但第五个调用从不返回。当我使用gdb
打破时，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。
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.
我写了一个重复我的问题的小测试程序。它从文件中读取
测试数据，可在此处获取：
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)
如果我使用FAAC，我可以使其全部工作直接，但是如果我可以使用libavcodec，代码将会更清晰一点，因为我也是编码视频，并将其写入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
有什么我吗在我的编解码器
上下文中，可能没有设置或设置不正确？任何帮助是非常感谢！
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\n", 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\n");
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
问题似乎消失如果比特率小于386000.不知道为什么这是，因为我可以直接使用FAAC的比特率进行编码。但128000对于我的目的来说足够好，所以我可以向前迈进。
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.
这篇关于用libavcodec编码音频到aac的文章就介绍到这了，希望我们推荐的答案对大家有所帮助，也希望大家多多支持IT屋！
