我正在尝试编码aac音频。 Example of MP2 here。我遵循了本文档。我通过调用startEncoding
使用以下代码对音频进行编码,并在2秒钟后调用stopEncoding
。我得到了一定大小的文件,似乎一切正常,但问题是我无法打开或播放它。我不知道为什么。我一定在代码中做错了。
header :
class MediaEncoder {
public:
MediaEncoder(char *filePath);
void startEncoding();
void stopEncoding();
void encode(AVFrame *frame);
bool isEncoding() const;
void startEncoderWorker();
int32_t check_sample_fmt(enum AVSampleFormat sample_fmt);
bool signalExitFuture = false;
int32_t ret;
private:
std::future<void> encodingFuture;
AVCodecContext *avCodecContext;
AVFrame *avFrame;
AVPacket *avPacket;
AVCodec *codec;
FILE* file;
};
cpp:MediaEncoder::MediaEncoder(char *filePath){
buffer = std::make_unique<LockFreeQueue<float>>(recorderBufferSize);
/* find the encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
avCodecContext = avcodec_alloc_context3(codec);
avCodecContext->bit_rate = 64000;
/* check that the encoder supports given sample format */
avCodecContext->sample_fmt = AVSampleFormat::AV_SAMPLE_FMT_FLTP;
/* select other audio parameters supported by the encoder */
avCodecContext->sample_rate = defaultSampleRate;
avCodecContext->channel_layout = AV_CH_LAYOUT_STEREO;
avCodecContext->channels = av_get_channel_layout_nb_channels(avCodecContext->channel_layout);
/* open it */
avcodec_open2(avCodecContext, codec, nullptr)
file = fopen(filePath, "wb");
/* packet for holding encoded output */
avPacket = av_packet_alloc();
/* frame containing input raw audio */
avFrame = av_frame_alloc();
avFrame->nb_samples = avCodecContext->frame_size;
avFrame->format = avCodecContext->sample_fmt;
avFrame->channel_layout = avCodecContext->channel_layout;
/* allocate the data buffers */
av_frame_get_buffer(avFrame, 0);
}
void MediaEncoder::startEncoding() {
// set flags for decoding thread
signalExitFuture = false;
encodingFuture = std::async(std::launch::async, &MediaEncoder::startEncoderWorker, this);
}
void MediaEncoder::stopEncoding() {
signalExitFuture = true;
}
bool MediaEncoder::isEncoding() const {
return !signalExitFuture;
}
void MediaEncoder::encode(AVFrame *frame) {
/* send the frame for encoding */
ret = avcodec_send_frame(avCodecContext, frame);
if (ret < 0) {
LOGE("Error sending the frame to the encoder %s",
av_err2str(ret));
*recorderStatePointer = MediaEncoderState::RUNTIME_FAIL;
return;
}
/* read all the available output packets (in general there may be any
* number of them */
while (ret >= 0) {
ret = avcodec_receive_packet(avCodecContext, avPacket);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
return;
} else if (ret < 0) {
LOGE("Error encoding audio frame %s",
av_err2str(ret));
*recorderStatePointer = MediaEncoderState::RUNTIME_FAIL;
return;
}
/* Solution begins here
int aac_profile = 2; // AAC LC
int frequencey_index = 4; // 44,100Hz
int channel_configuration = 2; // stereo (left, right)
int frame_length = avPacket->size + 7; // your frame length
unsigned char adts_header[7];
// fill in ADTS data
adts_header[0] = (unsigned char) 0xFF;
adts_header[1] = (unsigned char) 0xF9;
adts_header[2] = (unsigned char) (((aac_profile -1) << 6 ) + (frequencey_index << 2) + (channel_configuration >> 2));
adts_header[3] = (unsigned char) (((channel_configuration & 3) << 6) + (frame_length >> 11));
adts_header[4] = (unsigned char) ((frame_length & 0x7FF) >> 3);
adts_header[5] = (unsigned char) (((frame_length & 7) << 5) + 0x1F);
adts_header[6] = (unsigned char) 0xFC;
fwrite(adts_header, 1, 7, file);
Solution ends here */
fwrite(avPacket->data, 1, avPacket->size, file);
av_packet_unref(avPacket);
}
}
void MediaEncoder::startEncoderWorker() {
try {
float *leftChannel;
float *rightChannel;
float val;
while (!signalExitFuture) {
ret = av_frame_make_writable(avFrame);
if (ret < 0) {
LOGE("av_frame_make_writable Can not Ensure that the frame data is writable. %s",
av_err2str(ret));
*recorderStatePointer = MediaEncoderState::RUNTIME_FAIL;
return;
}
leftChannel = (float *) avFrame->data[0];
rightChannel = (float *) avFrame->data[1];
for (int32_t i = 0; i < avCodecContext->frame_size; ++i) {
leftChannel[i] = 0.4;
rightChannel[i] = 0.4;
}
encode(avFrame);
}
/* flush the encoder */
encode(nullptr);
fclose(file);
LOGE("Encoding finished!");
av_frame_free(&avFrame);
av_packet_free(&avPacket);
avcodec_free_context(&avCodecContext);
} catch (std::exception &e) {
LOGE("startEncoderWorker uncaught exception %s", e.what());
}
LOGE("Deleting Media Encoder!");
}
这是一个用实际 float pcm数据记录的aac文件记录的11秒,而不是我发布的代码中的0.4。 Google Drive Link上面的代码有效。感谢传说@ Markus-Schumann
最佳答案
我假设AAC编码器输出原始AAC帧。将原始AAC帧写入文件不会创建可播放的输出。
因此,您必须添加帧头或将输出复用到MP4文件中。
查找ADIF,ADTS或MP4作为文件格式。
假设您想进行ADTS:
int aac_profile = 2; // AAC LC
int frequencey_index = 4; // 44,100Hz
int channel_configuration = 2; // stereo (left, right)
// https://wiki.multimedia.cx/index.php/MPEG-4_Audio#Channel_Configurations
int frame_length; // your frame length + 7 bytes for the header
unsigned char adts_header[7];
// Take look here: https://wiki.multimedia.cx/index.php/ADTS
// fill in ADTS data
adts_header[0] = (unsigned char) 0xFF;
adts_header[1] = (unsigned char) 0xF9;
adts_header[2] = (unsigned char) (((aac_profile -1) << 6 ) + (frequencey_index << 2) + (channel_configuration >> 2));
adts_header[3] = (unsigned char) (((channel_configuration & 3) << 6) + (frame_length >> 11));
adts_header[4] = (unsigned char) ((frame_length & 0x7FF) >> 3);
adts_header[5] = (unsigned char) (((frame_length & 7) << 5) + 0x1F);
adts_header[6] = (unsigned char) 0xFC;
因此,您必须在编码器发出的每个帧之前加上上述 header ,以确保将frame_length设置为编码器发出的AAC缓冲区长度。在磁盘上,它看起来应该像:| ADTS header | AAC框架| ADTS header | AAC框架| ... | ADTS header | AAC框架|
关于c++ - FFmpeg编码AAC音频,无法播放编码文件,我们在Stack Overflow上找到一个类似的问题:https://stackoverflow.com/questions/65013622/