我正在研究将在 Unity 插件中使用的视频编码。我已经使图像编码工作,但现在我在处理音频。所以只尝试将音频转换为 AAC 编码的 mp4 文件。我卡住了。结果文件不包含任何内容。另外,据我了解,ffmpeg 中的 AAC 仅支持 AV_SAMPLE_FMT_FLTP,这就是我使用它的原因。这是我的代码:
设置:
int initialize_encoding_audio(const char *filename)
{
int ret;
AVCodecID aud_codec_id = AV_CODEC_ID_AAC;
AVSampleFormat sample_fmt = AV_SAMPLE_FMT_FLTP;
avcodec_register_all();
av_register_all();
aud_codec = avcodec_find_encoder(aud_codec_id);
avcodec_register(aud_codec);
if (!aud_codec)
return COULD_NOT_FIND_AUD_CODEC;
aud_codec_context = avcodec_alloc_context3(aud_codec);
if (!aud_codec_context)
return CONTEXT_CREATION_ERROR;
aud_codec_context->bit_rate = 192000;
aud_codec_context->sample_rate = select_sample_rate(aud_codec);
aud_codec_context->sample_fmt = sample_fmt;
aud_codec_context->channel_layout = AV_CH_LAYOUT_STEREO;
aud_codec_context->channels = av_get_channel_layout_nb_channels(aud_codec_context->channel_layout);
aud_codec_context->codec = aud_codec;
aud_codec_context->codec_id = aud_codec_id;
ret = avcodec_open2(aud_codec_context, aud_codec, NULL);
if (ret < 0)
return COULD_NOT_OPEN_AUD_CODEC;
outctx = avformat_alloc_context();
ret = avformat_alloc_output_context2(&outctx, NULL, "mp4", filename);
outctx->audio_codec = aud_codec;
outctx->audio_codec_id = aud_codec_id;
audio_st = avformat_new_stream(outctx, aud_codec);
audio_st->codecpar->bit_rate = aud_codec_context->bit_rate;
audio_st->codecpar->sample_rate = aud_codec_context->sample_rate;
audio_st->codecpar->channels = aud_codec_context->channels;
audio_st->codecpar->channel_layout = aud_codec_context->channel_layout;
audio_st->codecpar->codec_id = aud_codec_id;
audio_st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
audio_st->codecpar->format = sample_fmt;
audio_st->codecpar->frame_size = aud_codec_context->frame_size;
audio_st->codecpar->block_align = aud_codec_context->block_align;
audio_st->codecpar->initial_padding = aud_codec_context->initial_padding;
outctx->streams = new AVStream*[1];
outctx->streams[0] = audio_st;
av_dump_format(outctx, 0, filename, 1);
if (!(outctx->oformat->flags & AVFMT_NOFILE))
{
if (avio_open(&outctx->pb, filename, AVIO_FLAG_WRITE) < 0)
return COULD_NOT_OPEN_FILE;
}
ret = avformat_write_header(outctx, NULL);
aud_frame = av_frame_alloc();
aud_frame->nb_samples = aud_codec_context->frame_size;
aud_frame->format = aud_codec_context->sample_fmt;
aud_frame->channel_layout = aud_codec_context->channel_layout;
int buffer_size = av_samples_get_buffer_size(NULL, aud_codec_context->channels, aud_codec_context->frame_size,
aud_codec_context->sample_fmt, 0);
av_frame_get_buffer(aud_frame, buffer_size / aud_codec_context->channels);
if (!aud_frame)
return COULD_NOT_ALLOCATE_FRAME;
aud_frame_counter = 0;
return 0;
}
编码:
int encode_audio_samples(uint8_t **aud_samples)
{
int ret;
int buffer_size = av_samples_get_buffer_size(NULL, aud_codec_context->channels, aud_codec_context->frame_size,
aud_codec_context->sample_fmt, 0);
for (size_t i = 0; i < buffer_size / aud_codec_context->channels; i++)
{
aud_frame->data[0][i] = aud_samples[0][i];
aud_frame->data[1][i] = aud_samples[1][i];
}
aud_frame->pts = aud_frame_counter++;
ret = avcodec_send_frame(aud_codec_context, aud_frame);
if (ret < 0)
return ERROR_ENCODING_SAMPLES_SEND;
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
fflush(stdout);
while (true)
{
ret = avcodec_receive_packet(aud_codec_context, &pkt);
if (!ret)
{
