我正在使用FFmpeg库生成包含来自各种文件(如MP3,WAV,OGG)的音频的MP4文件,但我遇到了一些问题(为简化起见,此问题不考虑视频部分,因为我已经在那方面做得很好)。我的当前代码打开一个音频文件,解码内容并将其转换为MP4容器,最后将其作为交错帧写入目标文件。对于大多数MP3文件,它都能完美地工作,但是当输入WAV或OGG时,结果MP4中的音频会略微失真,并且经常以错误的速度播放(速度可能比实际快或慢许多倍)。我已经查看了无数使用转换函数(swr_convert)的示例,但似乎无法消除导出音频中的噪声。以下是我如何将音频流添加到MP4中(outContext是输出文件的AVFormatContext):
最后,要进行解码+转换+编码...
audioCodec = avcodec_find_encoder(outContext->oformat->audio_codec);
if (!audioCodec)
die("Could not find audio encoder!");
// Start stream
audioStream = avformat_new_stream(outContext, audioCodec);
if (!audioStream)
die("Could not allocate audio stream!");
audioCodecContext = audioStream->codec;
audioStream->id = 1;
// Setup
audioCodecContext->sample_fmt = AV_SAMPLE_FMT_S16;
audioCodecContext->bit_rate = 128000;
audioCodecContext->sample_rate = 44100;
audioCodecContext->channels = 2;
audioCodecContext->channel_layout = AV_CH_LAYOUT_STEREO;
// Open the codec
if (avcodec_open2(audioCodecContext, audioCodec, NULL) < 0)
die("Could not open audio codec");
要从MP3/WAV/OGG格式的声音文件中打开(使用文件名变量)...
// Create contex
formatContext = avformat_alloc_context();
if (avformat_open_input(&formatContext, filename, NULL, NULL)<0)
die("Could not open file");
// Find info
if (avformat_find_stream_info(formatContext, 0)<0)
die("Could not find file info");
av_dump_format(formatContext, 0, filename, false);
// Find audio stream
streamId = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, NULL, 0);
if (streamId < 0)
die("Could not find Audio Stream");
codecContext = formatContext->streams[streamId]->codec;
// Find decoder
codec = avcodec_find_decoder(codecContext->codec_id);
if (codec == NULL)
die("cannot find codec!");
// Open codec
if (avcodec_open2(codecContext, codec, 0)<0)
die("Codec cannot be found");
// Set up resample context
swrContext = swr_alloc();
if (!swrContext)
die("Failed to alloc swr context");
av_opt_set_int(swrContext, "in_channel_count", codecContext->channels, 0);
av_opt_set_int(swrContext, "in_channel_layout", codecContext->channel_layout, 0);
av_opt_set_int(swrContext, "in_sample_rate", codecContext->sample_rate, 0);
av_opt_set_sample_fmt(swrContext, "in_sample_fmt", codecContext->sample_fmt, 0);
av_opt_set_int(swrContext, "out_channel_count", audioCodecContext->channels, 0);
av_opt_set_int(swrContext, "out_channel_layout", audioCodecContext->channel_layout, 0);
av_opt_set_int(swrContext, "out_sample_rate", audioCodecContext->sample_rate, 0);
av_opt_set_sample_fmt(swrContext, "out_sample_fmt", audioCodecContext->sample_fmt, 0);
if (swr_init(swrContext))
die("Failed to init swr context");
最后,要进行解码+转换+编码...
// Allocate and init re-usable frames
audioFrameDecoded = av_frame_alloc();
if (!audioFrameDecoded)
die("Could not allocate audio frame");
audioFrameDecoded->format = fileCodecContext->sample_fmt;
audioFrameDecoded->channel_layout = fileCodecContext->channel_layout;
audioFrameDecoded->channels = fileCodecContext->channels;
audioFrameDecoded->sample_rate = fileCodecContext->sample_rate;
audioFrameConverted = av_frame_alloc();
if (!audioFrameConverted)
die("Could not allocate audio frame");
audioFrameConverted->nb_samples = audioCodecContext->frame_size;
audioFrameConverted->format = audioCodecContext->sample_fmt;
audioFrameConverted->channel_layout = audioCodecContext->channel_layout;
audioFrameConverted->channels = audioCodecContext->channels;
audioFrameConverted->sample_rate = audioCodecContext->sample_rate;
AVPacket inPacket;
av_init_packet(&inPacket);
inPacket.data = NULL;
inPacket.size = 0;
int frameFinished = 0;
while (av_read_frame(formatContext, &inPacket) >= 0) {
if (inPacket.stream_index == streamId) {
int len = avcodec_decode_audio4(fileCodecContext, audioFrameDecoded, &frameFinished, &inPacket);
if (frameFinished) {
// Convert
uint8_t *convertedData=NULL;
if (av_samples_alloc(&convertedData,
NULL,
audioCodecContext->channels,
audioFrameConverted->nb_samples,
audioCodecContext->sample_fmt, 0) < 0)
die("Could not allocate samples");
int outSamples = swr_convert(swrContext,
&convertedData,
audioFrameConverted->nb_samples,
(const uint8_t **)audioFrameDecoded->data,
audioFrameDecoded->nb_samples);
if (outSamples < 0)
die("Could not convert");
size_t buffer_size = av_samples_get_buffer_size(NULL,
audioCodecContext->channels,
audioFrameConverted->nb_samples,
audioCodecContext->sample_fmt,
0);
if (buffer_size < 0)
die("Invalid buffer size");
if (avcodec_fill_audio_frame(audioFrameConverted,
audioCodecContext->channels,
audioCodecContext->sample_fmt,
convertedData,
buffer_size,
0) < 0)
die("Could not fill frame");
AVPacket outPacket;
av_init_packet(&outPacket);
outPacket.data = NULL;
outPacket.size = 0;
if (avcodec_encode_audio2(audioCodecContext, &outPacket, audioFrameConverted, &frameFinished) < 0)
die("Error encoding audio frame");
if (frameFinished) {
outPacket.stream_index = audioStream->index;
if (av_interleaved_write_frame(outContext, &outPacket) != 0)
die("Error while writing audio frame");
av_free_packet(&outPacket);
}
}
}
}
av_frame_free(&audioFrameConverted);
av_frame_free(&audioFrameDecoded);
av_free_packet(&inPacket);
我还尝试设置适当的pts值来处理传出的帧,但这似乎并没有影响声音质量。
我也不确定是否应该为转换后的数据分配空间,可以使用av_samples_alloc吗?avcodec_fill_audio_frame呢?我是否走在正确的轨道上?
非常感谢您的任何建议(如果需要,我也可以发送导出的MP4文件,让您听到扭曲的声音)。