我已经使用 JNA 实现了 hello world libavcodec 来生成包含纯 440Hz 正弦波的 wav 文件。但是当我实际运行程序时the wav file contains annoying clicks and blips (与 pure sin wav created from the C program 相比)。我怎么调用 avcodec_encode_audio2 是错误的？
这是我的 Java 代码。 All the sources如果您想尝试编译它，也可以在 github 上找到。
import java.io.IOException;
import java.nio.ByteBuffer;
import java.nio.ByteOrder;
import java.nio.IntBuffer;
import java.util.Objects;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.TargetDataLine;
public class Sin {
/**
* Abstract class that allows you to put the initialization and cleanup
* code at the same place instead of separated by the big try block.
*/
public static abstract class SharedPtr<T> implements AutoCloseable {
public T ptr;
public SharedPtr(T ptr) {
this.ptr = ptr;
}
/**
* Abstract override forces method to throw no checked exceptions.
* Subclasses will call a C function that throws no exceptions.
*/
@Override public abstract void close();
}
/**
* @param args
* @throws IOException
* @throws LineUnavailableException
*/
public static void main(String[] args) throws IOException, LineUnavailableException {
final AvcodecLibrary avcodec = AvcodecLibrary.INSTANCE;
final AvformatLibrary avformat = AvformatLibrary.INSTANCE;
final AvutilLibrary avutil = AvutilLibrary.INSTANCE;
avcodec.avcodec_register_all();
avformat.av_register_all();
AVOutputFormat.ByReference format = null;
String format_name = "wav", file_url = "file:sinjava.wav";
for (AVOutputFormat.ByReference formatIter = avformat.av_oformat_next(null); formatIter != null; formatIter = avformat.av_oformat_next(formatIter)) {
formatIter.setAutoWrite(false);
String iterName = formatIter.name;
if (format_name.equals(iterName)) {
format = formatIter;
break;
}
}
Objects.requireNonNull(format);
System.out.format("Found format %s%n", format_name);
AVCodec codec = avcodec.avcodec_find_encoder(format.audio_codec); // one of AvcodecLibrary.CodecID
Objects.requireNonNull(codec);
codec.setAutoWrite(false);
try (
SharedPtr<AVFormatContext> fmtCtxPtr = new SharedPtr<AVFormatContext>(avformat.avformat_alloc_context()) {@Override public void close(){if (null!=ptr) avformat.avformat_free_context(ptr);}};
) {
AVFormatContext fmtCtx = Objects.requireNonNull(fmtCtxPtr.ptr);
fmtCtx.setAutoWrite(false);
fmtCtx.setAutoRead(false);
fmtCtx.oformat = format; fmtCtx.writeField("oformat");
AVStream st = avformat.avformat_new_stream(fmtCtx, codec);
if (null == st)
throw new IllegalStateException();
AVCodecContext c = st.codec;
if (null == c)
throw new IllegalStateException();
st.setAutoWrite(false);
fmtCtx.readField("nb_streams");
st.id = fmtCtx.nb_streams - 1; st.writeField("id");
assert st.id >= 0;
System.out.format("New stream: id=%d%n", st.id);
if (0 != (format.flags & AvformatLibrary.AVFMT_GLOBALHEADER)) {
c.flags |= AvcodecLibrary.CODEC_FLAG_GLOBAL_HEADER;
}
c.writeField("flags");
c.bit_rate = 64000; c.writeField("bit_rate");
int bestSampleRate;
if (null == codec.supported_samplerates) {
bestSampleRate = 44100;
} else {
bestSampleRate = 0;
for (int offset = 0, sample_rate = codec.supported_samplerates.getInt(offset); sample_rate != 0; codec.supported_samplerates.getInt(++offset)) {
bestSampleRate = Math.max(bestSampleRate, sample_rate);
}
assert bestSampleRate > 0;
}
c.sample_rate = bestSampleRate; c.writeField("sample_rate");
c.channel_layout = AvutilLibrary.AV_CH_LAYOUT_STEREO; c.writeField("channel_layout");
c.channels = avutil.av_get_channel_layout_nb_channels(c.channel_layout); c.writeField("channels");
assert 2 == c.channels;
c.sample_fmt = AvutilLibrary.AVSampleFormat.AV_SAMPLE_FMT_S16; c.writeField("sample_fmt");
c.time_base.num = 1;
c.time_base.den = bestSampleRate;
c.writeField("time_base");
c.setAutoWrite(false);
AudioFormat javaSoundFormat = new AudioFormat(bestSampleRate, Short.SIZE, c.channels, true, ByteOrder.nativeOrder() == ByteOrder.BIG_ENDIAN);
DataLine.Info javaDataLineInfo = new DataLine.Info(TargetDataLine.class, javaSoundFormat);
if (! AudioSystem.isLineSupported(javaDataLineInfo))
throw new IllegalStateException();
int err;
if ((err = avcodec.avcodec_open(c, codec)) < 0) {
throw new IllegalStateException();
}
assert c.channels != 0;
AVIOContext.ByReference[] ioCtxReference = new AVIOContext.ByReference[1];