av_packet_rescale_ts(&pkt, aud_codec_context->time_base, audio_st->time_base);
pkt.stream_index = audio_st->index;
av_write_frame(outctx, &pkt);
av_packet_unref(&pkt);
}
if (ret == AVERROR(EAGAIN))
break;
else if (ret < 0)
return ERROR_ENCODING_SAMPLES_RECEIVE;
else
break;
}
return 0;
}
完成编码:
int finish_audio_encoding()
{
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
fflush(stdout);
int ret = avcodec_send_frame(aud_codec_context, NULL);
if (ret < 0)
return ERROR_ENCODING_FRAME_SEND;
while (true)
{
ret = avcodec_receive_packet(aud_codec_context, &pkt);
if (!ret)
{
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, aud_codec_context->time_base, audio_st->time_base);
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, aud_codec_context->time_base, audio_st->time_base);
av_write_frame(outctx, &pkt);
av_packet_unref(&pkt);
}
if (ret == -AVERROR(AVERROR_EOF))
break;
else if (ret < 0)
return ERROR_ENCODING_FRAME_RECEIVE;
}
av_write_trailer(outctx);
}
主要内容:
void get_audio_frame(float_t *left_samples, float_t *right_samples, int frame_size, float* t, float* tincr, float* tincr2)
{
int j, i;
float v;
for (j = 0; j < frame_size; j++)
{
v = sin(*t);
*left_samples = v;
*right_samples = v;
left_samples++;
right_samples++;
*t += *tincr;
*tincr += *tincr2;
}
}
int main()
{
int frame_rate = 30; // this should be like 96000 / 1024 or somthing i guess?
float t, tincr, tincr2;
initialize_encoding_audio("audio.mp4");
int sec = 50;
float_t** aud_samples;
int src_samples_linesize;
int src_nb_samples = 1024;
int src_channels = 2;
int ret = av_samples_alloc_array_and_samples((uint8_t***)&aud_samples, &src_samples_linesize, src_channels,
src_nb_samples, AV_SAMPLE_FMT_FLTP, 0);
t = 0;
tincr = 0;
tincr2 = 0;
for (size_t i = 0; i < frame_rate * sec; i++)
{
get_audio_frame(aud_samples[0], aud_samples[1], src_nb_samples, &t, &tincr, &tincr2);
encode_audio_samples((uint8_t **)aud_samples);
}
finish_audio_encoding();
//cleanup();
return 0;
}
我想我想确保我做对的第一件事是合成声音的生成以及我如何将其传输到 AVFrame。我的转换是否正确?但请随时指出任何可能错误的地方。
提前致谢!
编辑:整个来源:http://pastebin.com/jYtmkhek
Edit2:添加了 tincr 和 tincr2 的初始化
最佳答案
除非我遗漏了 pastebin 中的某些内容,否则您忘记了初始化一些变量。您正在使用垃圾来生成样本。
float t, tincr, tincr2;
[...]
get_audio_frame(aud_samples[0], aud_samples[1], src_nb_samples, &t, &tincr, &tincr2);
对于正弦波,您可能希望从 t=0
开始并增加 2 * PI * 频率/采样率
。
此外,avformat_new_stream()
会为您创建流,不要用 new
来创建它。
更新:
我删除了所有 c++
的东西来测试这个。这是有效的代码:pastebin
这是生成的文件:audio.mp4
ffmpeg -i audio.mp4 -filter_complex "showwaves=s=640x120:mode=line:colors=white" -frames:v 1 wave.jpg
差异:
1,6d0
< #include "encoder.h"
< #include <algorithm>
< #include <iterator>
<
< extern "C"
< {
14a9
> #include <math.h>
40,41c35,36
< SwsContext *sws_ctx;
< SwrContext *swr_ctx = NULL;
---
> struct SwsContext *sws_ctx;
> struct SwrContext *swr_ctx = NULL;
76,77c71,72
< AVCodecID aud_codec_id = AV_CODEC_ID_AAC;
< AVSampleFormat sample_fmt = AV_SAMPLE_FMT_FLTP;
---
> enum AVCodecID aud_codec_id = AV_CODEC_ID_AAC;
> enum AVSampleFormat sample_fmt = AV_SAMPLE_FMT_FLTP;
125,126c120,121
< outctx->streams = new AVStream*[1];
< outctx->streams[0] = audio_st;
---
> //outctx->streams = new AVStream*[1];
> //outctx->streams[0] = audio_st;
182c177
< while (true)
---
> while (1)
216c211
< while (true)
---
> while (1)
291c286
< float t, tincr, tincr2;
---
> float t = 0, tincr = 2 * M_PI * 440.0 / 96000, tincr2 = 0;
317d311
< }
关于c++ - 使用 ffmpeg (c++) 编码 AAC,我们在Stack Overflow上找到一个类似的问题: https://stackoverflow.com/questions/40720142/