[英]FFmpeg encoding aac audio, encoded file can not be played
我正在尝试对 aac 音频进行编码。 MP2 的例子在这里。 我遵循了这个文档。 我通过调用startEncoding
使用下面的代码对音频进行编码,2 秒后我调用stopEncoding
。 一切似乎都很好我得到了一个大小不一的文件,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什么。
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;
};
cp:
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!");
}
这是用真实浮点 pcm 数据记录的 11 秒 aac 文件,而不是我发布的代码中的 0.4 秒。 谷歌云端硬盘链接
上面的代码有效。 感谢传奇@Markus-Schumann
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 output。 因此,您必须添加帧头或将 output 混合到 MP4 文件中。 寻找 ADIF、ADTS 或 MP4 作为文件格式。
假设您想做 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,您必须使用上述 header 为来自编码器的每个帧添加前缀,确保将 frame_length 设置为来自编码器的 AAC 缓冲区长度。 在磁盘上它应该看起来像:|ADTS 头|AAC 帧|ADTS 头|AAC 帧|...|ADTS 头|AAC 帧|
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