if (0 != (err = avformat.avio_open(ioCtxReference, file_url, AvformatLibrary.AVIO_FLAG_WRITE))) {
throw new IllegalStateException("averror " + err);
}
try (
SharedPtr<AVIOContext.ByReference> ioCtxPtr = new SharedPtr<AVIOContext.ByReference>(ioCtxReference[0]) {@Override public void close(){if (null!=ptr) avutil.av_free(ptr.getPointer());}}
) {
AVIOContext.ByReference ioCtx = Objects.requireNonNull(ioCtxPtr.ptr);
fmtCtx.pb = ioCtx; fmtCtx.writeField("pb");
int averr = avformat.avformat_write_header(fmtCtx, null);
if (averr < 0) {
throw new IllegalStateException("" + averr);
}
st.read(); // it is modified by avformat_write_header
System.out.format("Wrote header. fmtCtx->nb_streams=%d, st->time_base=%d/%d; st->avg_frame_rate=%d/%d%n", fmtCtx.nb_streams, st.time_base.num, st.time_base.den, st.avg_frame_rate.num, st.avg_frame_rate.den);
avformat.avio_flush(ioCtx);
int frame_size = c.frame_size != 0 ? c.frame_size : 4096;
int expectedBufferSize = frame_size * c.channels * (Short.SIZE/8);
boolean supports_small_last_frame = c.frame_size == 0 ? true : 0 != (codec.capabilities & AvcodecLibrary.CODEC_CAP_SMALL_LAST_FRAME);
int bufferSize = avutil.av_samples_get_buffer_size((IntBuffer)null, c.channels, frame_size, c.sample_fmt, 1);
assert bufferSize == expectedBufferSize: String.format("expected %d; got %d", expectedBufferSize, bufferSize);
ByteBuffer samples = ByteBuffer.allocate(expectedBufferSize);
samples.order(ByteOrder.nativeOrder());
int audio_time = 0; // unit: (c.time_base) s = (1/c.sample_rate) s
int audio_sample_count = supports_small_last_frame ?
3 * c.sample_rate :
3 * c.sample_rate / frame_size * frame_size;
while (audio_time < audio_sample_count) {
int frame_audio_time = audio_time;
samples.clear();
int nb_samples_in_frame = 0;
// encode a single tone sound
for (; samples.hasRemaining() && audio_time < audio_sample_count; nb_samples_in_frame++, audio_time++) {
double x = 2*Math.PI*440/c.sample_rate * audio_time;
double y = 10000 * Math.sin(x);
samples.putShort((short) y);
samples.putShort((short) y);
}
samples.flip();
try (
SharedPtr<AVFrame> framePtr = new SharedPtr<AVFrame>(avcodec.avcodec_alloc_frame()) {@Override public void close() {if (null!=ptr) avutil.av_free(ptr.getPointer());}};
) {
AVFrame frame = Objects.requireNonNull(framePtr.ptr);
frame.setAutoRead(false); // will be an in param
frame.setAutoWrite(false);
frame.nb_samples = nb_samples_in_frame; frame.writeField("nb_samples"); // actually unused during encoding
// Presentation time, in AVStream.time_base units.
frame.pts = avutil.av_rescale_q(frame_audio_time, c.time_base, st.time_base); // i * codec_time_base / st_time_base
frame.writeField("pts");
assert c.channels > 0;
int bytesPerSample = avutil.av_get_bytes_per_sample(c.sample_fmt);
assert bytesPerSample > 0;
if (0 != (err = avcodec.avcodec_fill_audio_frame(frame, c.channels, c.sample_fmt, samples, samples.capacity(), 1))) {
throw new IllegalStateException(""+err);
}
AVPacket packet = new AVPacket(); // one of the few structs from ffmpeg with guaranteed size
avcodec.av_init_packet(packet);
packet.size = 0;
packet.data = null;
packet.stream_index = st.index; packet.writeField("stream_index");
// encode the samples
IntBuffer gotPacket = IntBuffer.allocate(1);
if (0 != (err = avcodec.avcodec_encode_audio2(c, packet, frame, gotPacket))) {
throw new IllegalStateException("" + err);
} else if (0 != gotPacket.get()) {
packet.read();
averr = avformat.av_write_frame(fmtCtx, packet);
if (averr < 0)
throw new IllegalStateException("" + averr);
}
System.out.format("encoded frame: codec time = %d; pts=%d = av_rescale_q(%d,%d/%d,%d/%d) (%.02fs) contains %d samples (%.02fs); got_packet=%d; packet.size=%d%n",
frame_audio_time,
frame.pts,
frame_audio_time, st.codec.time_base.num,st.codec.time_base.den,st.time_base.num,st.time_base.den,
1.*frame_audio_time/c.sample_rate, frame.nb_samples, 1.*frame.nb_samples/c.sample_rate, gotPacket.array()[0], packet.size);
}
}
if (0 != (err = avformat.av_write_trailer(fmtCtx))) {
throw new IllegalStateException();
}
avformat.avio_flush(ioCtx);
}
}
System.out.println("Done writing");
}
}
我还用C重写了它，C版本运行良好，没有任何问题。但我不知道我如何以不同的方式使用这个库；所有库函数调用都应该相同!
//! gcc --std=c99 sin.c $(pkg-config --cflags --libs libavutil libavformat libavcodec) -o sin
// sudo apt-get install libswscale-dev
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libavcodec/avcodec.h>
int main(int argc, char *argv[]) {
const char *format_name = "wav", *file_url = "file:sin.wav";
avcodec_register_all();
av_register_all();
AVOutputFormat *format = NULL;
for (AVOutputFormat *formatIter = av_oformat_next(NULL); formatIter != NULL; formatIter = av_oformat_next(formatIter)) {
int hasEncoder = NULL != avcodec_find_encoder(formatIter->audio_codec);
if (0 == strcmp(format_name, formatIter->name)) {
format = formatIter;
break;
}
}
printf("Found format %s\n", format->name);
AVCodec *codec = avcodec_find_encoder(format->audio_codec);
if (! codec) {
fprintf(stderr, "Could not find codec %d\n", format->audio_codec);
exit(1);
}
AVFormatContext *fmtCtx = avformat_alloc_context();
if (! fmtCtx) {
fprintf(stderr, "error allocating AVFormatContext\n");
exit(1);
}
fmtCtx->oformat = format;
AVStream *st = avformat_new_stream(fmtCtx, codec);
if (! st) {
fprintf(stderr, "error allocating AVStream\n");
exit(1);
}
if (fmtCtx->nb_streams != 1) {
fprintf(stderr, "avformat_new_stream should have incremented nb_streams, but it's still %d\n", fmtCtx->nb_streams);
exit(1);
}
AVCodecContext *c = st->codec;
if (! c) {
fprintf(stderr, "avformat_new_stream should have allocated a AVCodecContext for my stream\n");
exit(1);
}
st->id = fmtCtx->nb_streams - 1;
printf("Created stream %d\n", st->id);
if (0 != (format->flags & AVFMT_GLOBALHEADER)) {
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
c->bit_rate = 64000;
int bestSampleRate;
if (NULL == codec->supported_samplerates) {
bestSampleRate = 44100;
printf("Setting sample rate: %d\n", bestSampleRate);
} else {
bestSampleRate = 0;
for (const int *sample_rate_iter = codec->supported_samplerates; *sample_rate_iter != 0; sample_rate_iter++) {
if (*sample_rate_iter >= bestSampleRate)
bestSampleRate = *sample_rate_iter;
}
printf("Using best supported sample rate: %d\n", bestSampleRate);
}
c->sample_rate = bestSampleRate;
c->channel_layout = AV_CH_LAYOUT_STEREO;
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
c->time_base.num = 1;
c->time_base.den = c->sample_rate;
if (c->channels != 2) {
fprintf(stderr, "av_get_channel_layout_nb_channels returned %d instead of 2\n", c->channels);
exit(1);
}
c->sample_fmt = AV_SAMPLE_FMT_S16;
int averr;
if ((averr = avcodec_open2(c, codec, NULL)) < 0) {
fprintf(stderr, "avcodec_open2 returned error %d\n", averr);
exit(1);
}
AVIOContext *ioCtx = NULL;
if (0 != (averr = avio_open(&ioCtx, file_url, AVIO_FLAG_WRITE))) {
fprintf(stderr, "avio_open returned error %d\n", averr);
exit(1);
}
if (ioCtx == NULL) {
fprintf(stderr, "AVIOContext should have been set by avio_open\n");
exit(1);
}
fmtCtx->pb = ioCtx;
if (0 != (averr = avformat_write_header(fmtCtx, NULL))) {
fprintf(stderr, "avformat_write_header returned error %d\n", averr);
exit(1);
}
printf("Wrote header. fmtCtx->nb_streams=%d, st->time_base=%d/%d; st->avg_frame_rate=%d/%d\n", fmtCtx->nb_streams, st->time_base.num, st->time_base.den, st->avg_frame_rate.num, st->avg_frame_rate.den);
int align = 1;
int sample_size = av_get_bytes_per_sample(c->sample_fmt);
if (sample_size != sizeof(int16_t)) {
fprintf(stderr, "expected sample size=%zu but got %d\n", sizeof(int16_t), sample_size);
exit(1);
}
int frame_size = c->frame_size != 0 ? c->frame_size : 4096;
int bufferSize = av_samples_get_buffer_size(NULL, c->channels, frame_size, c->sample_fmt, align);
int expectedBufferSize = frame_size * c->channels * sample_size;
int supports_small_last_frame = c->frame_size == 0 ? 1 : 0 != (codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME);
if (bufferSize != expectedBufferSize) {
fprintf(stderr, "expected buffer size=%d but got %d\n", expectedBufferSize, bufferSize);
exit(1);
}
int16_t *samples = (int16_t*)malloc(bufferSize);
uint32_t audio_time = 0; // unit: (1/c->sample_rate) s
uint32_t audio_sample_count = supports_small_last_frame ?
3 * c->sample_rate :
3 * c->sample_rate / frame_size * frame_size;
while (audio_time < audio_sample_count) {
uint32_t frame_audio_time = audio_time; // unit: (1/c->sample_rate) s
AVFrame *frame = avcodec_alloc_frame();
if (frame == NULL) {
fprintf(stderr, "avcodec_alloc_frame failed\n");
exit(1);
}
for (uint32_t i = 0; i != frame_size && audio_time < audio_sample_count; i++, audio_time++) {
samples[2*i] = samples[2*i + 1] = 10000 * sin(2*M_PI*440/c->sample_rate * audio_time);
frame->nb_samples = i+1; // actually unused during encoding
}
// frame->format = c->sample_fmt; // unused during encoding
frame->pts = av_rescale_q(frame_audio_time, c->time_base, st->time_base);
if (0 != (averr = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, (const uint8_t*)samples, bufferSize, align))) {
fprintf(stderr, "avcodec_fill_audio_frame returned error %d\n", averr);
exit(1);
}
AVPacket packet;
av_init_packet(&packet);
packet.data = NULL;
packet.size = 0;
int got_packet;
if (0 != (averr = avcodec_encode_audio2(c, &packet, frame, &got_packet))) {
fprintf(stderr, "avcodec_encode_audio2 returned error %d\n", averr);
exit(1);
}
if (got_packet) {
packet.stream_index = st->index;
if (0 < (averr = av_write_frame(fmtCtx, &packet))) {
fprintf(stderr, "av_write_frame returned error %d\n", averr);
exit(1);
} else if (averr == 1) {
// end of stream wanted.
}
}
printf("encoded frame: codec time = %u; format pts=%ld = av_rescale_q(%u,%d/%d,%d/%d) (%.02fs) contains %d samples (%.02fs); got_packet=%d; packet.size=%d\n",
frame_audio_time,
frame->pts,
frame_audio_time, c->time_base.num, c->time_base.den, st->time_base.num, st->time_base.den,
1.*frame_audio_time/c->sample_rate, frame->nb_samples, 1.*frame->nb_samples/c->sample_rate, got_packet, packet.size);
av_free(frame);
}
free(samples);
cleanupFile:
if (0 != (averr = av_write_trailer(fmtCtx))) {
fprintf(stderr, "av_write_trailer returned error %d\n", averr);
exit(1);
}
avio_flush(ioCtx);
avio_close(ioCtx);
avformat_free_context(fmtCtx);
}
最佳答案
问题是 ByteBuffer.allocate(int) 创建的缓冲区的地址在 JNA 函数调用中不稳定。每次调用 native 函数时，它都会将字节复制到临时数组中以供该调用使用。相比之下，ByteBuffer.allocateDirect(int) 创建一个 native 指针稳定的缓冲区。这显然是一个well-known pitfall of using ByteBuffer in JNA ，但我没有注意到 Using Pointers and Arrays 的细则中。 .
因此，我只需将示例创建修复为 ByteBuffer Samples = ByteBuffer.allocateDirect(expectedBufferSize);。随后的avcodec_fill_audio_frame调用不会复制样本；它只是将frame->data[0]指向uint8_t*地址，因此samples数组需要有一个稳定的地址。
关于ffmpeg - 为什么在使用 Java JNA 编码声音文件时出现提示音？，我们在Stack Overflow上找到一个类似的问题： https://stackoverflow.com/questions/18392817/